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From: Gerardo A. <ope...@ge...> - 2007-01-15 21:21:05
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This is working beautifully. Thanks Gerardo Amaya Joegen E. Baclor wrote: > Gerardo Amaya wrote: > >> >> SoftPhone|10.0.0.3----------10.0.0.2|OpenSBC|192.168.1.112-------192.168.1.2|Asterisk >> >> Here are the logs from the Registration Process >> ========================================================================================================================================== >> 2007/01/12 15:48:12.331 Transport( READ ) Debug3 RCV: XOR=0 514 >> Bytes from RCVADDR: 10.0.0.3:RCVPORT: 5052:UDP (REGISTER sip:10.0.0.2 >> SIP/2.0) >> REGISTER sip:10.0.0.2 SIP/2.0 >> From: "test" <sip:test@10.0.0.2>;tag=1c7cfc45 >> To: "test" <sip:test@10.0.0.2> >> Via: SIP/2.0/UDP >> 10.0.0.3:5052;branch=z9hG4bK-d87543-073cbf101c05244f-1--d87543- >> CSeq: 1 REGISTER >> Call-ID: MzkxNjAzYzI5NTgyYWNhMDhlNTY3YzkzOTFiYmVkMDI. >> Contact: <sip:test@10.0.0.3:5052;rinstance=c38cb44158e52847> >> User-Agent: X-Lite release 1006e stamp 34025 >> Expires: 3600 >> Max-Forwards: 70 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, >> SUBSCRIBE, INFO >> Content-Length: 0 >> >> >> >> >> > > Hi Gerardo, > > I mispoke a bit in my last email. the TO URI shoudl be > > To: "test" <address/domain of your registrar here> > > Since 10.0.0.2 is a local address of OpenSBC, it will try to > authenticate the registration as using local domain accounts. > > If you are using a softphone... see to it that configuration is > > 1. OutboundProxy = 10.0.0.2 > 2. Registar/Domain = asterisk box. > > This will result to REGISTERs being sent to 10.0.0.2 with a To: URI > pointing to asterisk > > Joegen > > > > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
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From: Joegen E. B. <joe...@gm...> - 2007-01-15 20:55:57
|
Gerardo Amaya wrote: > > > SoftPhone|10.0.0.3----------10.0.0.2|OpenSBC|192.168.1.112-------192.168.1.2|Asterisk > > Here are the logs from the Registration Process > ========================================================================================================================================== > 2007/01/12 15:48:12.331 Transport( READ ) Debug3 RCV: XOR=0 514 > Bytes from RCVADDR: 10.0.0.3:RCVPORT: 5052:UDP (REGISTER sip:10.0.0.2 > SIP/2.0) > REGISTER sip:10.0.0.2 SIP/2.0 > From: "test" <sip:test@10.0.0.2>;tag=1c7cfc45 > To: "test" <sip:test@10.0.0.2> > Via: SIP/2.0/UDP > 10.0.0.3:5052;branch=z9hG4bK-d87543-073cbf101c05244f-1--d87543- > CSeq: 1 REGISTER > Call-ID: MzkxNjAzYzI5NTgyYWNhMDhlNTY3YzkzOTFiYmVkMDI. > Contact: <sip:test@10.0.0.3:5052;rinstance=c38cb44158e52847> > User-Agent: X-Lite release 1006e stamp 34025 > Expires: 3600 > Max-Forwards: 70 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, INFO > Content-Length: 0 > > > > Hi Gerardo, I mispoke a bit in my last email. the TO URI shoudl be To: "test" <address/domain of your registrar here> Since 10.0.0.2 is a local address of OpenSBC, it will try to authenticate the registration as using local domain accounts. If you are using a softphone... see to it that configuration is 1. OutboundProxy = 10.0.0.2 2. Registar/Domain = asterisk box. This will result to REGISTERs being sent to 10.0.0.2 with a To: URI pointing to asterisk Joegen |
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From: Michael D. <mi...@st...> - 2007-01-15 20:06:28
|
Hello, I have difficulties compiling opensbc on FreeBSD-6.1-RELEASE-p10. Here is what I have done: ===> expat is installed from ports, version is: # pkg_info *expat* Information for expat-2.0.0_1: Comment: XML 1.0 parser written in C Description: Expat is an XML 1.0 parser written in C. It aims to be fully conforming. It is currently not a validating XML processor. WWW: http://sourceforge.net/projects/expat/ 1.) opensipstack version is 1.1.3 a) opensipstack with expat* (expat.h and expat_external.h) in /usr/local/include ===> Does opensipstack find expat correctly? See the configure output below: # ./configure [...] checking expat.h usability... no checking expat.h presence... no checking for expat.h... no checking for /usr/local/include/expat.h... yes [...] # gmake optnoshared This compiles fine. But opensbc does not compile with opensipstack compiled this way, see below in 2.) b) Since ./configure above in a) said ``expat.h usability... no'' I thought this could be related to finding it in /usr/include, so I (soft)linked expat* to /usr/include. Here is what ./configure thinks now about expat.h usability: # ./configure [...] checking expat.h usability... yes checking expat.h presence... yes checking for expat.h... yes [...] Unfortunately gmake does not finish properly: # gmake optnoshared [...] gmake[2]: Entering directory `/usr/local/src/opensipstack/opensipstack/src' g++ -DP_USE_PRAGMA -D_REENTRANT -I/usr/local/src/opensipstack/opensipstack/include -DP_USE_PRAGMA -D_REENTRANT -I/usr/local/src/opensipstack/opensipstack/include -Wall -DPTRACING -I/usr/local/src/opensipstack/opensipstack/include -I/usr/local/src/opensipstack/opensipstack/include -DPTRACING -I/usr/local/src/opensipstack/opensipstack/include -Os -felide-constructors -Wreorder -c PresencePackage.cxx -o /usr/local/src/opensipstack/opensipstack/lib/obj_FreeBSD_x86_r/PresencePackage.o In file included from /usr/local/src/opensipstack/opensipstack/include/PresencePackage.h:37, from PresencePackage.cxx:32: /usr/local/src/opensipstack/opensipstack/include/PresenceResource.h:43: error: `Presence' has not been declared /usr/local/src/opensipstack/opensipstack/include/PresenceResource.h:43: error: expected namespace-name before ';' token /usr/local/src/opensipstack/opensipstack/include/PresenceResource.h:43: error: `<type error>' is not a namespace /usr/local/src/opensipstack/opensipstack/include/PresenceResource.h:79: error: expected `,' or `...' before '&' token /usr/local/src/opensipstack/opensipstack/include/PresenceResource.h:80: error: ISO C++ forbids declaration of `PIDF' with no type In file included from PresencePackage.cxx:32: /usr/local/src/opensipstack/opensipstack/include/PresencePackage.h:82: error: expected `,' or `...' before '&' token /usr/local/src/opensipstack/opensipstack/include/PresencePackage.h:83: error: ISO C++ forbids declaration of `PIDF' with no type PresencePackage.cxx:99: error: expected `,' or `...' before '&' token PresencePackage.cxx:101: error: ISO C++ forbids declaration of `PIDF' with no type PresencePackage.cxx: In member function `virtual BOOL Presence::PresencePackage::OnNotification(Presence::PresenceResource&, int)': PresencePackage.cxx:102: error: `pidf' undeclared (first use this function) PresencePackage.cxx:102: error: (Each undeclared identifier is reported only once for each function it appears in.) gmake[2]: *** [/usr/local/src/opensipstack/opensipstack/lib/obj_FreeBSD_x86_r/PresencePackage.o] Error 1 gmake[2]: Leaving directory `/usr/local/src/opensipstack/opensipstack/src' gmake[1]: *** [opt] Error 2 gmake[1]: Leaving directory `/usr/local/src/opensipstack/opensipstack' gmake: *** [optnoshared] Error 2 2.) opensbc version is 1.1.1 ===> opensipstack is compiled with method a) and here is the output of where gmake stops in opensbc: # ./configure --prefix=/usr/local/opensbc-1.1.1 [...] # gmake optnoshared g++ -o obj_FreeBSD_x86_r/opensbc -L/usr/local/lib -L/usr/local/lib -L/usr/local/src/opensipstack/opensipstack/lib -L/usr/local/src/opensipstack/opensipstack/lib ./obj_FreeBSD_x86_r/Main.o ./obj_FreeBSD_x86_r/SBCCallHandler.o ./obj_FreeBSD_x86_r/SBCRoutingHandler.o ./obj_FreeBSD_x86_r/SBCAuthHandler.o ./obj_FreeBSD_x86_r/OpenSBC.o ./obj_FreeBSD_x86_r/Router.o ./obj_FreeBSD_x86_r/RouteRecord.o -lopensipstack_FreeBSD_x86_r_s -lpt_FreeBSD_x86_r_s -lpthread -lexpat -lpthread -lexpat /usr/local/src/opensipstack/opensipstack/lib/libopensipstack_FreeBSD_x86_r_s.a(manager.o)(.text+0x8fe): In function `OpalManager::CreateVideoInputDevice(OpalConnection const&, OpalMediaFormat const&, PVideoInputDevice*&, int&)': : undefined reference to `PVideoInputDevice::CreateDeviceByName(PString const&, PPluginManager*)' /usr/local/src/opensipstack/opensipstack/lib/libopensipstack_FreeBSD_x86_r_s.a(manager.o)(.text+0x9fb): In function `OpalManager::CreateVideoOutputDevice(OpalConnection const&, OpalMediaFormat const&, int, PVideoOutputDevice*&, int&)': : undefined reference to `PVideoOutputDevice::CreateDeviceByName(PString const&, PPluginManager*)' /usr/local/src/opensipstack/opensipstack/lib/libopensipstack_FreeBSD_x86_r_s.a(manager.o)(.text+0x1ef4): In function `OpalManager::OpalManager()': : undefined reference to `PVideoInputDevice::GetDriversDeviceNames(PString const&, PPluginManager*)' /usr/local/src/opensipstack/opensipstack/lib/libopensipstack_FreeBSD_x86_r_s.a(manager.o)(.text+0x1fa0): In function `OpalManager::OpalManager()': : undefined reference to `PVideoOutputDevice::GetDriversDeviceNames(PString const&, PPluginManager*)' /usr/local/src/opensipstack/opensipstack/lib/libopensipstack_FreeBSD_x86_r_s.a(manager.o)(.text+0x27d4): In function `OpalManager::OpalManager()': : undefined reference to `PVideoInputDevice::GetDriversDeviceNames(PString const&, PPluginManager*)' /usr/local/src/opensipstack/opensipstack/lib/libopensipstack_FreeBSD_x86_r_s.a(manager.o)(.text+0x2880): In function `OpalManager::OpalManager()': : undefined reference to `PVideoOutputDevice::GetDriversDeviceNames(PString const&, PPluginManager*)' /usr/local/src/opensipstack/opensipstack/lib/libopensipstack_FreeBSD_x86_r_s.a(manager.o)(.text+0x413b): In function `OpalManager::SetVideoInputDevice(PVideoDevice::OpenArgs const&)': : undefined reference to `PVideoInputDevice::GetDriverNames(PPluginManager*)' /usr/local/src/opensipstack/opensipstack/lib/libopensipstack_FreeBSD_x86_r_s.a(manager.o)(.text+0x415f): In function `OpalManager::SetVideoInputDevice(PVideoDevice::OpenArgs const&)': : undefined reference to `PVideoInputDevice::GetDriversDeviceNames(PString const&, PPluginManager*)' /usr/local/src/opensipstack/opensipstack/lib/libopensipstack_FreeBSD_x86_r_s.a(vidcodec.o)(.text+0x9aa): In function `OpalUncompVideoTranscoder::GetOptimalDataFrameSize(int) const': : undefined reference to `PVideoDevice::CalculateFrameBytes(unsigned int, unsigned int, PString const&)' /usr/local/src/opensipstack/opensipstack/lib/libopensipstack_FreeBSD_x86_r_s.a(vidcodec.o)(.text+0xa50): In function `OpalUncompVideoTranscoder::ConvertFrames(RTP_DataFrame const&, PList<RTP_DataFrame>&)': : undefined reference to `PVideoDevice::CalculateFrameBytes(unsigned int, unsigned int, PString const&)' gmake[1]: *** [obj_FreeBSD_x86_r/opensbc] Error 1 gmake[1]: Leaving directory `/usr/local/src/opensipstack/opensbc' gmake: *** [optnoshared] Error 2 ===> I don't understand the output exactly. Obviously something is missing in opensipstack, right? So what is wrong with the compile of opensipstack? Thanks for any help! Mic |
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From: Joegen E. B. <joe...@gm...> - 2007-01-15 18:59:21
|
Joegen E. Baclor wrote: > Gerardo Amaya wrote: >> >> >> SoftPhone|10.0.0.3----------10.0.0.2|OpenSBC|192.168.1.112-------192.168.1.2|Asterisk >> >> >> Here are the logs from the Registration Process >> ========================================================================================================================================== >> >> 2007/01/12 15:48:12.331 Transport( READ ) Debug3 RCV: XOR=0 >> 514 Bytes from RCVADDR: 10.0.0.3:RCVPORT: 5052:UDP (REGISTER >> sip:10.0.0.2 SIP/2.0) >> REGISTER sip:10.0.0.2 SIP/2.0 >> From: "test" <sip:test@10.0.0.2>;tag=1c7cfc45 >> To: "test" <sip:test@10.0.0.2> >> Via: SIP/2.0/UDP >> 10.0.0.3:5052;branch=z9hG4bK-d87543-073cbf101c05244f-1--d87543- >> CSeq: 1 REGISTER >> Call-ID: MzkxNjAzYzI5NTgyYWNhMDhlNTY3YzkzOTFiYmVkMDI. >> Contact: <sip:test@10.0.0.3:5052;rinstance=c38cb44158e52847> >> User-Agent: X-Lite release 1006e stamp 34025 >> Expires: 3600 >> Max-Forwards: 70 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, >> SUBSCRIBE, INFO >> Content-Length: 0 >> >> >> >> > > Hi Gerardo, > > I mispoke a bit in my last email. the TO URI shoudl be > > To: "test" <address/domain of your registrar here> > > Since 10.0.0.2 is a local address of OpenSBC, it will try to > authenticate the registration as using local domain accounts. > > If you are using a softphone... see to it that configuration is > > 1. OutboundProxy = 10.0.0.2 > 2. Registar/Domain = asterisk box. > > This will result to REGISTERs being sent to 10.0.0.2 with a To: URI > pointing to asterisk > > Joegen > > > > |
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From: Joegen E. B. <joe...@gm...> - 2007-01-15 14:25:29
|
Joegen E. Baclor wrote: > Claudio Miceli wrote: >> I started to look for the media stack implementation, but I got lost. >> Which files correspond to that implementation ? >> (Sorry I am starting to work with OpenSipStack). Can you help me ? >> Thanks, >> Claudio Miceli de Farias > > > For the SoftPhone, the media stack is in OPAL classes. If you are > using MSVC, look at the OPAL/Opal Core directory in you project > explorer. On the other hand, OpenSBC media handlers can be found in > B2BMediaInterface.* > > Joegen > |
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From: Joegen E. B. <joe...@gm...> - 2007-01-15 07:52:25
|
Joegen E. Baclor wrote: > Hi Gerardo, > > Can you send me the B2BUA log + the console log + 'ifconfig -a' > printout for this call. You can send it offlist to > jo...@op... > > Joegen > > Gerardo Amaya wrote: >> Hello All. Now regarding the sound, I just found something really >> interesting in the Logs, take a look at this: >> >> 2007/01/12 12:13:18.770 0x83463e8 OSS DTL: >> [YTA2OGI0NzNkYTljMWUwYjBlZmQ2ZmI5ODE1YjgyNjI.-inbound]......| RTP: >> (Audio) addr=10.0.0.2:30000->0.0.0.0:0 xor=0 rx=0 tx=0 lost=0 >> outOfOrder=0 late=0 rxTime=0/0 txTime=0/0 jitter=0/0 >> >> 2007/01/12 12:13:18.770 0x83463e8 OSS DTL: >> [YTA2OGI0NzNkYTljMWUwYjBlZmQ2ZmI5ODE1YjgyNjI.-1-outbound]...| RTP: >> (Audio) addr=192.168.1.112:30002->192.168.1.2:62876 xor=0 rx=3774 >> tx=0 lost=0 outOfOrder=0 late=0 rxTime=19/47 txTime=0/0 jitter=38/39 >> >> From the first Log line you can see that the RTP media connection >> between the OpenSBC and the Softphone is not working, The connection >> is going to 0.0.0.0:0. I wonder if this is because I have no NAT >> configured on the Server and I just have the OpenSBC connected to >> each subnet by an Interface? Could that be the reason? I have no NAT >> configured at all. >> >> The second log makes more sense and thats the RTP media connection >> with the Asterisk Box, It is able to successfully connect, if I check >> the asterisk, the connection appear to be normal, although there is >> no evidence of sound. >> >> Hope This Helps >> >> Gerardo Amaya >> >> Joegen E. Baclor wrote: >> > > |
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From: Joegen E. B. <joe...@gm...> - 2007-01-14 14:57:49
|
Claudio Miceli wrote: > I started to look for the media stack implementation, but I got lost. > Which files correspond to that implementation ? > (Sorry I am starting to work with OpenSipStack). Can you help me ? > Thanks, > Claudio Miceli de Farias For the SoftPhone, the media stack is in OPAL classes. If you are using MSVC, look at the OPAL/Opal Core directory in you project explorer. On the other hand, OpenSBC media handlers can be found in B2BMediaInterface.* Joegen |
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From: Joegen E. B. <joe...@gm...> - 2007-01-14 13:26:40
|
Hi Gerardo, Can you send me the B2BUA log + the console log + 'ifconfig -a' printout for this call. You can send it offlist to jo...@op... Joegen Gerardo Amaya wrote: > Hello All. Now regarding the sound, I just found something really > interesting in the Logs, take a look at this: > > 2007/01/12 12:13:18.770 0x83463e8 OSS DTL: > [YTA2OGI0NzNkYTljMWUwYjBlZmQ2ZmI5ODE1YjgyNjI.-inbound]......| RTP: > (Audio) addr=10.0.0.2:30000->0.0.0.0:0 xor=0 rx=0 tx=0 lost=0 > outOfOrder=0 late=0 rxTime=0/0 txTime=0/0 jitter=0/0 > > 2007/01/12 12:13:18.770 0x83463e8 OSS DTL: > [YTA2OGI0NzNkYTljMWUwYjBlZmQ2ZmI5ODE1YjgyNjI.-1-outbound]...| RTP: > (Audio) addr=192.168.1.112:30002->192.168.1.2:62876 xor=0 rx=3774 tx=0 > lost=0 outOfOrder=0 late=0 rxTime=19/47 txTime=0/0 jitter=38/39 > > From the first Log line you can see that the RTP media connection > between the OpenSBC and the Softphone is not working, The connection is > going to 0.0.0.0:0. I wonder if this is because I have no NAT configured > on the Server and I just have the OpenSBC connected to each subnet by an > Interface? Could that be the reason? I have no NAT configured at all. > > The second log makes more sense and thats the RTP media connection with > the Asterisk Box, It is able to successfully connect, if I check the > asterisk, the connection appear to be normal, although there is no > evidence of sound. > > Hope This Helps > > Gerardo Amaya > > Joegen E. Baclor wrote: > |
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From: Gerardo A. <ope...@ge...> - 2007-01-12 22:00:53
|
Hello All. Now regarding the sound, I just found something really interesting in the Logs, take a look at this: 2007/01/12 12:13:18.770 0x83463e8 OSS DTL: [YTA2OGI0NzNkYTljMWUwYjBlZmQ2ZmI5ODE1YjgyNjI.-inbound]......| RTP: (Audio) addr=10.0.0.2:30000->0.0.0.0:0 xor=0 rx=0 tx=0 lost=0 outOfOrder=0 late=0 rxTime=0/0 txTime=0/0 jitter=0/0 2007/01/12 12:13:18.770 0x83463e8 OSS DTL: [YTA2OGI0NzNkYTljMWUwYjBlZmQ2ZmI5ODE1YjgyNjI.-1-outbound]...| RTP: (Audio) addr=192.168.1.112:30002->192.168.1.2:62876 xor=0 rx=3774 tx=0 lost=0 outOfOrder=0 late=0 rxTime=19/47 txTime=0/0 jitter=38/39 From the first Log line you can see that the RTP media connection between the OpenSBC and the Softphone is not working, The connection is going to 0.0.0.0:0. I wonder if this is because I have no NAT configured on the Server and I just have the OpenSBC connected to each subnet by an Interface? Could that be the reason? I have no NAT configured at all. The second log makes more sense and thats the RTP media connection with the Asterisk Box, It is able to successfully connect, if I check the asterisk, the connection appear to be normal, although there is no evidence of sound. Hope This Helps Gerardo Amaya Joegen E. Baclor wrote: > Hi Gerardo and Natambu, > > Ok you got me there. Try the latest CVS once again ;-) > > When you REGISTER to OSBC make sure that the TO URI of the REGISTER > request corresponds to the IP or the domain of you asterisk box. > > For example: > > [sip:*10.0.0.2*] sip:192.168.1.2:5060 > > the REGISTER request To URI should be To: > sip:somebody@10.0.0.2:5060;tag=123456 > > also take note of this entry > > Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060 > > You must remove the ":" from > > [sip:*@meta1.fasttrackcomm.net:*] > > This because this would effectively filter out registration with URI not having a port. > > If you still get into trouble please send me a copy of the b2bua logs + your configuration ini. > > Thanks > > Joegen > > > Gerardo Amaya wrote: > >> I agree I just updated to the latest CVS Version and the same results, >> no registration and no audio whatsoever. What are we doing wrong? >> >> Here is my opensbc.ini file content as well: >> >> [OpenSBC General Parameters] >> Active Sessions=0 >> Log Level=5 >> Log File=b2bua >> Syslog Server=127.0.0.1 >> Transaction Thread Count=10 >> Session Thread Count=10 >> 1xx Timeout To Invite=5000 >> Connection Timeout=60000 >> Alerting Timeout=30000 >> Final Response Timeout To Invite=60000 >> Application Log Level=5 >> SIP and RTP Log Level=5 >> SBC Mode=B2BUpperReg Mode >> Always Proxy Media=True >> Application Log File=b2bua >> SIP Session Timer=False >> Hash Key 1=0 >> Hash Key 2=0 >> >> [OpenSBC Configuration Page] >> HTTP User=admin >> HTTP Password=zhvzOdSghXTf1TitVdenjQ== >> Log Level=5 >> >> [OpenSBC Routes] >> [OpenSIPStack Application Configuration Page] >> HTTP User=admin >> HTTP Password=zhvzOdSghXRTMDct6GPfPw== >> Log Level=5 >> >> [Relay Routes] >> Route Array Size=1 >> Route 1=[sip:*10.0.0.2*] sip:192.168.1.2:5060 >> >> [Upper Registration Routes] >> Route Array Size=1 >> Route 1=[sip:*10.0.0.2*] sip:192.168.1.2:5060 >> >> [Local Domain Accounts] >> Account Array Size=0 >> Accept All Registration=False >> >> [OpenSBC HTTP Admin] >> HTTP User=admin >> HTTP Password=zhvzOdSghXRaS6a/FrQbGA== >> >> >> Rewrite TO URI=False >> Route Array Size=1 >> Route 1=[sip:*10.0.0.2*] sip:192.168.1.2:5060 >> >> >> Best Regards >> >> Gerardo Amaya >> >> Natambu Obleton wrote: >> >> >>> Hello, >>> >>> I follow the setup you posted here >>> >>> https://sourceforge.net/mailarchive/message.php?msg_id=37882035 >>> >>> But it isn’t working. I looked in /root/.pwlib_config/opensbc.ini and >>> it looks like this.. >>> >>> [root@laplata obj_linux_x86_r]# cat /root/.pwlib_config/opensbc.ini >>> >>> [OpenSBC General Parameters] >>> >>> Active Sessions=0 >>> >>> Application Log Level=5 >>> >>> SIP and RTP Log Level=3 >>> >>> SBC Mode=B2BUpperReg Mode >>> >>> Always Proxy Media=True >>> >>> Application Log File=b2bua >>> >>> Syslog Server=127.0.0.1 >>> >>> SIP Session Timer=False >>> >>> Hash Key 1=0 >>> >>> Hash Key 2=0 >>> >>> Transaction Thread Count=10 >>> >>> Session Thread Count=10 >>> >>> 1xx Timeout To Invite=5000 >>> >>> Connection Timeout=60000 >>> >>> Alerting Timeout=30000 >>> >>> Final Response Timeout To Invite=60000 >>> >>> [Upper Registration Routes] >>> >>> Route Array Size=1 >>> >>> Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060 >>> >>> [OpenSBC Routes] >>> >>> Rewrite TO URI=False >>> >>> Route Array Size=1 >>> >>> Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060 >>> >>> [Relay Routes] >>> >>> Route Array Size=1 >>> >>> Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060 >>> >>> [root@laplata obj_linux_x86_r]# >>> >>> But when I start opensbc it says it is still in “Full Mode” and not >>> “B2BUpperReg Mode” .. >>> >>> [root@laplata obj_linux_x86_r]# ./opensbc -x -c >>> >>> All output for OpenSBC is to console. >>> >>> 2007/01/11 10:37:41.431 OpenSBC Message Starting service process >>> "OpenSBC" v1.1.4 >>> >>> 2007/01/11 10:37:41.433 OpenSBC Debug3 OpenSBC Process is starting >>> >>> 2007/01/11 10:37:41.457 OpenSBC Debug3 Running in Full Mode >>> >>> 2007/01/11 10:37:41.462 OpenSBC Debug3 OpenSBC STARTED >>> >>> 2007/01/11 10:37:41.464 OpenSBC Debug3 Configuration change detected >>> >>> 2007/01/11 10:37:41.466 OpenSBC Debug3 OpalMan Created manager. >>> >>> 2007/01/11 10:37:41.470 OpenSBC Debug3 Found 4 interfaces >>> >>> Natambu Obleton >>> >>> Network Engineer >>> >>> FastTrack Communications >>> >>> nob...@fa... <mailto:nob...@fa...> >>> >>> (970) 247-3366 office >>> >>> (970) 247-2426 fax >>> >>> ------------------------------------------------------------------------ >>> >>> ------------------------------------------------------------------------- >>> Take Surveys. Earn Cash. Influence the Future of IT >>> Join SourceForge.net's Techsay panel and you'll get the chance to share your >>> opinions on IT & business topics through brief surveys - and earn cash >>> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> opensipstack-devel mailing list >>> ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >>> >>> >>> >> >> ------------------------------------------------------------------------- >> Take Surveys. Earn Cash. Influence the Future of IT >> Join SourceForge.net's Techsay panel and you'll get the chance to share your >> opinions on IT & business topics through brief surveys - and earn cash >> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> > > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
|
From: Gerardo A. <ope...@ge...> - 2007-01-12 21:51:52
|
Hello All. Just for testing purposes I forced the B2BUpper Reg mode in
the code like this:
if( mode.IsEmpty() )
config.GetString( configKeySection, configKeySBCMode, "B2BUpperReg
Mode" );
B2BUserAgent::UAMode uaMode = B2BUserAgent::B2BUpperRegMode;
In the log file, it seems that the Mode is working:
2007/01/12 15:37:42.147 OpenSBC Message Starting service
process "OpenSBC" v1.1.4
2007/01/12 15:37:42.148 OpenSBC Debug3 OpenSBC Process
is starting
2007/01/12 15:37:42.158 OpenSBC Debug3 Running in Back
To Back with Upper Registration Mode
2007/01/12 15:37:42.159 OpenSBC Debug3 OpenSBC STARTED
2007/01/12 15:37:42.160 OpenSBC Debug3 Opened master
socket for HTTP: 9999
2007/01/12 15:37:42.160 OnConfigChanged Debug3 Configuration
change detected
2007/01/12 15:37:42.161 OnConfigChanged Debug3 OpalMan Created
manager.
For some reason the mode parameter is empty, Now the bad news is that I
ran the server like that and the Registration still not working, here
are the results of the registration process. Just a reminder that I want
to register my softphone like this
SoftPhone|10.0.0.3----------10.0.0.2|OpenSBC|192.168.1.112-------192.168.1.2|Asterisk
Here are the logs from the Registration Process
==========================================================================================================================================
2007/01/12 15:48:12.331 Transport( READ ) Debug3 RCV: XOR=0 514
Bytes from RCVADDR: 10.0.0.3:RCVPORT: 5052:UDP (REGISTER sip:10.0.0.2
SIP/2.0)
REGISTER sip:10.0.0.2 SIP/2.0
From: "test" <sip:test@10.0.0.2>;tag=1c7cfc45
To: "test" <sip:test@10.0.0.2>
Via: SIP/2.0/UDP
10.0.0.3:5052;branch=z9hG4bK-d87543-073cbf101c05244f-1--d87543-
CSeq: 1 REGISTER
Call-ID: MzkxNjAzYzI5NTgyYWNhMDhlNTY3YzkzOTFiYmVkMDI.
Contact: <sip:test@10.0.0.3:5052;rinstance=c38cb44158e52847>
User-Agent: X-Lite release 1006e stamp 34025
Expires: 3600
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Length: 0
2007/01/12 15:48:12.336 Transport( WRITE ) Debug3 SEND: XOR=0 496
Bytes to 10.0.0.3:5052:UDP (SIP/2.0 401 Unauthorized) Interface
Address=10.0.0.2
SIP/2.0 401 Unauthorized
From: "test" <sip:test@10.0.0.2>;tag=1c7cfc45
To: "test" <sip:test@10.0.0.2>;tag=d226be44f4a0db118411916496ddc768
Via: SIP/2.0/UDP
10.0.0.3:5052;iid=1;branch=z9hG4bK-d87543-073cbf101c05244f-1--d87543-;rport=5052;received=10.0.0.3
CSeq: 1 REGISTER
Call-ID: MzkxNjAzYzI5NTgyYWNhMDhlNTY3YzkzOTFiYmVkMDI.
WWW-Authenticate: Digest realm="10.0.0.2",
nonce="ca28482a10e7b827f46711063d0d0efa",
opaque="0c842362a4cd72447dac636ed6a4da99", algorithm=MD5
Content-Length: 0
2007/01/12 15:48:12.337 Transport( WRITE ) Debug3 Using Iface:
10.0.0.2 to send to Dest: 10.0.0.3
2007/01/12 15:48:12.440 Transport( READ ) Debug3 RCV: XOR=0 730
Bytes from RCVADDR: 10.0.0.3:RCVPORT: 5052:UDP (REGISTER sip:10.0.0.2
SIP/2.0)
REGISTER sip:10.0.0.2 SIP/2.0
From: "test" <sip:test@10.0.0.2>;tag=1c7cfc45
To: "test" <sip:test@10.0.0.2>
Via: SIP/2.0/UDP
10.0.0.3:5052;branch=z9hG4bK-d87543-9d3b6e3286304b18-1--d87543-
CSeq: 2 REGISTER
Call-ID: MzkxNjAzYzI5NTgyYWNhMDhlNTY3YzkzOTFiYmVkMDI.
Contact: <sip:test@10.0.0.3:5052;rinstance=c38cb44158e52847>
User-Agent: X-Lite release 1006e stamp 34025
Expires: 3600
Max-Forwards: 70
Authorization: Digest username="test", realm="10.0.0.2",
nonce="ca28482a10e7b827f46711063d0d0efa", uri="sip:10.0.0.2",
response="c2d55e2f676d6b86c37da9b8860349ca", algorithm=MD5,
opaque="0c842362a4cd72447dac636ed6a4da99"
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Content-Length: 0
2007/01/12 15:48:12.443 Registrar Debug3 Registration
rejected for sip:test@10.0.0.2
2007/01/12 15:48:12.445 Transport( WRITE ) Debug3 SEND: XOR=0 350
Bytes to 10.0.0.3:5052:UDP (SIP/2.0 403 Forbidden) Interface
Address=10.0.0.2
SIP/2.0 403 Forbidden
From: "test" <sip:test@10.0.0.2>;tag=1c7cfc45
To: "test" <sip:test@10.0.0.2>;tag=e6cace44f4a0db118411916496ddc768
Via: SIP/2.0/UDP
10.0.0.3:5052;iid=1;branch=z9hG4bK-d87543-9d3b6e3286304b18-1--d87543-;rport=5052;received=10.0.0.3
CSeq: 2 REGISTER
Call-ID: MzkxNjAzYzI5NTgyYWNhMDhlNTY3YzkzOTFiYmVkMDI.
Content-Length: 0
2007/01/12 15:48:12.446 Transport( WRITE ) Debug3 Using Iface:
10.0.0.2 to send to Dest: 10.0.0.3
==========================================================================================================================================
Best Regards
Gerardo Amaya
Natambu Obleton wrote:
> Thanks for working on this, but I get the same result as before. It only
> starts in "Full Mode". I have looked at the code and added some debugging
> and it seems that mode is empty.
>
> if( mode.IsEmpty() ) {
> config.GetString( configKeySection, configKeySBCMode, "Full Mode" );
> PTRACE( 1, "mode is Empty" );
> }
>
>
> [root@laplata obj_linux_x86_r]# ./opensbc -x -c
> All output for OpenSBC is to console.
> 2007/01/12 08:26:09.212 OpenSBC Message Starting service
> process "OpenSBC" v1.1.4
> 2007/01/12 08:26:09.213 OpenSBC Debug3 OpenSBC Process is
> starting
> 2007/01/12 08:26:09.226 OpenSBC Debug3 mode is Empty
> 2007/01/12 08:26:09.229 OpenSBC Debug3 Running in Full Mode
> 2007/01/12 08:26:09.234 OpenSBC Debug3 OpenSBC STARTED
> 2007/01/12 08:26:09.236 OnConfigChanged Debug3 Configuration change
> detected
> 2007/01/12 08:26:09.238 OnConfigChanged Debug3 OpalMan Created
> manager.
> 2007/01/12 08:26:09.242 OnConfigChanged Debug3 Found 4 interfaces
>
>
>
> Natambu Obleton
> Network Engineer
> FastTrack Communications
> nob...@fa...
> (970) 247-3366 office
> (970) 247-2426 fax
>
>
>> -----Original Message-----
>> From: ope...@li...
>> [mailto:ope...@li...] On Behalf Of
>> Joegen E. Baclor
>> Sent: Friday, January 12, 2007 3:56 AM
>> To: ope...@li...
>> Subject: Re: [OpenSIPStack] OpenSBC REGISTER not Working and No Sound
>>
>> Hi Gerardo and Natambu,
>>
>> Ok you got me there. Try the latest CVS once again ;-)
>>
>> When you REGISTER to OSBC make sure that the TO URI of the REGISTER
>> request corresponds to the IP or the domain of you asterisk box.
>>
>> For example:
>>
>> [sip:*10.0.0.2*] sip:192.168.1.2:5060
>>
>> the REGISTER request To URI should be To:
>> sip:somebody@10.0.0.2:5060;tag=123456
>>
>> also take note of this entry
>>
>> Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060
>>
>> You must remove the ":" from
>>
>> [sip:*@meta1.fasttrackcomm.net:*]
>>
>> This because this would effectively filter out registration with URI not
>> having a port.
>>
>> If you still get into trouble please send me a copy of the b2bua logs +
>> your configuration ini.
>>
>> Thanks
>>
>> Joegen
>>
>>
>> Gerardo Amaya wrote:
>>
>>> I agree I just updated to the latest CVS Version and the same results,
>>> no registration and no audio whatsoever. What are we doing wrong?
>>>
>>> Here is my opensbc.ini file content as well:
>>>
>>> [OpenSBC General Parameters]
>>> Active Sessions=0
>>> Log Level=5
>>> Log File=b2bua
>>> Syslog Server=127.0.0.1
>>> Transaction Thread Count=10
>>> Session Thread Count=10
>>> 1xx Timeout To Invite=5000
>>> Connection Timeout=60000
>>> Alerting Timeout=30000
>>> Final Response Timeout To Invite=60000
>>> Application Log Level=5
>>> SIP and RTP Log Level=5
>>> SBC Mode=B2BUpperReg Mode
>>> Always Proxy Media=True
>>> Application Log File=b2bua
>>> SIP Session Timer=False
>>> Hash Key 1=0
>>> Hash Key 2=0
>>>
>>> [OpenSBC Configuration Page]
>>> HTTP User=admin
>>> HTTP Password=zhvzOdSghXTf1TitVdenjQ==
>>> Log Level=5
>>>
>>> [OpenSBC Routes]
>>> [OpenSIPStack Application Configuration Page]
>>> HTTP User=admin
>>> HTTP Password=zhvzOdSghXRTMDct6GPfPw==
>>> Log Level=5
>>>
>>> [Relay Routes]
>>> Route Array Size=1
>>> Route 1=[sip:*10.0.0.2*] sip:192.168.1.2:5060
>>>
>>> [Upper Registration Routes]
>>> Route Array Size=1
>>> Route 1=[sip:*10.0.0.2*] sip:192.168.1.2:5060
>>>
>>> [Local Domain Accounts]
>>> Account Array Size=0
>>> Accept All Registration=False
>>>
>>> [OpenSBC HTTP Admin]
>>> HTTP User=admin
>>> HTTP Password=zhvzOdSghXRaS6a/FrQbGA==
>>>
>>>
>>> Rewrite TO URI=False
>>> Route Array Size=1
>>> Route 1=[sip:*10.0.0.2*] sip:192.168.1.2:5060
>>>
>>>
>>> Best Regards
>>>
>>> Gerardo Amaya
>>>
>>> Natambu Obleton wrote:
>>>
>>>
>>>> Hello,
>>>>
>>>> I follow the setup you posted here
>>>>
>>>> https://sourceforge.net/mailarchive/message.php?msg_id=37882035
>>>>
>>>> But it isn't working. I looked in /root/.pwlib_config/opensbc.ini and
>>>> it looks like this..
>>>>
>>>> [root@laplata obj_linux_x86_r]# cat /root/.pwlib_config/opensbc.ini
>>>>
>>>> [OpenSBC General Parameters]
>>>>
>>>> Active Sessions=0
>>>>
>>>> Application Log Level=5
>>>>
>>>> SIP and RTP Log Level=3
>>>>
>>>> SBC Mode=B2BUpperReg Mode
>>>>
>>>> Always Proxy Media=True
>>>>
>>>> Application Log File=b2bua
>>>>
>>>> Syslog Server=127.0.0.1
>>>>
>>>> SIP Session Timer=False
>>>>
>>>> Hash Key 1=0
>>>>
>>>> Hash Key 2=0
>>>>
>>>> Transaction Thread Count=10
>>>>
>>>> Session Thread Count=10
>>>>
>>>> 1xx Timeout To Invite=5000
>>>>
>>>> Connection Timeout=60000
>>>>
>>>> Alerting Timeout=30000
>>>>
>>>> Final Response Timeout To Invite=60000
>>>>
>>>> [Upper Registration Routes]
>>>>
>>>> Route Array Size=1
>>>>
>>>> Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060
>>>>
>>>> [OpenSBC Routes]
>>>>
>>>> Rewrite TO URI=False
>>>>
>>>> Route Array Size=1
>>>>
>>>> Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060
>>>>
>>>> [Relay Routes]
>>>>
>>>> Route Array Size=1
>>>>
>>>> Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060
>>>>
>>>> [root@laplata obj_linux_x86_r]#
>>>>
>>>> But when I start opensbc it says it is still in "Full Mode" and not
>>>> "B2BUpperReg Mode" ..
>>>>
>>>> [root@laplata obj_linux_x86_r]# ./opensbc -x -c
>>>>
>>>> All output for OpenSBC is to console.
>>>>
>>>> 2007/01/11 10:37:41.431 OpenSBC Message Starting service process
>>>> "OpenSBC" v1.1.4
>>>>
>>>> 2007/01/11 10:37:41.433 OpenSBC Debug3 OpenSBC Process is starting
>>>>
>>>> 2007/01/11 10:37:41.457 OpenSBC Debug3 Running in Full Mode
>>>>
>>>> 2007/01/11 10:37:41.462 OpenSBC Debug3 OpenSBC STARTED
>>>>
>>>> 2007/01/11 10:37:41.464 OpenSBC Debug3 Configuration change detected
>>>>
>>>> 2007/01/11 10:37:41.466 OpenSBC Debug3 OpalMan Created manager.
>>>>
>>>> 2007/01/11 10:37:41.470 OpenSBC Debug3 Found 4 interfaces
>>>>
>>>> Natambu Obleton
>>>>
>>>> Network Engineer
>>>>
>>>> FastTrack Communications
>>>>
>>>> nob...@fa... <mailto:nob...@fa...>
>>>>
>>>> (970) 247-3366 office
>>>>
>>>> (970) 247-2426 fax
>>>>
>>>> -----------------------------------------------------------------------
>>>>
>> -
>>
>>>> -----------------------------------------------------------------------
>>>>
>> --
>>
>>>> Take Surveys. Earn Cash. Influence the Future of IT
>>>> Join SourceForge.net's Techsay panel and you'll get the chance to share
>>>>
>> your
>>
>>>> opinions on IT & business topics through brief surveys - and earn cash
>>>>
>>>>
>> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV
>>
>>>> -----------------------------------------------------------------------
>>>>
>> -
>>
>>>> _______________________________________________
>>>> opensipstack-devel mailing list
>>>> ope...@li...
>>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>>>
>>>>
>>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>> -
>>
>>> Take Surveys. Earn Cash. Influence the Future of IT
>>> Join SourceForge.net's Techsay panel and you'll get the chance to share
>>>
>> your
>>
>>> opinions on IT & business topics through brief surveys - and earn cash
>>>
>>>
>> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV
>>
>>> _______________________________________________
>>> opensipstack-devel mailing list
>>> ope...@li...
>>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>>
>>>
>>>
>>>
>> -------------------------------------------------------------------------
>> Take Surveys. Earn Cash. Influence the Future of IT
>> Join SourceForge.net's Techsay panel and you'll get the chance to share
>> your
>> opinions on IT & business topics through brief surveys - and earn cash
>> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV
>> _______________________________________________
>> opensipstack-devel mailing list
>> ope...@li...
>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>
>
>
> -------------------------------------------------------------------------
> Take Surveys. Earn Cash. Influence the Future of IT
> Join SourceForge.net's Techsay panel and you'll get the chance to share your
> opinions on IT & business topics through brief surveys - and earn cash
> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>
|
|
From: Natambu O. <nob...@fa...> - 2007-01-12 16:26:07
|
Thanks for working on this, but I get the same result as before. It only
starts in "Full Mode". I have looked at the code and added some debugging
and it seems that mode is empty.
if( mode.IsEmpty() ) {
config.GetString( configKeySection, configKeySBCMode, "Full Mode" );
PTRACE( 1, "mode is Empty" );
}
[root@laplata obj_linux_x86_r]# ./opensbc -x -c
All output for OpenSBC is to console.
2007/01/12 08:26:09.212 OpenSBC Message Starting service
process "OpenSBC" v1.1.4
2007/01/12 08:26:09.213 OpenSBC Debug3 OpenSBC Process is
starting
2007/01/12 08:26:09.226 OpenSBC Debug3 mode is Empty
2007/01/12 08:26:09.229 OpenSBC Debug3 Running in Full Mode
2007/01/12 08:26:09.234 OpenSBC Debug3 OpenSBC STARTED
2007/01/12 08:26:09.236 OnConfigChanged Debug3 Configuration change
detected
2007/01/12 08:26:09.238 OnConfigChanged Debug3 OpalMan Created
manager.
2007/01/12 08:26:09.242 OnConfigChanged Debug3 Found 4 interfaces
Natambu Obleton
Network Engineer
FastTrack Communications
nob...@fa...
(970) 247-3366 office
(970) 247-2426 fax
> -----Original Message-----
> From: ope...@li...
> [mailto:ope...@li...] On Behalf Of
> Joegen E. Baclor
> Sent: Friday, January 12, 2007 3:56 AM
> To: ope...@li...
> Subject: Re: [OpenSIPStack] OpenSBC REGISTER not Working and No Sound
>
> Hi Gerardo and Natambu,
>
> Ok you got me there. Try the latest CVS once again ;-)
>
> When you REGISTER to OSBC make sure that the TO URI of the REGISTER
> request corresponds to the IP or the domain of you asterisk box.
>
> For example:
>
> [sip:*10.0.0.2*] sip:192.168.1.2:5060
>
> the REGISTER request To URI should be To:
> sip:somebody@10.0.0.2:5060;tag=123456
>
> also take note of this entry
>
> Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060
>
> You must remove the ":" from
>
> [sip:*@meta1.fasttrackcomm.net:*]
>
> This because this would effectively filter out registration with URI not
> having a port.
>
> If you still get into trouble please send me a copy of the b2bua logs +
> your configuration ini.
>
> Thanks
>
> Joegen
>
>
> Gerardo Amaya wrote:
> > I agree I just updated to the latest CVS Version and the same results,
> > no registration and no audio whatsoever. What are we doing wrong?
> >
> > Here is my opensbc.ini file content as well:
> >
> > [OpenSBC General Parameters]
> > Active Sessions=0
> > Log Level=5
> > Log File=b2bua
> > Syslog Server=127.0.0.1
> > Transaction Thread Count=10
> > Session Thread Count=10
> > 1xx Timeout To Invite=5000
> > Connection Timeout=60000
> > Alerting Timeout=30000
> > Final Response Timeout To Invite=60000
> > Application Log Level=5
> > SIP and RTP Log Level=5
> > SBC Mode=B2BUpperReg Mode
> > Always Proxy Media=True
> > Application Log File=b2bua
> > SIP Session Timer=False
> > Hash Key 1=0
> > Hash Key 2=0
> >
> > [OpenSBC Configuration Page]
> > HTTP User=admin
> > HTTP Password=zhvzOdSghXTf1TitVdenjQ==
> > Log Level=5
> >
> > [OpenSBC Routes]
> > [OpenSIPStack Application Configuration Page]
> > HTTP User=admin
> > HTTP Password=zhvzOdSghXRTMDct6GPfPw==
> > Log Level=5
> >
> > [Relay Routes]
> > Route Array Size=1
> > Route 1=[sip:*10.0.0.2*] sip:192.168.1.2:5060
> >
> > [Upper Registration Routes]
> > Route Array Size=1
> > Route 1=[sip:*10.0.0.2*] sip:192.168.1.2:5060
> >
> > [Local Domain Accounts]
> > Account Array Size=0
> > Accept All Registration=False
> >
> > [OpenSBC HTTP Admin]
> > HTTP User=admin
> > HTTP Password=zhvzOdSghXRaS6a/FrQbGA==
> >
> >
> > Rewrite TO URI=False
> > Route Array Size=1
> > Route 1=[sip:*10.0.0.2*] sip:192.168.1.2:5060
> >
> >
> > Best Regards
> >
> > Gerardo Amaya
> >
> > Natambu Obleton wrote:
> >
> >> Hello,
> >>
> >> I follow the setup you posted here
> >>
> >> https://sourceforge.net/mailarchive/message.php?msg_id=37882035
> >>
> >> But it isn't working. I looked in /root/.pwlib_config/opensbc.ini and
> >> it looks like this..
> >>
> >> [root@laplata obj_linux_x86_r]# cat /root/.pwlib_config/opensbc.ini
> >>
> >> [OpenSBC General Parameters]
> >>
> >> Active Sessions=0
> >>
> >> Application Log Level=5
> >>
> >> SIP and RTP Log Level=3
> >>
> >> SBC Mode=B2BUpperReg Mode
> >>
> >> Always Proxy Media=True
> >>
> >> Application Log File=b2bua
> >>
> >> Syslog Server=127.0.0.1
> >>
> >> SIP Session Timer=False
> >>
> >> Hash Key 1=0
> >>
> >> Hash Key 2=0
> >>
> >> Transaction Thread Count=10
> >>
> >> Session Thread Count=10
> >>
> >> 1xx Timeout To Invite=5000
> >>
> >> Connection Timeout=60000
> >>
> >> Alerting Timeout=30000
> >>
> >> Final Response Timeout To Invite=60000
> >>
> >> [Upper Registration Routes]
> >>
> >> Route Array Size=1
> >>
> >> Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060
> >>
> >> [OpenSBC Routes]
> >>
> >> Rewrite TO URI=False
> >>
> >> Route Array Size=1
> >>
> >> Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060
> >>
> >> [Relay Routes]
> >>
> >> Route Array Size=1
> >>
> >> Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060
> >>
> >> [root@laplata obj_linux_x86_r]#
> >>
> >> But when I start opensbc it says it is still in "Full Mode" and not
> >> "B2BUpperReg Mode" ..
> >>
> >> [root@laplata obj_linux_x86_r]# ./opensbc -x -c
> >>
> >> All output for OpenSBC is to console.
> >>
> >> 2007/01/11 10:37:41.431 OpenSBC Message Starting service process
> >> "OpenSBC" v1.1.4
> >>
> >> 2007/01/11 10:37:41.433 OpenSBC Debug3 OpenSBC Process is starting
> >>
> >> 2007/01/11 10:37:41.457 OpenSBC Debug3 Running in Full Mode
> >>
> >> 2007/01/11 10:37:41.462 OpenSBC Debug3 OpenSBC STARTED
> >>
> >> 2007/01/11 10:37:41.464 OpenSBC Debug3 Configuration change detected
> >>
> >> 2007/01/11 10:37:41.466 OpenSBC Debug3 OpalMan Created manager.
> >>
> >> 2007/01/11 10:37:41.470 OpenSBC Debug3 Found 4 interfaces
> >>
> >> Natambu Obleton
> >>
> >> Network Engineer
> >>
> >> FastTrack Communications
> >>
> >> nob...@fa... <mailto:nob...@fa...>
> >>
> >> (970) 247-3366 office
> >>
> >> (970) 247-2426 fax
> >>
> >> -----------------------------------------------------------------------
> -
> >>
> >> -----------------------------------------------------------------------
> --
> >> Take Surveys. Earn Cash. Influence the Future of IT
> >> Join SourceForge.net's Techsay panel and you'll get the chance to share
> your
> >> opinions on IT & business topics through brief surveys - and earn cash
> >>
> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV
> >> -----------------------------------------------------------------------
> -
> >>
> >> _______________________________________________
> >> opensipstack-devel mailing list
> >> ope...@li...
> >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> >>
> >>
> >
> >
> >
> > ------------------------------------------------------------------------
> -
> > Take Surveys. Earn Cash. Influence the Future of IT
> > Join SourceForge.net's Techsay panel and you'll get the chance to share
> your
> > opinions on IT & business topics through brief surveys - and earn cash
> >
> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV
> > _______________________________________________
> > opensipstack-devel mailing list
> > ope...@li...
> > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
> >
> >
> >
>
>
> -------------------------------------------------------------------------
> Take Surveys. Earn Cash. Influence the Future of IT
> Join SourceForge.net's Techsay panel and you'll get the chance to share
> your
> opinions on IT & business topics through brief surveys - and earn cash
> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV
> _______________________________________________
> opensipstack-devel mailing list
> ope...@li...
> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
|
|
From: Claudio M. <cmi...@gm...> - 2007-01-12 11:18:48
|
I started to look for the media stack implementation, but I got lost.
Which files correspond to that implementation ?
(Sorry I am starting to work with OpenSipStack). Can you help me ?
Thanks,
Claudio Miceli de Farias
|
|
From: Joegen E. B. <joe...@gm...> - 2007-01-12 10:56:31
|
Hi Gerardo and Natambu, Ok you got me there. Try the latest CVS once again ;-) When you REGISTER to OSBC make sure that the TO URI of the REGISTER request corresponds to the IP or the domain of you asterisk box. For example: [sip:*10.0.0.2*] sip:192.168.1.2:5060 the REGISTER request To URI should be To: sip:somebody@10.0.0.2:5060;tag=123456 also take note of this entry Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060 You must remove the ":" from [sip:*@meta1.fasttrackcomm.net:*] This because this would effectively filter out registration with URI not having a port. If you still get into trouble please send me a copy of the b2bua logs + your configuration ini. Thanks Joegen Gerardo Amaya wrote: > I agree I just updated to the latest CVS Version and the same results, > no registration and no audio whatsoever. What are we doing wrong? > > Here is my opensbc.ini file content as well: > > [OpenSBC General Parameters] > Active Sessions=0 > Log Level=5 > Log File=b2bua > Syslog Server=127.0.0.1 > Transaction Thread Count=10 > Session Thread Count=10 > 1xx Timeout To Invite=5000 > Connection Timeout=60000 > Alerting Timeout=30000 > Final Response Timeout To Invite=60000 > Application Log Level=5 > SIP and RTP Log Level=5 > SBC Mode=B2BUpperReg Mode > Always Proxy Media=True > Application Log File=b2bua > SIP Session Timer=False > Hash Key 1=0 > Hash Key 2=0 > > [OpenSBC Configuration Page] > HTTP User=admin > HTTP Password=zhvzOdSghXTf1TitVdenjQ== > Log Level=5 > > [OpenSBC Routes] > [OpenSIPStack Application Configuration Page] > HTTP User=admin > HTTP Password=zhvzOdSghXRTMDct6GPfPw== > Log Level=5 > > [Relay Routes] > Route Array Size=1 > Route 1=[sip:*10.0.0.2*] sip:192.168.1.2:5060 > > [Upper Registration Routes] > Route Array Size=1 > Route 1=[sip:*10.0.0.2*] sip:192.168.1.2:5060 > > [Local Domain Accounts] > Account Array Size=0 > Accept All Registration=False > > [OpenSBC HTTP Admin] > HTTP User=admin > HTTP Password=zhvzOdSghXRaS6a/FrQbGA== > > > Rewrite TO URI=False > Route Array Size=1 > Route 1=[sip:*10.0.0.2*] sip:192.168.1.2:5060 > > > Best Regards > > Gerardo Amaya > > Natambu Obleton wrote: > >> Hello, >> >> I follow the setup you posted here >> >> https://sourceforge.net/mailarchive/message.php?msg_id=37882035 >> >> But it isn’t working. I looked in /root/.pwlib_config/opensbc.ini and >> it looks like this.. >> >> [root@laplata obj_linux_x86_r]# cat /root/.pwlib_config/opensbc.ini >> >> [OpenSBC General Parameters] >> >> Active Sessions=0 >> >> Application Log Level=5 >> >> SIP and RTP Log Level=3 >> >> SBC Mode=B2BUpperReg Mode >> >> Always Proxy Media=True >> >> Application Log File=b2bua >> >> Syslog Server=127.0.0.1 >> >> SIP Session Timer=False >> >> Hash Key 1=0 >> >> Hash Key 2=0 >> >> Transaction Thread Count=10 >> >> Session Thread Count=10 >> >> 1xx Timeout To Invite=5000 >> >> Connection Timeout=60000 >> >> Alerting Timeout=30000 >> >> Final Response Timeout To Invite=60000 >> >> [Upper Registration Routes] >> >> Route Array Size=1 >> >> Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060 >> >> [OpenSBC Routes] >> >> Rewrite TO URI=False >> >> Route Array Size=1 >> >> Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060 >> >> [Relay Routes] >> >> Route Array Size=1 >> >> Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060 >> >> [root@laplata obj_linux_x86_r]# >> >> But when I start opensbc it says it is still in “Full Mode” and not >> “B2BUpperReg Mode” .. >> >> [root@laplata obj_linux_x86_r]# ./opensbc -x -c >> >> All output for OpenSBC is to console. >> >> 2007/01/11 10:37:41.431 OpenSBC Message Starting service process >> "OpenSBC" v1.1.4 >> >> 2007/01/11 10:37:41.433 OpenSBC Debug3 OpenSBC Process is starting >> >> 2007/01/11 10:37:41.457 OpenSBC Debug3 Running in Full Mode >> >> 2007/01/11 10:37:41.462 OpenSBC Debug3 OpenSBC STARTED >> >> 2007/01/11 10:37:41.464 OpenSBC Debug3 Configuration change detected >> >> 2007/01/11 10:37:41.466 OpenSBC Debug3 OpalMan Created manager. >> >> 2007/01/11 10:37:41.470 OpenSBC Debug3 Found 4 interfaces >> >> Natambu Obleton >> >> Network Engineer >> >> FastTrack Communications >> >> nob...@fa... <mailto:nob...@fa...> >> >> (970) 247-3366 office >> >> (970) 247-2426 fax >> >> ------------------------------------------------------------------------ >> >> ------------------------------------------------------------------------- >> Take Surveys. Earn Cash. Influence the Future of IT >> Join SourceForge.net's Techsay panel and you'll get the chance to share your >> opinions on IT & business topics through brief surveys - and earn cash >> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> > > > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > |
|
From: Gerardo A. <ope...@ge...> - 2007-01-11 20:36:18
|
I agree I just updated to the latest CVS Version and the same results, no registration and no audio whatsoever. What are we doing wrong? Here is my opensbc.ini file content as well: [OpenSBC General Parameters] Active Sessions=0 Log Level=5 Log File=b2bua Syslog Server=127.0.0.1 Transaction Thread Count=10 Session Thread Count=10 1xx Timeout To Invite=5000 Connection Timeout=60000 Alerting Timeout=30000 Final Response Timeout To Invite=60000 Application Log Level=5 SIP and RTP Log Level=5 SBC Mode=B2BUpperReg Mode Always Proxy Media=True Application Log File=b2bua SIP Session Timer=False Hash Key 1=0 Hash Key 2=0 [OpenSBC Configuration Page] HTTP User=admin HTTP Password=zhvzOdSghXTf1TitVdenjQ== Log Level=5 [OpenSBC Routes] [OpenSIPStack Application Configuration Page] HTTP User=admin HTTP Password=zhvzOdSghXRTMDct6GPfPw== Log Level=5 [Relay Routes] Route Array Size=1 Route 1=[sip:*10.0.0.2*] sip:192.168.1.2:5060 [Upper Registration Routes] Route Array Size=1 Route 1=[sip:*10.0.0.2*] sip:192.168.1.2:5060 [Local Domain Accounts] Account Array Size=0 Accept All Registration=False [OpenSBC HTTP Admin] HTTP User=admin HTTP Password=zhvzOdSghXRaS6a/FrQbGA== Rewrite TO URI=False Route Array Size=1 Route 1=[sip:*10.0.0.2*] sip:192.168.1.2:5060 Best Regards Gerardo Amaya Natambu Obleton wrote: > > Hello, > > I follow the setup you posted here > > https://sourceforge.net/mailarchive/message.php?msg_id=37882035 > > But it isn’t working. I looked in /root/.pwlib_config/opensbc.ini and > it looks like this.. > > [root@laplata obj_linux_x86_r]# cat /root/.pwlib_config/opensbc.ini > > [OpenSBC General Parameters] > > Active Sessions=0 > > Application Log Level=5 > > SIP and RTP Log Level=3 > > SBC Mode=B2BUpperReg Mode > > Always Proxy Media=True > > Application Log File=b2bua > > Syslog Server=127.0.0.1 > > SIP Session Timer=False > > Hash Key 1=0 > > Hash Key 2=0 > > Transaction Thread Count=10 > > Session Thread Count=10 > > 1xx Timeout To Invite=5000 > > Connection Timeout=60000 > > Alerting Timeout=30000 > > Final Response Timeout To Invite=60000 > > [Upper Registration Routes] > > Route Array Size=1 > > Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060 > > [OpenSBC Routes] > > Rewrite TO URI=False > > Route Array Size=1 > > Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060 > > [Relay Routes] > > Route Array Size=1 > > Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060 > > [root@laplata obj_linux_x86_r]# > > But when I start opensbc it says it is still in “Full Mode” and not > “B2BUpperReg Mode” .. > > [root@laplata obj_linux_x86_r]# ./opensbc -x -c > > All output for OpenSBC is to console. > > 2007/01/11 10:37:41.431 OpenSBC Message Starting service process > "OpenSBC" v1.1.4 > > 2007/01/11 10:37:41.433 OpenSBC Debug3 OpenSBC Process is starting > > 2007/01/11 10:37:41.457 OpenSBC Debug3 Running in Full Mode > > 2007/01/11 10:37:41.462 OpenSBC Debug3 OpenSBC STARTED > > 2007/01/11 10:37:41.464 OpenSBC Debug3 Configuration change detected > > 2007/01/11 10:37:41.466 OpenSBC Debug3 OpalMan Created manager. > > 2007/01/11 10:37:41.470 OpenSBC Debug3 Found 4 interfaces > > Natambu Obleton > > Network Engineer > > FastTrack Communications > > nob...@fa... <mailto:nob...@fa...> > > (970) 247-3366 office > > (970) 247-2426 fax > > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > ------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
|
From: Natambu O. <nob...@fa...> - 2007-01-11 18:37:08
|
Hello, I follow the setup you posted here https://sourceforge.net/mailarchive/message.php?msg_id=37882035 But it isn't working. I looked in /root/.pwlib_config/opensbc.ini and it looks like this.. [root@laplata obj_linux_x86_r]# cat /root/.pwlib_config/opensbc.ini [OpenSBC General Parameters] Active Sessions=0 Application Log Level=5 SIP and RTP Log Level=3 SBC Mode=B2BUpperReg Mode Always Proxy Media=True Application Log File=b2bua Syslog Server=127.0.0.1 SIP Session Timer=False Hash Key 1=0 Hash Key 2=0 Transaction Thread Count=10 Session Thread Count=10 1xx Timeout To Invite=5000 Connection Timeout=60000 Alerting Timeout=30000 Final Response Timeout To Invite=60000 [Upper Registration Routes] Route Array Size=1 Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060 [OpenSBC Routes] Rewrite TO URI=False Route Array Size=1 Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060 [Relay Routes] Route Array Size=1 Route 1=[sip:*@meta1.fasttrackcomm.net:*] sip:206.123.214.84:5060 [root@laplata obj_linux_x86_r]# But when I start opensbc it says it is still in "Full Mode" and not "B2BUpperReg Mode" .. [root@laplata obj_linux_x86_r]# ./opensbc -x -c All output for OpenSBC is to console. 2007/01/11 10:37:41.431 OpenSBC Message Starting service process "OpenSBC" v1.1.4 2007/01/11 10:37:41.433 OpenSBC Debug3 OpenSBC Process is starting 2007/01/11 10:37:41.457 OpenSBC Debug3 Running in Full Mode 2007/01/11 10:37:41.462 OpenSBC Debug3 OpenSBC STARTED 2007/01/11 10:37:41.464 OpenSBC Debug3 Configuration change detected 2007/01/11 10:37:41.466 OpenSBC Debug3 OpalMan Created manager. 2007/01/11 10:37:41.470 OpenSBC Debug3 Found 4 interfaces Natambu Obleton Network Engineer FastTrack Communications nob...@fa... (970) 247-3366 office (970) 247-2426 fax |
|
From: Gerardo A. <ope...@ge...> - 2007-01-11 15:07:37
|
Thank you so much for your response. I'll try it and let you know how it goes. A little off-topic of the thread. But I'm really excited about this opensbc and opensipstack project. It's the best project I've seen in the past time. I will really like to volunteer for help in the Documentation process so there will be more information available for the community. If you need help on this or in any other area of the project, please let me know. Best Regards Gerardo Amaya Joegen E. Baclor wrote: > Hi Gerardo, > > 1. Download the latest from CVS. Just this morning, there were > several improvements commited to opensbc. > 2. Recompile opensbc and opensipstack and make sure you use the latest > oss-application.conf.xml template. > 3. In "HTTP Admin:9999/OpenSBC General Paramaters", set the following > - SBC Mode = B2BUpperRegMode (This mode would allow you to forward > registrations to asterisk) > - Always Proxy Media = TRUE (This is required so that media is > proxied accross the multi interfaces) > - If you don't see the above parameters in your config, it means you > don't have the latest opensbc and opensipstack source. > 4. Restart OpenSBC for the mode change to take effect > 5. In "HTTP Admin:9999/Upper Registration Routes" add the following > - [sip:*domain*] sip:192.168.1.2:5060 -- where domain is either an > FQDN or host alias that asterisk would recognize as its own > 6. Repeat step five for "HTTP Admin:9999/Relay Routes" > 7. Repeat step five for "HTTP Admin:9999/OpenSBC Routes" > > Thats it. > > Joegen > > Gerardo Amaya wrote: > >> Hello All. I just started using OpenSBC. I have an Asterisk Box that is >> taking care of all the SIP Registration and Call Handling Process. This >> Asterisk Box is on the Network 192.168.1.0/24 and I have the OpenSBC >> Server 8 network cards that connect me from different networks(e.g. >> 10.0.0.0/24, 172.16.18.0/24, etc). I just started trying with a simple >> rule from one of the Networks I wish to connect. >> >> [sip:*@10.0.0.2] sip:192.168.1.2:5060 >> >> Since I still don't know what I'm doing I create the same route in >> OpenSBC, Relay and Upper Registration, just to make sure, Now when I >> tried to register to the Asterisk Box I get the following message: >> >> 2007/01/10 12:25:18.232 Registrar Debug3 Registration >> rejected for sip:test@10.0.0.2 >> 2007/01/10 12:31:21.835 Registrar Debug3 Registration >> rejected for sip:test@10.0.0.2 >> >> What do I need to do in order to get the Registration process working. >> >> RTP and Sound >> >> In order to test the RTP Capabilities of OpenSBC I do not register for a >> moment and just try to set a call(INVITE) the call is successfully >> created and the other softphone answers, but I don't get any sound at >> all. I see from the logs that there is RTP traffic but there is no >> sound. Can you please help me telling me what I'm doing wrong >> >> 2007/01/10 12:17:15.135 OpalMediaThread:83f2738 Debug3 RTP Receive >> statistics: packets=4525 octets=724000 lost=0 tooLate=0 order=0 >> avgTime=18 maxTime=47 minTime=0 jitter=37 maxJitter=45 >> 2007/01/10 12:17:17.049 OpalMediaThread:83f2738 Debug3 RTP Receive >> statistics: packets=4625 octets=740000 lost=0 tooLate=0 order=0 >> avgTime=19 maxTime=47 minTime=0 jitter=38 maxJitter=45 >> 2007/01/10 12:17:17.129 OpalMediaThread:83f2738 Debug3 RTP >> SentReceiverReport: ssrc=1888406904 fraction=0 lost=0 last_seq=37365 >> jitter=307 lsr=0 dlsr=0 >> 2007/01/10 12:17:17.129 OpalMediaThread:83f2738 Debug3 RTP Sending >> SDES: root@atel-router >> 2007/01/10 12:17:18.946 OpalMediaThread:83f2738 Debug3 RTP Receive >> statistics: packets=4725 octets=756000 lost=0 tooLate=0 order=0 >> avgTime=18 maxTime=48 minTime=0 jitter=37 maxJitter=45 >> 2007/01/10 12:17:20.173 Call Debug3 RTP Found >> existing session 1 >> 2007/01/10 12:17:20.173 Call Debug3 RTP_UDP Session >> 1, Shutting down read. >> 2007/01/10 12:17:20.173 OpalMediaThread:836cbb0 Debug3 RTP_UDP Session >> 1, Read shutdown. >> 2007/01/10 12:17:20.173 OpalMediaThread:836cbb0 Debug3 RTP_UDP Closing >> Media Stream >> 2007/01/10 12:17:20.173 Call Debug3 RTP_UDP Session >> 1, Shutting down write. >> 2007/01/10 12:17:20.173 Call Debug3 RTP Found >> existing session 1 >> 2007/01/10 12:17:20.173 Call Debug3 RTP_UDP Session >> 1, Shutting down read. >> 2007/01/10 12:17:20.173 Call Debug3 RTP_UDP Session >> 1, Shutting down write. >> 2007/01/10 12:17:20.174 OpalMediaThread:83f2738 Debug3 RTP_UDP Session >> 1, Read shutdown. >> 2007/01/10 12:17:20.174 OpalMediaThread:83f2738 Debug3 RTP_UDP Closing >> Media Stream >> 2007/01/10 12:17:20.175 Garbage Collector Debug3 RTP_UDP Session >> 1, Shutting down read. >> 2007/01/10 12:17:20.175 Garbage Collector Debug3 RTP_UDP Session >> 1, Shutting down write. >> 2007/01/10 12:17:20.175 Garbage Collector Debug3 RTP_UDP Session >> 1, Shutting down read. >> 2007/01/10 12:17:20.176 Garbage Collector Debug3 RTP_UDP Session >> 1, Shutting down write. >> 2007/01/10 12:17:20.176 Garbage Collector Debug3 RTP Final >> statistics: >> packetsSent = 0 >> octetsSent = 0 >> averageSendTime = 0 >> maximumSendTime = 0 >> minimumSendTime = 0 >> packetsReceived = 4790 >> octetsReceived = 766400 >> packetsLost = 0 >> packetsTooLate = 0 >> packetsOutOfOrder = 0 >> averageReceiveTime= 18 >> maximumReceiveTime= 48 >> minimumReceiveTime= 0 >> averageJitter = 37 >> >> >> Thanks in advance for all the help. >> >> Gerardo Amaya >> >> >> ------------------------------------------------------------------------- >> Take Surveys. Earn Cash. Influence the Future of IT >> Join SourceForge.net's Techsay panel and you'll get the chance to share your >> opinions on IT & business topics through brief surveys - and earn cash >> http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV >> _______________________________________________ >> opensipstack-devel mailing list >> ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel >> >> >> >> > > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
|
From: Joegen E. B. <joe...@gm...> - 2007-01-11 07:38:29
|
Hi Gerardo,
1. Download the latest from CVS. Just this morning, there were
several improvements commited to opensbc.
2. Recompile opensbc and opensipstack and make sure you use the latest
oss-application.conf.xml template.
3. In "HTTP Admin:9999/OpenSBC General Paramaters", set the following
- SBC Mode = B2BUpperRegMode (This mode would allow you to forward
registrations to asterisk)
- Always Proxy Media = TRUE (This is required so that media is
proxied accross the multi interfaces)
- If you don't see the above parameters in your config, it means you
don't have the latest opensbc and opensipstack source.
4. Restart OpenSBC for the mode change to take effect
5. In "HTTP Admin:9999/Upper Registration Routes" add the following
- [sip:*domain*] sip:192.168.1.2:5060 -- where domain is either an
FQDN or host alias that asterisk would recognize as its own
6. Repeat step five for "HTTP Admin:9999/Relay Routes"
7. Repeat step five for "HTTP Admin:9999/OpenSBC Routes"
Thats it.
Joegen
Gerardo Amaya wrote:
> Hello All. I just started using OpenSBC. I have an Asterisk Box that is
> taking care of all the SIP Registration and Call Handling Process. This
> Asterisk Box is on the Network 192.168.1.0/24 and I have the OpenSBC
> Server 8 network cards that connect me from different networks(e.g.
> 10.0.0.0/24, 172.16.18.0/24, etc). I just started trying with a simple
> rule from one of the Networks I wish to connect.
>
> [sip:*@10.0.0.2] sip:192.168.1.2:5060
>
> Since I still don't know what I'm doing I create the same route in
> OpenSBC, Relay and Upper Registration, just to make sure, Now when I
> tried to register to the Asterisk Box I get the following message:
>
> 2007/01/10 12:25:18.232 Registrar Debug3 Registration
> rejected for sip:test@10.0.0.2
> 2007/01/10 12:31:21.835 Registrar Debug3 Registration
> rejected for sip:test@10.0.0.2
>
> What do I need to do in order to get the Registration process working.
>
> RTP and Sound
>
> In order to test the RTP Capabilities of OpenSBC I do not register for a
> moment and just try to set a call(INVITE) the call is successfully
> created and the other softphone answers, but I don't get any sound at
> all. I see from the logs that there is RTP traffic but there is no
> sound. Can you please help me telling me what I'm doing wrong
>
> 2007/01/10 12:17:15.135 OpalMediaThread:83f2738 Debug3 RTP Receive
> statistics: packets=4525 octets=724000 lost=0 tooLate=0 order=0
> avgTime=18 maxTime=47 minTime=0 jitter=37 maxJitter=45
> 2007/01/10 12:17:17.049 OpalMediaThread:83f2738 Debug3 RTP Receive
> statistics: packets=4625 octets=740000 lost=0 tooLate=0 order=0
> avgTime=19 maxTime=47 minTime=0 jitter=38 maxJitter=45
> 2007/01/10 12:17:17.129 OpalMediaThread:83f2738 Debug3 RTP
> SentReceiverReport: ssrc=1888406904 fraction=0 lost=0 last_seq=37365
> jitter=307 lsr=0 dlsr=0
> 2007/01/10 12:17:17.129 OpalMediaThread:83f2738 Debug3 RTP Sending
> SDES: root@atel-router
> 2007/01/10 12:17:18.946 OpalMediaThread:83f2738 Debug3 RTP Receive
> statistics: packets=4725 octets=756000 lost=0 tooLate=0 order=0
> avgTime=18 maxTime=48 minTime=0 jitter=37 maxJitter=45
> 2007/01/10 12:17:20.173 Call Debug3 RTP Found
> existing session 1
> 2007/01/10 12:17:20.173 Call Debug3 RTP_UDP Session
> 1, Shutting down read.
> 2007/01/10 12:17:20.173 OpalMediaThread:836cbb0 Debug3 RTP_UDP Session
> 1, Read shutdown.
> 2007/01/10 12:17:20.173 OpalMediaThread:836cbb0 Debug3 RTP_UDP Closing
> Media Stream
> 2007/01/10 12:17:20.173 Call Debug3 RTP_UDP Session
> 1, Shutting down write.
> 2007/01/10 12:17:20.173 Call Debug3 RTP Found
> existing session 1
> 2007/01/10 12:17:20.173 Call Debug3 RTP_UDP Session
> 1, Shutting down read.
> 2007/01/10 12:17:20.173 Call Debug3 RTP_UDP Session
> 1, Shutting down write.
> 2007/01/10 12:17:20.174 OpalMediaThread:83f2738 Debug3 RTP_UDP Session
> 1, Read shutdown.
> 2007/01/10 12:17:20.174 OpalMediaThread:83f2738 Debug3 RTP_UDP Closing
> Media Stream
> 2007/01/10 12:17:20.175 Garbage Collector Debug3 RTP_UDP Session
> 1, Shutting down read.
> 2007/01/10 12:17:20.175 Garbage Collector Debug3 RTP_UDP Session
> 1, Shutting down write.
> 2007/01/10 12:17:20.175 Garbage Collector Debug3 RTP_UDP Session
> 1, Shutting down read.
> 2007/01/10 12:17:20.176 Garbage Collector Debug3 RTP_UDP Session
> 1, Shutting down write.
> 2007/01/10 12:17:20.176 Garbage Collector Debug3 RTP Final
> statistics:
> packetsSent = 0
> octetsSent = 0
> averageSendTime = 0
> maximumSendTime = 0
> minimumSendTime = 0
> packetsReceived = 4790
> octetsReceived = 766400
> packetsLost = 0
> packetsTooLate = 0
> packetsOutOfOrder = 0
> averageReceiveTime= 18
> maximumReceiveTime= 48
> minimumReceiveTime= 0
> averageJitter = 37
>
>
> Thanks in advance for all the help.
>
> Gerardo Amaya
>
>
> -------------------------------------------------------------------------
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>
>
>
|
|
From: Gerardo A. <ope...@ge...> - 2007-01-10 18:42:31
|
Hello All. I just started using OpenSBC. I have an Asterisk Box that is
taking care of all the SIP Registration and Call Handling Process. This
Asterisk Box is on the Network 192.168.1.0/24 and I have the OpenSBC
Server 8 network cards that connect me from different networks(e.g.
10.0.0.0/24, 172.16.18.0/24, etc). I just started trying with a simple
rule from one of the Networks I wish to connect.
[sip:*@10.0.0.2] sip:192.168.1.2:5060
Since I still don't know what I'm doing I create the same route in
OpenSBC, Relay and Upper Registration, just to make sure, Now when I
tried to register to the Asterisk Box I get the following message:
2007/01/10 12:25:18.232 Registrar Debug3 Registration
rejected for sip:test@10.0.0.2
2007/01/10 12:31:21.835 Registrar Debug3 Registration
rejected for sip:test@10.0.0.2
What do I need to do in order to get the Registration process working.
RTP and Sound
In order to test the RTP Capabilities of OpenSBC I do not register for a
moment and just try to set a call(INVITE) the call is successfully
created and the other softphone answers, but I don't get any sound at
all. I see from the logs that there is RTP traffic but there is no
sound. Can you please help me telling me what I'm doing wrong
2007/01/10 12:17:15.135 OpalMediaThread:83f2738 Debug3 RTP Receive
statistics: packets=4525 octets=724000 lost=0 tooLate=0 order=0
avgTime=18 maxTime=47 minTime=0 jitter=37 maxJitter=45
2007/01/10 12:17:17.049 OpalMediaThread:83f2738 Debug3 RTP Receive
statistics: packets=4625 octets=740000 lost=0 tooLate=0 order=0
avgTime=19 maxTime=47 minTime=0 jitter=38 maxJitter=45
2007/01/10 12:17:17.129 OpalMediaThread:83f2738 Debug3 RTP
SentReceiverReport: ssrc=1888406904 fraction=0 lost=0 last_seq=37365
jitter=307 lsr=0 dlsr=0
2007/01/10 12:17:17.129 OpalMediaThread:83f2738 Debug3 RTP Sending
SDES: root@atel-router
2007/01/10 12:17:18.946 OpalMediaThread:83f2738 Debug3 RTP Receive
statistics: packets=4725 octets=756000 lost=0 tooLate=0 order=0
avgTime=18 maxTime=48 minTime=0 jitter=37 maxJitter=45
2007/01/10 12:17:20.173 Call Debug3 RTP Found
existing session 1
2007/01/10 12:17:20.173 Call Debug3 RTP_UDP Session
1, Shutting down read.
2007/01/10 12:17:20.173 OpalMediaThread:836cbb0 Debug3 RTP_UDP Session
1, Read shutdown.
2007/01/10 12:17:20.173 OpalMediaThread:836cbb0 Debug3 RTP_UDP Closing
Media Stream
2007/01/10 12:17:20.173 Call Debug3 RTP_UDP Session
1, Shutting down write.
2007/01/10 12:17:20.173 Call Debug3 RTP Found
existing session 1
2007/01/10 12:17:20.173 Call Debug3 RTP_UDP Session
1, Shutting down read.
2007/01/10 12:17:20.173 Call Debug3 RTP_UDP Session
1, Shutting down write.
2007/01/10 12:17:20.174 OpalMediaThread:83f2738 Debug3 RTP_UDP Session
1, Read shutdown.
2007/01/10 12:17:20.174 OpalMediaThread:83f2738 Debug3 RTP_UDP Closing
Media Stream
2007/01/10 12:17:20.175 Garbage Collector Debug3 RTP_UDP Session
1, Shutting down read.
2007/01/10 12:17:20.175 Garbage Collector Debug3 RTP_UDP Session
1, Shutting down write.
2007/01/10 12:17:20.175 Garbage Collector Debug3 RTP_UDP Session
1, Shutting down read.
2007/01/10 12:17:20.176 Garbage Collector Debug3 RTP_UDP Session
1, Shutting down write.
2007/01/10 12:17:20.176 Garbage Collector Debug3 RTP Final
statistics:
packetsSent = 0
octetsSent = 0
averageSendTime = 0
maximumSendTime = 0
minimumSendTime = 0
packetsReceived = 4790
octetsReceived = 766400
packetsLost = 0
packetsTooLate = 0
packetsOutOfOrder = 0
averageReceiveTime= 18
maximumReceiveTime= 48
minimumReceiveTime= 0
averageJitter = 37
Thanks in advance for all the help.
Gerardo Amaya
|
|
From: Dung Ho <ent...@ya...> - 2007-01-02 10:29:48
|
Thanks Joegen. =0AI downloaded the latest ATLSIP and tryed. It work. =0A=0A= =0A----- Original Message ----=0AFrom: Joegen E. Baclor <joegen.baclor@gmai= l.com>=0ATo: ope...@li...=0ASent: Tuesday, Janu= ary 2, 2007 5:21:48 PM=0ASubject: Re: [OpenSIPStack] Proxy Authorization fa= iled, 403 Forbidden=0A=0ATry downloading the latest ATLSIP code from CVS. = I added a separate =0Aprovision for setting ProxyAuthenticationUser and =0A= ProxyAuthenticationPassword property and test if you still get the =0AForbi= dden response. Be sure to set both Authetication and =0AProxyAuthenticatio= n properties.=0A=0AJoegen=0A=0A=0A=0A=0A=0A=0A_____________________________= _____________________=0ADo You Yahoo!?=0ATired of spam? Yahoo! Mail has th= e best spam protection around =0Ahttp://mail.yahoo.com |
|
From: Joegen E. B. <joe...@gm...> - 2007-01-02 10:22:01
|
Try downloading the latest ATLSIP code from CVS. I added a separate provision for setting ProxyAuthenticationUser and ProxyAuthenticationPassword property and test if you still get the Forbidden response. Be sure to set both Authetication and ProxyAuthentication properties. Joegen Dung Ho wrote: > Hi All, > I am a beginner SIP and Opensipstack. I used OpenSipStack library to > wirte a sipphone. After registered successfully, I make call to a > phone number and finally I got "403 Forbidden" from Registrar server. > What is my problem, why cannot I make call although I login successfull? > > Thanks. > Ken. > > I attach some sip message in file sip.log > > *** CREATED *** CLIENT REGISTER Session REGISTER-dung@192.168.8.43 > REGISTER: Starting Registration Process > > REGISTER sip:192.168.8.43 SIP/2.0 > From: <sip:192.168.8.43>;tag=7b07d6c2a3f7181098d2fd574a9dd258 > To: sip:192.168.8.43 > Via: SIP/2.0/UDP > 192.168.8.43:5061;branch=z9hG4bK960dd6c2a3f7181098d2fd574a9dd258;rport > CSeq: 1 REGISTER > Call-ID: 960dd6c2-a3f7-1810-977e-fd574a9dd258 > Contact: <sip:192.168.8.43:5061;transport=udp> > Expires: 3600 > Max-Forwards: 70 > Content-Length: 0 > > > Finding transaction for REGISTER sip:192.168.8.43 SIP/2.0 > Setting Transaction ID to > 960dd6c2-a3f7-1810-977e-fd574a9dd258|z9hG4bK960dd6c2a3f7181098d2fd574a9dd258|REGISTER > > *** CREATING TRANSACTION (NICT) *** > Message: REGISTER sip:192.168.8.43 SIP/2.0 > Call-Id: 960dd6c2-a3f7-1810-977e-fd574a9dd258 > > NICT(9575473) *** CREATED *** - > NICT|960dd6c2-a3f7-1810-977e-fd574a9dd258|z9hG4bK960dd6c2a3f7181098d2fd574a9dd258|REGISTER > NICT(9575473) Event(SIPMessage) - REGISTER sip:192.168.8.43 SIP/2.0 > TRANSACTION: (NICT) REGISTER sip:192.168.8.43 SIP/2.0 State: 0 > > TRANSMIT: REQ: REGISTER (1) > TO: sip:192.168.8.43 > XOR: FALSE > DST: 192.168.8.43 PORT: 5060 > > Finding transaction for REGISTER sip:192.168.8.43 SIP/2.0 > Setting Transaction ID to > 960dd6c2-a3f7-1810-977e-fd574a9dd258|z9hG4bK960dd6c2a3f7181098d2fd574a9dd258|REGISTER > Found > NICT|960dd6c2-a3f7-1810-977e-fd574a9dd258|z9hG4bK960dd6c2a3f7181098d2fd574a9dd258|REGISTER > for REGISTER sip:192.168.8.43 SIP/2.0 > Event: ---> TransportWrite - REGISTER > NICT(9575473) Event(SIPMessage) - REGISTER sip:192.168.8.43 SIP/2.0 > TRANSACTION: (NICT) REGISTER sip:192.168.8.43 SIP/2.0 State: 0 > NICT(9575473) StateIdle->StateTrying(REGISTER sip:192.168.8.43 > SIP/2.0) > NICT(9575473) Timer E( 500 ms ) STARTED > NICT(9575473) Timer F( 32000 ms ) STARTED > > REGISTER sip:192.168.8.43 SIP/2.0 > From: <sip:192.168.8.43>;tag=7b07d6c2a3f7181098d2fd574a9dd258 > To: sip:192.168.8.43 > Via: SIP/2.0/UDP > 192.168.8.43:5061;iid=1;branch=z9hG4bK960dd6c2a3f7181098d2fd574a9dd258;uas-addr=192.168.8.43;rport > CSeq: 1 REGISTER > Call-ID: 960dd6c2-a3f7-1810-977e-fd574a9dd258 > Contact: <sip:192.168.8.43:5061;transport=udp> > Expires: 3600 > Max-Forwards: 70 > Content-Length: 0 > > > > RECEIVE: RESPONSE: SIP/2.0 407 Proxy Authentication Required > TO: sip:192.168.8.43 > REQ: REGISTER (1) > XOR: FALSE > SRC: 192.168.8.43 PORT: 5060 > > Finding transaction for SIP/2.0 407 Proxy Authentication Required > Setting Transaction ID to > 960dd6c2-a3f7-1810-977e-fd574a9dd258|z9hG4bK960dd6c2a3f7181098d2fd574a9dd258|REGISTER > Found > NICT|960dd6c2-a3f7-1810-977e-fd574a9dd258|z9hG4bK960dd6c2a3f7181098d2fd574a9dd258|REGISTER > for SIP/2.0 407 Proxy Authentication Required > NICT(9575473) Event(SIPMessage) - SIP/2.0 407 Proxy Authentication > Required > TRANSACTION: (NICT) SIP/2.0 407 Proxy Authentication Required State: 1 > NICT(9575473) StateTrying->StateCompleted > NICT(9575473) Timer E STOPPED > NICT(9575473) Timer F STOPPED > NICT(9575473) Timer K( 5000 ms ) STARTED > Event: ---> Inbound - SIP/2.0 407 Proxy Authentication Required > > SIP/2.0 407 Proxy Authentication Required > From: <sip:192.168.8.43>;tag=7b07d6c2a3f7181098d2fd574a9dd258 > To: sip:192.168.8.43 > Via: SIP/2.0/UDP > 192.168.8.43:5061;iid=1;branch=z9hG4bK960dd6c2a3f7181098d2fd574a9dd258;uas-addr=192.168.8.43 > CSeq: 1 REGISTER > Call-ID: 960dd6c2-a3f7-1810-977e-fd574a9dd258 > Proxy-Authenticate: DIGEST realm="VirtualSoftswitch", nonce="1089adef" > Content-Length: 0 > > > REGISTER: Registration being authenticated > Finding transaction for REGISTER sip:192.168.8.43 SIP/2.0 > Setting Transaction ID to > 960dd6c2-a3f7-1810-977e-fd574a9dd258|z9hG4bK3532d6c2a3f7181098d3fd574a9dd258|REGISTER > > *** CREATING TRANSACTION (NICT) *** > Message: REGISTER sip:192.168.8.43 SIP/2.0 > Call-Id: 960dd6c2-a3f7-1810-977e-fd574a9dd258 > > NICT(9575474) *** CREATED *** - > NICT|960dd6c2-a3f7-1810-977e-fd574a9dd258|z9hG4bK3532d6c2a3f7181098d3fd574a9dd258|REGISTER > NICT(9575474) Event(SIPMessage) - REGISTER sip:192.168.8.43 SIP/2.0 > TRANSACTION: (NICT) REGISTER sip:192.168.8.43 SIP/2.0 State: 0 > > TRANSMIT: REQ: REGISTER (2) > TO: sip:192.168.8.43 > XOR: FALSE > DST: 192.168.8.43 PORT: 5060 > > Finding transaction for REGISTER sip:192.168.8.43 SIP/2.0 > Setting Transaction ID to > 960dd6c2-a3f7-1810-977e-fd574a9dd258|z9hG4bK3532d6c2a3f7181098d3fd574a9dd258|REGISTER > Found > NICT|960dd6c2-a3f7-1810-977e-fd574a9dd258|z9hG4bK3532d6c2a3f7181098d3fd574a9dd258|REGISTER > for REGISTER sip:192.168.8.43 SIP/2.0 > Event: ---> TransportWrite - REGISTER > Event(SIPMessage) - REGISTER sip:192.168.8.43 SIP/2.0 > TRANSACTION: (NICT) REGISTER sip:192.168.8.43 SIP/2.0 State: 0 > NICT(9575474) StateIdle->StateTrying(REGISTER sip:192.168.8.43 SIP/2.0) > NICT(9575474) Timer E( 500 ms ) STARTED > NICT(9575474) Timer F( 32000 ms ) STARTED > > *** MESSAGE SENT *** for SIP Session REGISTER-dung@192.168.8.43 > > > REGISTER sip:192.168.8.43 SIP/2.0 > From: <sip:192.168.8.43>;tag=3532d6c2a3f7181098d2fd574a9dd258 > To: sip:192.168.8.43 > Via: SIP/2.0/UDP > 192.168.8.43:5061;iid=1;branch=z9hG4bK3532d6c2a3f7181098d3fd574a9dd258;uas-addr=192.168.8.43;rport > CSeq: 2 REGISTER > Call-ID: 960dd6c2-a3f7-1810-977e-fd574a9dd258 > Contact: <sip:192.168.8.43:5061;transport=udp> > Expires: 3600 > Max-Forwards: 70 > Proxy-Authorization: Digest username="dung", > realm="VirtualSoftswitch", nonce="1089adef", uri="sip:192.168.8.43", > response="eed9ae8b62f398f88da68cea10ecea6b", algorithm=MD5 > Content-Length: 0 > > > > RECEIVE: RESPONSE: SIP/2.0 200 OK > TO: sip:192.168.8.43 > REQ: REGISTER (2) > XOR: FALSE > SRC: 192.168.8.43 PORT: 5060 > > Finding transaction for SIP/2.0 200 OK > Setting Transaction ID to > 960dd6c2-a3f7-1810-977e-fd574a9dd258|z9hG4bK3532d6c2a3f7181098d3fd574a9dd258|REGISTER > Found > NICT|960dd6c2-a3f7-1810-977e-fd574a9dd258|z9hG4bK3532d6c2a3f7181098d3fd574a9dd258|REGISTER > for SIP/2.0 200 OK > NICT(9575474) Event(SIPMessage) - SIP/2.0 200 OK > TRANSACTION: (NICT) SIP/2.0 200 OK State: 1 > NICT(9575474) StateTrying->StateCompleted > NICT(9575474) Timer E STOPPED > NICT(9575474) Timer F STOPPED > NICT(9575474) Timer K( 5000 ms ) STARTED > Event: ---> Inbound - SIP/2.0 200 OK > > SIP/2.0 200 OK > From: <sip:192.168.8.43>;tag=3532d6c2a3f7181098d2fd574a9dd258 > To: sip:192.168.8.43 > Via: SIP/2.0/UDP > 192.168.8.43:5061;iid=1;branch=z9hG4bK3532d6c2a3f7181098d3fd574a9dd258;uas-addr=192.168.8.43 > CSeq: 2 REGISTER > Call-ID: 960dd6c2-a3f7-1810-977e-fd574a9dd258 > Expires: 600 > Content-Length: 0 > > > REGISTER: Registration Accepted > > Session CREATED > *** CREATED *** Call Session > [SIP Session Manager]....................| CREATED > via=192.168.8.43:5061 for target=192.168.8.43 protocol=UDP > Finding transaction for INVITE sip:123456715032231700@192.168.8.43 SIP/2.0 > Setting Transaction ID to > 8d05e2c2-a3f7-1810-977f-fd574a9dd258|z9hG4bKa70be2c2a3f7181098d3fd574a9dd258|INVITE > > *** CREATING TRANSACTION (ICT) *** > Message: INVITE sip:123456715032231700@192.168.8.43 SIP/2.0 > Call-Id: 8d05e2c2-a3f7-1810-977f-fd574a9dd258 > > ICT(9575475) *** CREATED *** - > ICT|8d05e2c2-a3f7-1810-977f-fd574a9dd258|z9hG4bKa70be2c2a3f7181098d3fd574a9dd258|INVITE > ICT(9575475) Event(SIPMessage) - INVITE > sip:123456715032231700@192.168.8.43 SIP/2.0 > TRANSACTION: (ICT) INVITE sip:123456715032231700@192.168.8.43 SIP/2.0 > State: 0 > > TRANSMIT: REQ: INVITE (4711) > TO: sip:123456715032231700@192.168.8.43 > XOR: FALSE > DST: 192.168.8.43 PORT: 5060 > > Finding transaction for INVITE sip:123456715032231700@192.168.8.43 SIP/2.0 > Setting Transaction ID to > 8d05e2c2-a3f7-1810-977f-fd574a9dd258|z9hG4bKa70be2c2a3f7181098d3fd574a9dd258|INVITE > Found > ICT|8d05e2c2-a3f7-1810-977f-fd574a9dd258|z9hG4bKa70be2c2a3f7181098d3fd574a9dd258|INVITE > for INVITE sip:123456715032231700@192.168.8.43 SIP/2.0 > ICT(9575475) Event(SIPMessage) - INVITE > sip:123456715032231700@192.168.8.43 SIP/2.0 > TRANSACTION: (ICT) INVITE sip:123456715032231700@192.168.8.43 SIP/2.0 > State: 0 > ICT(9575475) StateIdle->StateCalling(INVITE > sip:123456715032231700@192.168.8.43 SIP/2.0) > ICT(9575475) Timer A( 500 ms ) STARTED > ICT(9575475) Timer B( 32000 ms ) STARTED > Event: ---> TransportWrite - INVITE > *** MESSAGE SENT *** for SIP Session 8d05e2c2-a3f7-1810-977f-fd574a9dd258 > > INVITE sip:123456715032231700@192.168.8.43 SIP/2.0 > From: tandung > <sip:tandung@192.168.8.43>;tag=a70be2c2a3f7181098d4fd574a9dd258 > To: sip:123456715032231700@192.168.8.43 > Via: SIP/2.0/UDP > 192.168.8.43:5061;iid=2;branch=z9hG4bKa70be2c2a3f7181098d3fd574a9dd258;uas-addr=192.168.8.43;rport > CSeq: 4711 INVITE > Call-ID: 8d05e2c2-a3f7-1810-977f-fd574a9dd258 > Contact: "tandung" <sip:tandung@192.168.8.43:5061> > Max-Forwards: 70 > Content-Type: application/sdp > Content-Length: 295 > > v=0 > o=- 1167708023 1167708023 IN IP4 192.168.8.43 > s=OSS RTP Session > c=IN IP4 192.168.8.43 > t=0 0 > m=audio 5000 RTP/AVP 101 18 106 103 3 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=rtpmap:18 G729/8000 > a=rtpmap:106 iLBC/8000 > a=rtpmap:103 SpeexNarrow-15k/8000 > a=rtpmap:3 GSM/8000 > > > > > RECEIVE: RESPONSE: SIP/2.0 407 Proxy Authentication Required > TO: sip:123456715032231700@192.168.8.43 > REQ: INVITE (4711) > XOR: FALSE > SRC: 192.168.8.43 PORT: 5060 > > Finding transaction for SIP/2.0 407 Proxy Authentication Required > Setting Transaction ID to > 8d05e2c2-a3f7-1810-977f-fd574a9dd258|z9hG4bKa70be2c2a3f7181098d3fd574a9dd258|INVITE > Found > ICT|8d05e2c2-a3f7-1810-977f-fd574a9dd258|z9hG4bKa70be2c2a3f7181098d3fd574a9dd258|INVITE > for SIP/2.0 407 Proxy Authentication Required > ICT(9575475) Event(SIPMessage) - SIP/2.0 407 Proxy Authentication > Required > TRANSACTION: (ICT) SIP/2.0 407 Proxy Authentication Required State: 1 > ICT(9575475) Timer A STOPPED > ICT(9575475) Timer B STOPPED > ICT(9575475) StateCalling->StateCompleted(SIP/2.0 407 Proxy > Authentication Required) > Event: ---> Inbound - SIP/2.0 407 Proxy Authentication Required > *** MESSAGE ARRIVAL *** for SIP Session > 8d05e2c2-a3f7-1810-977f-fd574a9dd258 > > SIP/2.0 407 Proxy Authentication Required > From: tandung > <sip:tandung@192.168.8.43>;tag=a70be2c2a3f7181098d4fd574a9dd258 > To: sip:123456715032231700@192.168.8.43 > Via: SIP/2.0/UDP > 192.168.8.43:5061;iid=2;branch=z9hG4bKa70be2c2a3f7181098d3fd574a9dd258;uas-addr=192.168.8.43 > CSeq: 4711 INVITE > Call-ID: 8d05e2c2-a3f7-1810-977f-fd574a9dd258 > Contact: <sip:192.168.8.43:5060;user=phone;transport=udp> > Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="1089adef" > Content-Length: 0 > > > Finding transaction for INVITE sip:123456715032231700@192.168.8.43 SIP/2.0 > Setting Transaction ID to > 8d05e2c2-a3f7-1810-977f-fd574a9dd258|z9hG4bKe554e2c2a3f7181098d4fd574a9dd258|INVITE > > *** CREATING TRANSACTION (ICT) *** > Message: INVITE sip:123456715032231700@192.168.8.43 SIP/2.0 > Call-Id: 8d05e2c2-a3f7-1810-977f-fd574a9dd258 > > ICT(9575476) *** CREATED *** - > ICT|8d05e2c2-a3f7-1810-977f-fd574a9dd258|z9hG4bKe554e2c2a3f7181098d4fd574a9dd258|INVITE > ICT(9575476) Event(SIPMessage) - INVITE > sip:123456715032231700@192.168.8.43 SIP/2.0 > TRANSACTION: (ICT) INVITE sip:123456715032231700@192.168.8.43 SIP/2.0 > State: 0 > > TRANSMIT: REQ: ACK (4711) > TO: sip:123456715032231700@192.168.8.43 > XOR: FALSE > DST: 192.168.8.43 PORT: 5060 > > ICT(9575475) Timer D( 32000 ms ) STARTED > > TRANSMIT: REQ: INVITE (4712) > TO: sip:123456715032231700@192.168.8.43 > XOR: FALSE > DST: 192.168.8.43 PORT: 5060 > > Finding transaction for ACK sip:123456715032231700@192.168.8.43 SIP/2.0 > Setting Transaction ID to > 8d05e2c2-a3f7-1810-977f-fd574a9dd258|z9hG4bKa70be2c2a3f7181098d3fd574a9dd258|ACK > > *** TRANSACTION DOES NOT EXIST *** > Message: ACK sip:123456715032231700@192.168.8.43 SIP/2.0 > Call-Id: 8d05e2c2-a3f7-1810-977f-fd574a9dd258 > > Event: ---> TransportWrite - INVITE > *** MESSAGE SENT *** for SIP Session 8d05e2c2-a3f7-1810-977f-fd574a9dd258 > > ACK sip:123456715032231700@192.168.8.43 SIP/2.0 > From: tandung > <sip:tandung@192.168.8.43>;tag=a70be2c2a3f7181098d4fd574a9dd258 > To: sip:123456715032231700@192.168.8.43 > Via: SIP/2.0/UDP > 192.168.8.43:5061;iid=2;branch=z9hG4bKa70be2c2a3f7181098d3fd574a9dd258;uas-addr=192.168.8.43;rport > CSeq: 4711 ACK > Call-ID: 8d05e2c2-a3f7-1810-977f-fd574a9dd258 > Contact: "tandung" <sip:tandung@192.168.8.43:5061> > Max-Forwards: 70 > Content-Length: 0 > > > Event: ---> UnknownTransaction - ACK > Finding transaction for INVITE sip:123456715032231700@192.168.8.43 SIP/2.0 > Setting Transaction ID to > 8d05e2c2-a3f7-1810-977f-fd574a9dd258|z9hG4bKe554e2c2a3f7181098d4fd574a9dd258|INVITE > Found > ICT|8d05e2c2-a3f7-1810-977f-fd574a9dd258|z9hG4bKe554e2c2a3f7181098d4fd574a9dd258|INVITE > for INVITE sip:123456715032231700@192.168.8.43 SIP/2.0 > ICT(9575476) Event(SIPMessage) - INVITE > sip:123456715032231700@192.168.8.43 SIP/2.0 > TRANSACTION: (ICT) INVITE sip:123456715032231700@192.168.8.43 SIP/2.0 > State: 0 > ICT(9575476) StateIdle->StateCalling(INVITE > sip:123456715032231700@192.168.8.43 SIP/2.0) > ICT(9575476) Timer A( 500 ms ) STARTED > ICT(9575476) Timer B( 32000 ms ) STARTED > Event: ---> TransportWrite - INVITE > *** MESSAGE SENT *** for SIP Session 8d05e2c2-a3f7-1810-977f-fd574a9dd258 > > INVITE sip:123456715032231700@192.168.8.43 SIP/2.0 > From: tandung > <sip:tandung@192.168.8.43>;tag=a70be2c2a3f7181098d4fd574a9dd258 > To: sip:123456715032231700@192.168.8.43 > Via: SIP/2.0/UDP > 192.168.8.43:5061;iid=2;branch=z9hG4bKe554e2c2a3f7181098d4fd574a9dd258;uas-addr=192.168.8.43;rport > CSeq: 4712 INVITE > Call-ID: 8d05e2c2-a3f7-1810-977f-fd574a9dd258 > Contact: "tandung" <sip:tandung@192.168.8.43:5061> > Max-Forwards: 70 > Proxy-Authorization: Digest username="dung", realm="VoipSwitch", > nonce="1089adef", uri="sip:123456715032231700@192.168.8.43", > response="50d2cb7adda208d7d4aef2fd59f3f842", algorithm=MD5 > Content-Type: application/sdp > Content-Length: 295 > > v=0 > o=- 1167708023 1167708023 IN IP4 192.168.8.43 > s=OSS RTP Session > c=IN IP4 192.168.8.43 > t=0 0 > m=audio 5000 RTP/AVP 101 18 106 103 3 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=rtpmap:18 G729/8000 > a=rtpmap:106 iLBC/8000 > a=rtpmap:103 SpeexNarrow-15k/8000 > a=rtpmap:3 GSM/8000 > > > > > RECEIVE: RESPONSE: SIP/2.0 403 Forbidden > TO: sip:123456715032231700@192.168.8.43 > REQ: INVITE (4712) > XOR: FALSE > SRC: 192.168.8.43 PORT: 5060 > > Finding transaction for SIP/2.0 403 Forbidden > Setting Transaction ID to > 8d05e2c2-a3f7-1810-977f-fd574a9dd258|z9hG4bKa70be2c2a3f7181098d3fd574a9dd258|INVITE > Found > ICT|8d05e2c2-a3f7-1810-977f-fd574a9dd258|z9hG4bKa70be2c2a3f7181098d3fd574a9dd258|INVITE > for SIP/2.0 403 Forbidden > ICT(9575475) Event(SIPMessage) - SIP/2.0 403 Forbidden > TRANSACTION: (ICT) SIP/2.0 403 Forbidden State: 3 > > TRANSMIT: REQ: ACK (4711) > TO: sip:123456715032231700@192.168.8.43 > XOR: FALSE > DST: 192.168.8.43 PORT: 5060 > > > RECEIVE: RESPONSE: 52685 > TO: sip:123456715032231700@192.168.8.43 > REQ: INVITE (4712) > XOR: FALSE > SRC: 192.168.8.43 PORT: 5060 > > Finding transaction for 52685 > Setting Transaction ID to > 8d05e2c2-a3f7-1810-977f-fd574a9dd258|z9hG4bKa70be2c2a3f7181098d3fd574a9dd258|INVITE > Found > ICT|8d05e2c2-a3f7-1810-977f-fd574a9dd258|z9hG4bKa70be2c2a3f7181098d3fd574a9dd258|INVITE > for 52685 > ICT(9575475) Event(SIPMessage) - 52685 > TRANSACTION: (ICT) 52685 State: 3 > > TRANSMIT: REQ: ACK (4711) > TO: sip:123456715032231700@192.168.8.43 > XOR: FALSE > DST: 192.168.8.43 PORT: 5060 > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > ------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
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From: Joegen E. B. <joe...@gm...> - 2007-01-02 01:28:33
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Gustavo Curetti wrote: > > Hi Joegen, > > I was configuring " [sip:*@192.168.0.206*] sip:192.168.0.206:5060 " > with an extra space in the beginning instead of > "[sip:*@192.168.0.206*] sip:192.168.0.206:5060". > > I suggest to make this change: > > PString buff = routeRec.Mid( less+1 , greater-1 ); --> PString buff = > routeRec.Mid( less+1 , greater-less-1 ); > > because PString::Mid() receive "PINDEX len" for the second argument > instead of an index. > > Gustavo > This has been commited to CVS. Thanks for the bug report. Joegen |
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From: Joegen E. B. <joe...@gm...> - 2007-01-02 01:17:58
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rh...@te... wrote: > About LogOut : currently on the ActiveX the LogOut procedure seems to > process nothing => None Event (OK or Error). > Still not sure what you mean by this. It fires events just fine here. If you are calling DoLogout() on exit, then you should use a semaphore to wait for the ok or error event. > I had a quick look on the low level API. I'm very surprised by the > potential of such framework (XMPP, flexible codec, AEC,...) this is a Credit is not for OpenSIPStack alone but to the great work the OPAL project pulled together over the years. OpenSIPStack uses a fork of OPAL media classes for RTP and Codecs. Joegen |
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From: <rh...@te...> - 2006-12-30 10:22:35
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Thanks Joegen for your quick answer, By " Sound Manager for Call event" you had to understand a way to change sound associated to different SIP states during an INVITE. In my previous project based upon MS RTC, it was a feature asked by users :) About Log file and Config INI file, the subject is to be able to get a flag to avoid generation of such files (which can generate pb for multiusers Windows configuration : all can access to the same profile) About LogOut : currently on the ActiveX the LogOut procedure seems to process nothing =3D> None Event (OK or Error). I had a quick look on the low level API. I'm very surprised by the potential of such framework (XMPP, flexible codec, AEC,...) this is a fantastic work. I guess it will be difficult to include most of features in the AtlSIP ActiveX, but I hope, my knowledge on C/CPP are so low. Is there a way I can help you? Regards ... I wish to all a happy new year and a fantastic development for OpenSIPStack project!! > -----Message d'origine----- > De=A0: ope...@li... > [mailto:ope...@li...] De la part de > Joegen E. Baclor > Envoy=E9=A0: samedi 30 d=E9cembre 2006 05:05 > =C0=A0: ope...@li... > Objet=A0: Re: [OpenSIPStack] AltSip : a pearl near to a gem... > > rh...@te... wrote: > > Hi, > > > > I was looking for a high level API to help me to develop a Softphone in > > place of buggy MS RTC for Windows, and I found AltSIP ActiveX. > > > > 1 hour later, my sample project in Delphi 7 was able to register, make > > and receive SIP Call, select Codec, select audio device, manage DTMF,... > > in one word, wonderfull! > > > > > Thanks for the kind words. I truly enjoy hearing that people are > starting to use OpenSIPStack. > > > I know the ActiveX is currently in development but I'm impatient to find > > next features : > > - Presence and Watcher management (SIMPLE mode) > > > > Presence and Messaging support is currently on the operating table. > There will be development on this subject in the weeks to come. I > previous announced that SIMPLE support would be available this December > but activities somewhere else deprived me of the time to accomplish it > in time. > > > - Video Support > > > Video is supported by the lower level API but I didn't find the time yet > to hook this up to the ActiveX layer. > > > - IM feature (like events) > > > See above > > - Media manager for player and mic level > > > > This is also available in the lower layer but isn't hooked yet to ATLSIP > > - Sound Manager for Call event (ringing, trying,...) > > > > > > Not sure what you mean by this. Can you expound a bit? > > > If I can ask for something to developers : > > - property to disable usage of INI files : to enable coders to manager > > with their own way Profile, contacts, codecs,... > > > > This was a design decision because ATLSIP uses an Interface that is > suppose to run on multiple operating system. Therefore, a common > configuration framework should be utilized to assure uniformity across > all supported platforms. If you want to implement your own config, > you must do a bit of coding in C++ and modify the setters and getters in > SoftPhoneInterface.cxx > > > - property to disable usage of Log files : may be with a specific event > > for logs > > > > > > Ahh, thats right. Let me hack the code a bit and expose this right > away first chance I get. > > > There is some feature I don't understand (yet) : > > - GetAccountBalance > > > > This is suppose to be the start of a built in accounting functionality. > however, I can't find a standard means of doing this in any SIP or SIP > related standards. I am currently looking at possibly using Diameter as > soon as proper correlation of SIP and Diameter is defined by the > standards body. > > > - XORHash > > > > > > This is proprietary hash encryption supported by some commercial > phones. This is basically a simple XOR hash algorithm applied to RTP > and SIP packet as they are sent and received . As long as two UAs > share a common hash, they would be able to communicate with each > other. I woudln't advise use of this functionality if there is a > proxy being utilized for surely, this wouldn't work as expected. > > > > I think there's also a pb with Logout function. > > > > What's a 'pb' ? > > In any case, I'm really happy to find such project. > > > > Regards > > > > > > > > > > > > ------------------------------------------------------------------------ > - > > Take Surveys. Earn Cash. Influence the Future of IT > > Join SourceForge.net's Techsay panel and you'll get the chance to share > your > > opinions on IT & business topics through brief surveys - and earn cash > > > http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3DDE= VDE V > > _______________________________________________ > > opensipstack-devel mailing list > > ope...@li... > > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > > > > > > > ------------------------------------------------------------------------ - > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share > your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=3Djoin.php&p=3Dsourceforge&CID=3DDE= VDE V > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel |
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From: Leo A. B. <le...@da...> - 2006-12-30 07:40:44
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Ammar Hassan wrote: > > Hi i am new to the project. I donwloaded Opensipstack and as per > instructions downloaded VC++ 2005. Now when i ran i get 241 errors of > > \opensipstack\include\ptlib\msos/ptlib/contain.h(285) : fatal error > C1083: Cannot open include file: 'windows.h': No such file or directory. > Did you download and install the Windows Platform SDK as well? The downloaded version of VC++ 2005 does not include the header files required for compiling native Win32 binaries. Leo |
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From: Joegen E. B. <joe...@gm...> - 2006-12-30 07:33:03
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Ammar, If you are using Visual C++ 2005 Express edition, you also need to download the Platform SDK. There is an instruction provided for this in the Microsoft download site where you downloaded the Express Edition. If you already have the platform SDK, you need to include the platform SDK in the Include Path. In my case this is `C:\Program Files (x86)\Microsoft Visual Studio 8\VC\PlatformSDK\Include` Joegen Ammar Hassan wrote: > > Hi i am new to the project. I donwloaded Opensipstack and as per > instructions downloaded VC++ 2005. Now when i ran i get 241 errors of > > \opensipstack\include\ptlib\msos/ptlib/contain.h(285) : fatal error > C1083: Cannot open include file: 'windows.h': No such file or directory. > > don't really know how to solve the problem. I tried removing > > #ifndef _WINDOWS_ > # include <windows.h> > #endif > > considering that windows.h doens't need to be added in 2005 projects > apropos some link at msdn. > Any ideas on how to solve this. I am balked down > Ammar > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > Take Surveys. Earn Cash. Influence the Future of IT > Join SourceForge.net's Techsay panel and you'll get the chance to share your > opinions on IT & business topics through brief surveys - and earn cash > http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV > ------------------------------------------------------------------------ > > _______________________________________________ > opensipstack-devel mailing list > ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-devel > |
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