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From: Reg <ope...@op...> - 2008-02-11 18:53:05
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Is OpenSBC also a STUN Server? Thank you for response. |
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From: Joegen E. B. <joe...@gm...> - 2008-02-10 08:53:36
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Seems like a system limit getting hit to me. Try increasing your limits and see if you could stretch concurrency a bit further. craig guy wrote: > Hi again, > > > I think it died through the number of concurrent calls rather than the CPS. I performed another test at 60 concurrent calls and found a peak CPS of 18. OpenSBC stayed up at the 60 concurrent call level with media proxying. > > > Craig > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Opensipstack-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > |
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From: craig g. <cra...@ze...> - 2008-02-10 07:24:15
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Hi again, I think it died through the number of concurrent calls rather than the CPS. I performed another test at 60 concurrent calls and found a peak CPS of 18. OpenSBC stayed up at the 60 concurrent call level with media proxying. Craig |
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From: craig g. <cra...@ze...> - 2008-02-10 06:46:35
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Thanks Joegen, I have set a softt limit of 4000 and a hard limit of 6000 for nofiles in limits.conf. I have no idea if that is too many or too little. Would you suggest I also set the -H opensbc option? I will set up a test to see if it is the CPS or concurrent calls that seems to be problematic. Craig |
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From: Joegen E. B. <joe...@gm...> - 2008-02-09 13:00:09
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I would check the max file handles per process setting. However, it would be good if you could also test gradual rise to see if it is the CPS that OSBC can't handle or is it the concurrent calls. From experience the optimum CPS for opensbc is between 15-20. Beyond that, the queue can grow beyond recovery. Increasing CPS is one of the major deliverables in OpenSBC version 2. A lot of redesign is happening at this very moment to make this happen. Joegen craig guy wrote: > Hi, I am running an instance of OpenSBC 1.1.5 and opensipstack pulled from CVS on 26 January. I'm running it in B2BUA mode with 'always proxy media' set. The instance of OpenSBC has two interfaces, one external on the internet and one internal. There are 2 x Asterisk servers sitting on the internal interface with incoming calls being routed to them in round robin fashion. I sent it 26 concurrent calls (setup in about 1 second) in its default configuration successfully. I then upped the ante by sending it 120 concurrent calls (all hitting OpenSBC within a second) and it just stopped working (process died). What settings should I configure in order to support 120 concurrent calls with media proxying? Each call will have 4 rtp sessions and hence 480 total rtp sessions. The default logs don't really show anything useful, they are fine until they stop at the point OpenSBC dies. OS is Centos 5. > > > Craig > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Opensipstack-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > |
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From: craig g. <cra...@ze...> - 2008-02-09 04:32:34
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Hi, I am running an instance of OpenSBC 1.1.5 and opensipstack pulled from CVS on 26 January. I'm running it in B2BUA mode with 'always proxy media' set. The instance of OpenSBC has two interfaces, one external on the internet and one internal. There are 2 x Asterisk servers sitting on the internal interface with incoming calls being routed to them in round robin fashion. I sent it 26 concurrent calls (setup in about 1 second) in its default configuration successfully. I then upped the ante by sending it 120 concurrent calls (all hitting OpenSBC within a second) and it just stopped working (process died). What settings should I configure in order to support 120 concurrent calls with media proxying? Each call will have 4 rtp sessions and hence 480 total rtp sessions. The default logs don't really show anything useful, they are fine until they stop at the point OpenSBC dies. OS is Centos 5. Craig |
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From: Yuriy <yuk...@ma...> - 2008-02-08 14:40:53
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Joegen, I sent you the letter with logs. I'll try to install OpenSBC under Debian. May be It can help me... |
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From: Parikshit K. D. <pkd...@gm...> - 2008-02-07 10:05:45
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Hi, I am trying to setup OpenSBC on my linux machine. After downloading and compiling the openSIPStack I need the openSBC unix/linux binaries to launch openSBC. Can anyone share or let me know where to get it from ? The [http://www.opensipstack.org/downloads.html] does not show any available unix/linux binaries ready for download. Regards, Parikshit |
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From: Joegen E. B. <joe...@gm...> - 2008-02-05 15:37:44
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You need to send in level 5 logs. Mail it to me directly if they are too big. jo...@op... Erik Boyer wrote: > Hello, > > > I've set up OSBC in B2BUpperReg mode so that it is a proxy between the UA and our Metaswitch (a class 5 softswitch). The UA successfully registers with the meta and is able to make and recieve calls. Incoming calls are perfectly fine, however, when a call is made +from+ the UA instead of +to+ the UA, the audio drops after about 20 seconds. The call remains connected, but there is no audio on either end of the call. I'm using the CVS version from yesterday, Feb 4 v1.1.5; the reason I think this is a bug with 1.1.5 is that 1.1.4 does not have this issue and works as expected. Any help you can provide me on this issue would be greatly appreciated. > > > Thanks in advance, > > > Erik > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Opensipstack-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > |
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From: Erik B. <eri...@ma...> - 2008-02-05 15:27:38
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Hello,
I've set up OSBC in B2BUpperReg mode so that it is a proxy between the UA and our Metaswitch (a class 5 softswitch). The UA successfully registers with the meta and is able to make and recieve calls. Incoming calls are perfectly fine, however, when a call is made +from+ the UA instead of +to+ the UA, the audio drops after about 20 seconds. The call remains connected, but there is no audio on either end of the call. I'm using the CVS version from yesterday, Feb 4 v1.1.5; the reason I think this is a bug with 1.1.5 is that 1.1.4 does not have this issue and works as expected. Any help you can provide me on this issue would be greatly appreciated.
Thanks in advance,
Erik
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From: Joegen E. B. <joe...@gm...> - 2008-02-05 11:34:30
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Hi Yuriy, There have been recent patches in CVS that concerned a deadlock between register authentication and the mutex guarding the Local Domain Accounts items. If your problem still persists, feel free to send me the logs at jo...@op... so I can take a look. Joegen Yuriy wrote: > Good day! > > OpeSBC it is configured and correctly works in b2bUpperReg Mode. But there is a serious problem, which reason for me is not known. At a plenty of requests REGISTER (for example at input wrong password in UA) OpenSBC ceases to work - any more does not write messages to log, answers from OpenSBC no more send. EventQueue reaches nearby 150-200. Process does not take off. I restart process, but the problem repeats. The more requests to OpenSBC, than the more quickly OpenSBC will break. > > How to solve this problem? > > #uname -a > Linux classV 2.6.18-8.el5 #1 SMP Fri Jan 26 14:15:21 EST 2007 i686 i686 i386 GNU/Linu > > Last messages in log: > 333925:23:34.062 INF: [CID=0x0e6f] *** UPPER REGISTRATION ENABLE *** for sip:6563630554@192.168.1.105 > 333925:23:34.079 INF: [CID=0x0e6f] Event: Setting UA Core [Proxy] to handle event REGISTER > 333925:23:34.299 INF: [CID=0x0e6f] Event: Setting UA Core [Proxy] to handle event REGISTER > 333925:23:34.299 INF: [CID=0x0e6f] Session CREATED > 333925:23:34.299 INF: [CID=0x0e6f] *** CREATED *** PROXY Session MDZmNDAwNDI1NDEyYmYzNDc5MDhlMjZmYmE1OWJhZDk. > 333925:23:34.304 INF: [CID=0x0e6f] Routing REGISTER for URI sip:6563630554@192.168.1.105 > 333925:23:34.305 PWL: [CID=0x0000] -->> From: sip:6563630554@192.168.1.105 Target: REGISTER sip:6563630554@192.168.1.105:5060 > 333925:23:34.305 INF: [CID=0x0e6f] *** UPPER REGISTRATION ENABLE *** for sip:6563630554@192.168.1.105 > 333925:23:34.970 INF: [CID=0x0e6f] Event: Setting UA Core [Proxy] to handle event REGISTER > 333925:23:34.975 INF: [CID=0x0e6f] *** DESTROYED *** PROXY Session MDZmNDAwNDI1NDEyYmYzNDc5MDhlMjZmYmE1OWJhZDk. > 333925:23:34.976 INF: [CID=0x0e6f] PROXY: Session DESTROYED > 333925:23:34.977 INF: [CID=0x0e6f] *** DESTROYED *** PROXY Session MDZmNDAwNDI1NDEyYmYzNDc5MDhlMjZmYmE1OWJhZDk. > 333925:23:34.977 INF: [CID=0x0e6f] PROXY: Session DESTROYED > 333925:23:35.194 INF: [CID=0x0e6f] Event: Setting UA Core [Proxy] to handle event REGISTER > 333925:23:35.194 INF: [CID=0x0e6f] Session CREATED > 333925:23:35.194 INF: [CID=0x0e6f] *** CREATED *** PROXY Session MDZmNDAwNDI1NDEyYmYzNDc5MDhlMjZmYmE1OWJhZDk. > 333925:23:35.198 INF: [CID=0x0e6f] Routing REGISTER for URI sip:6563630554@192.168.1.105 > 333925:23:35.198 PWL: [CID=0x0000] -->> From: sip:6563630554@192.168.1.105 Target: REGISTER sip:6563630554@192.168.1.105:5060 > 333925:23:35.199 INF: [CID=0x0e6f] *** UPPER REGISTRATION ENABLE *** for sip:6563630554@192.168.1.105 > 333925:23:35.220 INF: [CID=0x0e6f] Event: Setting UA Core [Proxy] to handle event REGISTER > 333925:23:35.439 INF: [CID=0x0e6f] Event: Setting UA Core [Proxy] to handle event REGISTER > 333925:23:35.440 INF: [CID=0x0e6f] Session CREATED > 333925:23:35.440 INF: [CID=0x0e6f] *** CREATED *** PROXY Session MDZmNDAwNDI1NDEyYmYzNDc5MDhlMjZmYmE1OWJhZDk. > 333925:23:35.444 INF: [CID=0x0e6f] Routing REGISTER for URI sip:6563630554@192.168.1.105 > 333925:23:35.444 PWL: [CID=0x0000] -->> From: sip:6563630554@192.168.1.105 Target: REGISTER sip:6563630554@192.168.1.105:5060 > 333925:23:35.445 INF: [CID=0x0e6f] *** UPPER REGISTRATION ENABLE *** for sip:6563630554@192.168.1.105 > 333925:23:35.463 INF: [CID=0x0e6f] Event: Setting UA Core [Proxy] to handle event REGISTER > 333925:23:35.467 INF: [CID=0x0e6f] *** DESTROYED *** PROXY Session MDZmNDAwNDI1NDEyYmYzNDc5MDhlMjZmYmE1OWJhZDk. > 333925:23:35.467 INF: [CID=0x0e6f] PROXY: Session DESTROYED > 333925:23:35.468 INF: [CID=0x0e6f] *** DESTROYED *** PROXY Session MDZmNDAwNDI1NDEyYmYzNDc5MDhlMjZmYmE1OWJhZDk. > 333925:23:35.468 INF: [CID=0x0e6f] PROXY: Session DESTROYED > 333925:23:35.682 INF: [CID=0x0e6f] Event: Setting UA Core [Proxy] to handle event REGISTER > 333925:23:35.683 INF: [CID=0x0e6f] Session CREATED > 333925:23:35.683 INF: [CID=0x0e6f] *** CREATED *** PROXY Session MDZmNDAwNDI1NDEyYmYzNDc5MDhlMjZmYmE1OWJhZDk. > 333925:23:35.691 INF: [CID=0x0e6f] Routing REGISTER for URI sip:6563630554@192.168.1.105 > 333925:23:35.692 PWL: [CID=0x0000] -->> From: sip:6563630554@192.168.1.105 Target: REGISTER sip:6563630554@192.168.1.105:5060 > 333925:23:35.692 INF: [CID=0x0e6f] *** UPPER REGISTRATION ENABLE *** for sip:6563630554@192.168.1.105 > 333925:23:35.694 INF: [CID=0x0e66] Event: Setting UA Core [Proxy] to handle event REGISTER > 333925:23:35.695 INF: [CID=0x0e66] Session CREATED > 333925:23:35.696 INF: [CID=0x0e66] *** CREATED *** PROXY Session MjliMjEwMGU3ZmM3YTQ3MTJiOGJlZGY0MjYzZDdkOTA. > > Best regards, > Yuriy. > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Opensipstack-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > |
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From: Yuriy <yuk...@ma...> - 2008-02-05 08:25:07
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Good day! OpeSBC it is configured and correctly works in b2bUpperReg Mode. But there is a serious problem, which reason for me is not known. At a plenty of requests REGISTER (for example at input wrong password in UA) OpenSBC ceases to work - any more does not write messages to log, answers from OpenSBC no more send. EventQueue reaches nearby 150-200. Process does not take off. I restart process, but the problem repeats. The more requests to OpenSBC, than the more quickly OpenSBC will break. How to solve this problem? #uname -a Linux classV 2.6.18-8.el5 #1 SMP Fri Jan 26 14:15:21 EST 2007 i686 i686 i386 GNU/Linu Last messages in log: 333925:23:34.062 INF: [CID=0x0e6f] *** UPPER REGISTRATION ENABLE *** for sip:6563630554@192.168.1.105 333925:23:34.079 INF: [CID=0x0e6f] Event: Setting UA Core [Proxy] to handle event REGISTER 333925:23:34.299 INF: [CID=0x0e6f] Event: Setting UA Core [Proxy] to handle event REGISTER 333925:23:34.299 INF: [CID=0x0e6f] Session CREATED 333925:23:34.299 INF: [CID=0x0e6f] *** CREATED *** PROXY Session MDZmNDAwNDI1NDEyYmYzNDc5MDhlMjZmYmE1OWJhZDk. 333925:23:34.304 INF: [CID=0x0e6f] Routing REGISTER for URI sip:6563630554@192.168.1.105 333925:23:34.305 PWL: [CID=0x0000] -->> From: sip:6563630554@192.168.1.105 Target: REGISTER sip:6563630554@192.168.1.105:5060 333925:23:34.305 INF: [CID=0x0e6f] *** UPPER REGISTRATION ENABLE *** for sip:6563630554@192.168.1.105 333925:23:34.970 INF: [CID=0x0e6f] Event: Setting UA Core [Proxy] to handle event REGISTER 333925:23:34.975 INF: [CID=0x0e6f] *** DESTROYED *** PROXY Session MDZmNDAwNDI1NDEyYmYzNDc5MDhlMjZmYmE1OWJhZDk. 333925:23:34.976 INF: [CID=0x0e6f] PROXY: Session DESTROYED 333925:23:34.977 INF: [CID=0x0e6f] *** DESTROYED *** PROXY Session MDZmNDAwNDI1NDEyYmYzNDc5MDhlMjZmYmE1OWJhZDk. 333925:23:34.977 INF: [CID=0x0e6f] PROXY: Session DESTROYED 333925:23:35.194 INF: [CID=0x0e6f] Event: Setting UA Core [Proxy] to handle event REGISTER 333925:23:35.194 INF: [CID=0x0e6f] Session CREATED 333925:23:35.194 INF: [CID=0x0e6f] *** CREATED *** PROXY Session MDZmNDAwNDI1NDEyYmYzNDc5MDhlMjZmYmE1OWJhZDk. 333925:23:35.198 INF: [CID=0x0e6f] Routing REGISTER for URI sip:6563630554@192.168.1.105 333925:23:35.198 PWL: [CID=0x0000] -->> From: sip:6563630554@192.168.1.105 Target: REGISTER sip:6563630554@192.168.1.105:5060 333925:23:35.199 INF: [CID=0x0e6f] *** UPPER REGISTRATION ENABLE *** for sip:6563630554@192.168.1.105 333925:23:35.220 INF: [CID=0x0e6f] Event: Setting UA Core [Proxy] to handle event REGISTER 333925:23:35.439 INF: [CID=0x0e6f] Event: Setting UA Core [Proxy] to handle event REGISTER 333925:23:35.440 INF: [CID=0x0e6f] Session CREATED 333925:23:35.440 INF: [CID=0x0e6f] *** CREATED *** PROXY Session MDZmNDAwNDI1NDEyYmYzNDc5MDhlMjZmYmE1OWJhZDk. 333925:23:35.444 INF: [CID=0x0e6f] Routing REGISTER for URI sip:6563630554@192.168.1.105 333925:23:35.444 PWL: [CID=0x0000] -->> From: sip:6563630554@192.168.1.105 Target: REGISTER sip:6563630554@192.168.1.105:5060 333925:23:35.445 INF: [CID=0x0e6f] *** UPPER REGISTRATION ENABLE *** for sip:6563630554@192.168.1.105 333925:23:35.463 INF: [CID=0x0e6f] Event: Setting UA Core [Proxy] to handle event REGISTER 333925:23:35.467 INF: [CID=0x0e6f] *** DESTROYED *** PROXY Session MDZmNDAwNDI1NDEyYmYzNDc5MDhlMjZmYmE1OWJhZDk. 333925:23:35.467 INF: [CID=0x0e6f] PROXY: Session DESTROYED 333925:23:35.468 INF: [CID=0x0e6f] *** DESTROYED *** PROXY Session MDZmNDAwNDI1NDEyYmYzNDc5MDhlMjZmYmE1OWJhZDk. 333925:23:35.468 INF: [CID=0x0e6f] PROXY: Session DESTROYED 333925:23:35.682 INF: [CID=0x0e6f] Event: Setting UA Core [Proxy] to handle event REGISTER 333925:23:35.683 INF: [CID=0x0e6f] Session CREATED 333925:23:35.683 INF: [CID=0x0e6f] *** CREATED *** PROXY Session MDZmNDAwNDI1NDEyYmYzNDc5MDhlMjZmYmE1OWJhZDk. 333925:23:35.691 INF: [CID=0x0e6f] Routing REGISTER for URI sip:6563630554@192.168.1.105 333925:23:35.692 PWL: [CID=0x0000] -->> From: sip:6563630554@192.168.1.105 Target: REGISTER sip:6563630554@192.168.1.105:5060 333925:23:35.692 INF: [CID=0x0e6f] *** UPPER REGISTRATION ENABLE *** for sip:6563630554@192.168.1.105 333925:23:35.694 INF: [CID=0x0e66] Event: Setting UA Core [Proxy] to handle event REGISTER 333925:23:35.695 INF: [CID=0x0e66] Session CREATED 333925:23:35.696 INF: [CID=0x0e66] *** CREATED *** PROXY Session MjliMjEwMGU3ZmM3YTQ3MTJiOGJlZGY0MjYzZDdkOTA. Best regards, Yuriy. |
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From: Joegen E. B. <joe...@gm...> - 2008-02-02 12:50:53
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sales@ER wrote: > Hi Joegen > > Is there a relatonship between the Media trunk port setting in the Transport > Status and the the General parameters Stack RPT Media Address in ? No. RTP Media address started as a hack around hosted servers that has a private IP address but with a 1 to 1 public mapping. One good example is the amazon cloud. > I > noticed that Main, Backup, CALEA, SIP Trunk and Media are all running on > differenct ports. Is there a doc that explaiins this thinking? > Each user agent has it's own transport. Since these UA behave differently than the other, we can't really mix messages using a single transport and multiplex each message and sort them according to UA assignment. I am not saying it's an impossibility. I am saying the job of the multiplexer will be heavy. Thus we arrived at a design decision to use one transport per specialized UA. > Warren Kreckler > > > > ----- Original Message ----- > From: "Joegen E. Baclor" <joe...@gm...> > To: "syfaure" <sy...@gm...> > Cc: <ope...@li...> > Sent: Monday, January 28, 2008 7:19 PM > Subject: Re: [OpenSBC] OpenSBC Registration > > > >> If you are expecting an answer as to why you are not successful with the >> call, you need to provide more information like version of the opensbc, >> Configuration and Level 5 logs. Packet captures are always nice. >> >> Joegen >> >> syfaure wrote: >> >>> Hi, >>> >>> I try to implement a OpenSBC in an IMS core like >>> http://www.openimscore.org/. >>> >>> I can register my client on the IMS but It's impossible to create a >>> > call. > >>> Kind regards, >>> >>> >>> >>> >>> >>> >>> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by: Microsoft >> Defy all challenges. Microsoft(R) Visual Studio 2008. >> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >> _______________________________________________ >> Opensipstack-osbcdevel mailing list >> Ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel >> >> > > > > > |
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From: Joegen E. B. <joe...@gm...> - 2008-02-02 12:46:01
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I am guessing. Your server will accept multiple bindings with the same AOR . Based on the configuration, the server may either fork using a ring-all approach or serially using forked fail-over. With OpenSBC in the middle, OpenSBC will change the binding to point to itself. Thus, your server will see _the same binding_ (OpenSBC Address) for the multiple registrations and it would think that this is the same device and overwrites the registration record. I am sorry to say that this setup will not work if this is indeed how your server behaves. Joegen Harley Stack wrote: > I am trying to set up OpenSBC as a proxy for registered SIP clients against our MetaSwitch. I have been successful for the most part. SIP users are able to register to the upstream proxy. Invites and Subscribe events are being proxied correctly and end users are able to make calls. > > > > The issue we are having is implementing SIP forking. The last software release for the MetaSwitch call agents has a new feature they name Shared Line Appearance. In short it is SIP forking with presence information. We have been able to get this working with Snom 360 IP phones (running v7.1.30) when they directly register to the MetaSwitch. However when we attempt to register through our instance of OpenSBC the last phone that registers to the shared line is the only one that get's Invites for incoming calls. > > > > We are running OpenSBC 1.1.5 and OpenSIPStack 1.1.7 that was pulled from CVS on Jan 31, 2008. > > > Any advice on this would be greatly appreciated. > > > Thanks > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Opensipstack-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > |
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From: Harley S. <hs...@ut...> - 2008-02-01 23:29:49
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I am trying to set up OpenSBC as a proxy for registered SIP clients against our MetaSwitch. I have been successful for the most part. SIP users are able to register to the upstream proxy. Invites and Subscribe events are being proxied correctly and end users are able to make calls. The issue we are having is implementing SIP forking. The last software release for the MetaSwitch call agents has a new feature they name Shared Line Appearance. In short it is SIP forking with presence information. We have been able to get this working with Snom 360 IP phones (running v7.1.30) when they directly register to the MetaSwitch. However when we attempt to register through our instance of OpenSBC the last phone that registers to the shared line is the only one that get's Invites for incoming calls. We are running OpenSBC 1.1.5 and OpenSIPStack 1.1.7 that was pulled from CVS on Jan 31, 2008. Any advice on this would be greatly appreciated. Thanks |
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From: <sa...@ER...> - 2008-01-31 21:19:47
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Hi Joegen Is there a relatonship between the Media trunk port setting in the Transport Status and the the General parameters Stack RPT Media Address in ? I noticed that Main, Backup, CALEA, SIP Trunk and Media are all running on differenct ports. Is there a doc that explaiins this thinking? Warren Kreckler ----- Original Message ----- From: "Joegen E. Baclor" <joe...@gm...> To: "syfaure" <sy...@gm...> Cc: <ope...@li...> Sent: Monday, January 28, 2008 7:19 PM Subject: Re: [OpenSBC] OpenSBC Registration > If you are expecting an answer as to why you are not successful with the > call, you need to provide more information like version of the opensbc, > Configuration and Level 5 logs. Packet captures are always nice. > > Joegen > > syfaure wrote: > > Hi, > > > > I try to implement a OpenSBC in an IMS core like > > http://www.openimscore.org/. > > > > I can register my client on the IMS but It's impossible to create a call. > > > > Kind regards, > > > > > > > > > > > > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Opensipstack-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > |
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From: Ilian J. C. P. <ip...@so...> - 2008-01-31 03:49:25
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Hi, I think you're talking about ATLSIP so we're in the wrong thread. Anyway, in SoftPhoneManager::Initialise() change PTrace::SetLevel( 3 ); to PTrace::SetLevel( 5 );. The logs will be dumped in sip.log. Regards, Ilian Hikmet Cengiz wrote: > How can we get the level 5 trace under windows environment? > > > All I can see is the sip.log file and the event related messages dumped to the command window. (Referencing ATLSIPLib.dll in c# console application) I'm using Win XP and VStudio 2005. > > > Thanks much. > > > > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Opensipstack-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > |
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From: Hikmet C. <hik...@ve...> - 2008-01-30 19:10:44
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How can we get the level 5 trace under windows environment? All I can see is the sip.log file and the event related messages dumped to the command window. (Referencing ATLSIPLib.dll in c# console application) I'm using Win XP and VStudio 2005. Thanks much. |
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From: Joegen E. B. <joe...@gm...> - 2008-01-29 01:19:55
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If you are expecting an answer as to why you are not successful with the call, you need to provide more information like version of the opensbc, Configuration and Level 5 logs. Packet captures are always nice. Joegen syfaure wrote: > Hi, > > I try to implement a OpenSBC in an IMS core like > http://www.openimscore.org/. > > I can register my client on the IMS but It's impossible to create a call. > > Kind regards, > > > > > > |
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From: syfaure <sy...@gm...> - 2008-01-28 21:26:43
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Hi, I try to implement a OpenSBC in an IMS core like http://www.openimscore.org/. I can register my client on the IMS but It's impossible to create a call. Kind regards, |
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From: Joegen E. B. <joe...@gm...> - 2008-01-25 03:15:03
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Make sure you have a <root> </root> enclosing your XML adams wrote: > Hi, > > I'm trying to setup OpenSBC as a B2BUA between a sip provider (voipchea= p) and sipX. > > I configured OpenSBC using the manual in [http://www.opensourcesip.org:= 8080/clearspacex/docs/DOC-1040|d-1040] but OpenSBC doesn't register. > When the requests arrive the OpenSBC, it starts sending INVITEs before = registering with the provider. > > I can't get the configuration right. > This is my current configuration: > > OpenSBC General Parameters > {code:java} > SBC Mode: B2B Only Mode > {code} > {code:java} > Interface Address: sip:*:9000 > {code} > > Trusted Domains > {code:java} > Accept All Calls: V > {code} > > Local Domain Accounts > {code:java} > Accept All Registration: V > {code} > > B2BUA Routes=20 > {code:java} > Route: [sip:*] sip:voipcheap.com > {code} > > SIP Trunk Config: > =ABsiptrunk trunk-name=3D"voipcheap.com" route-set=3D"sip.voipcheap.com= " sip-domain=3D"voipcheap.com" expires=3D"10"=BB > =ABtrunk-accounts=BB > =ABaccount user-name=3D"**username**" auth-user-name=3D"**username**" a= uth-password=3D"**password**" inboundroute=3D"sip:XXX@**internal.sip.doma= in**" expires=3D"3600" /=BB > =AB/trunk-accounts=BB > =AB/siptrunk=BB > > > =20 >> the sip trunk config is correctly escaped with the xml characters inst= ead of « and », I couldn't paste here those characters=20 >> =20 > > Thank you > > adams > > -----------------------------------------------------------------------= -- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Opensipstack-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > =20 |
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From: adams <ada...@gm...> - 2008-01-24 19:28:26
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Hi, I'm trying to setup OpenSBC as a B2BUA between a sip provider (voipcheap) a= nd sipX. I configured OpenSBC using the manual in [http://www.opensourcesip.org:8080= /clearspacex/docs/DOC-1040|d-1040] but OpenSBC doesn't register. When the requests arrive the OpenSBC, it starts sending INVITEs before regi= stering with the provider. I can't get the configuration right. This is my current configuration: OpenSBC General Parameters {code:java} SBC Mode: B2B Only Mode {code} {code:java} Interface Address: sip:*:9000 {code} Trusted Domains {code:java} Accept All Calls: V {code} Local Domain Accounts {code:java} Accept All Registration: V {code} B2BUA Routes=20 {code:java} Route: [sip:*] sip:voipcheap.com {code} SIP Trunk Config: =ABsiptrunk trunk-name=3D"voipcheap.com" route-set=3D"sip.voipcheap.com" si= p-domain=3D"voipcheap.com" expires=3D"10"=BB =ABtrunk-accounts=BB =ABaccount user-name=3D"**username**" auth-user-name=3D"**username**" auth-= password=3D"**password**" inboundroute=3D"sip:XXX@**internal.sip.domain**" = expires=3D"3600" /=BB =AB/trunk-accounts=BB =AB/siptrunk=BB > the sip trunk config is correctly escaped with the xml characters instead= of « and », I couldn't paste here those characters=20 Thank you adams |
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From: Joegen E. B. <joe...@gm...> - 2008-01-18 04:05:04
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The log indicates that everything is well. There are a lot of things that can go wrong with audio which are outside of OpenSBC control. What you can do to verify where the fault is, is to get your favorite packet sniffer out of the tool box and check whether both your phones are sending RTP and if OpenSBC are relaying the packets correctly and reaching the destination. Another cause of audio problems are codec related issues. I can see that your phone negotiated G.729. Can you test jsut using plain old G.711 and see if you get bidirectional audio. If you do, then you might want to check the TX packetization interval of G.729 on both your phones to properly match. 20 ms is a good ptime value for G.729. Jonas Gauffin wrote: > I'm not really sure what you mean by "media anchor". > > I got 2 phones and an opensbc on my private lan, then I have a router > which forwards sip/rtp to opensbc. > When I call from one of the phones to the other (through opensbc) I > get audio one way only (check the attached log) > > As for external calls, I'm willing to do a "hack" to get it working. I > guess it involves modifying the IP in SDP, Via header and Contact > header. > Some directions would be really appreciated. > > On Jan 17, 2008 9:28 AM, Joegen E. Baclor <joe...@gm...> wrote: > >> Jonas Gauffin wrote: >> >>> Hello >>> >>> I got a opensbc listening on it's local ip address (192.168.1.102). >>> How do I make opensbc handle my domain (let's say "domain.com")? >>> >>> >> Wait. Are you trying to use OpenSBC as a media anchor from within your >> NAT? If that's what you are intending to do, then I hate to disappoint >> you that this is not possible with OpenSBC. OpenSBC needs to be in a >> public space for it to sucessfully anchor media. >> >> >> >>> Regards, >>> Jonas >>> >>> ------------------------------------------------------------------------- >>> This SF.net email is sponsored by: Microsoft >>> Defy all challenges. Microsoft(R) Visual Studio 2008. >>> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >>> _______________________________________________ >>> Opensipstack-osbcdevel mailing list >>> Ope...@li... >>> https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel >>> >>> >>> >>> >> ------------------------------------------------------------------------- >> This SF.net email is sponsored by: Microsoft >> Defy all challenges. Microsoft(R) Visual Studio 2008. >> http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ >> _______________________________________________ >> Opensipstack-osbcdevel mailing list >> Ope...@li... >> https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel >> >> >> ------------------------------------------------------------------------ >> >> No virus found in this incoming message. >> Checked by AVG Free Edition. >> Version: 7.5.516 / Virus Database: 269.19.5/1228 - Release Date: 1/16/2008 9:01 AM |
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From: Jonas G. <jon...@gm...> - 2008-01-17 08:47:35
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I'm not really sure what you mean by "media anchor". I got 2 phones and an opensbc on my private lan, then I have a router which forwards sip/rtp to opensbc. When I call from one of the phones to the other (through opensbc) I get audio one way only (check the attached log) As for external calls, I'm willing to do a "hack" to get it working. I guess it involves modifying the IP in SDP, Via header and Contact header. Some directions would be really appreciated. On Jan 17, 2008 9:28 AM, Joegen E. Baclor <joe...@gm...> wrote: > Jonas Gauffin wrote: > > Hello > > > > I got a opensbc listening on it's local ip address (192.168.1.102). > > How do I make opensbc handle my domain (let's say "domain.com")? > > > > > Wait. Are you trying to use OpenSBC as a media anchor from within your > NAT? If that's what you are intending to do, then I hate to disappoint > you that this is not possible with OpenSBC. OpenSBC needs to be in a > public space for it to sucessfully anchor media. > > > > > > Regards, > > Jonas > > > > ------------------------------------------------------------------------- > > This SF.net email is sponsored by: Microsoft > > Defy all challenges. Microsoft(R) Visual Studio 2008. > > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > > _______________________________________________ > > Opensipstack-osbcdevel mailing list > > Ope...@li... > > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > > > > > > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Opensipstack-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > |
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From: Joegen E. B. <joe...@gm...> - 2008-01-17 08:28:24
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Jonas Gauffin wrote: > Hello > > I got a opensbc listening on it's local ip address (192.168.1.102). > How do I make opensbc handle my domain (let's say "domain.com")? > Wait. Are you trying to use OpenSBC as a media anchor from within your NAT? If that's what you are intending to do, then I hate to disappoint you that this is not possible with OpenSBC. OpenSBC needs to be in a public space for it to sucessfully anchor media. > > Regards, > Jonas > > ------------------------------------------------------------------------- > This SF.net email is sponsored by: Microsoft > Defy all challenges. Microsoft(R) Visual Studio 2008. > http://clk.atdmt.com/MRT/go/vse0120000070mrt/direct/01/ > _______________________________________________ > Opensipstack-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > |
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