opensipstack-osbcdevel Mailing List for OpenSIPStack (Page 7)
Brought to you by:
joegenbaclor
Menu
▾
▴
opensipstack-osbcdevel — Discussions on OpenSBC
You can subscribe to this list here.
| 2007 |
Jan
|
Feb
|
Mar
|
Apr
|
May
|
Jun
|
Jul
|
Aug
(22) |
Sep
(29) |
Oct
(19) |
Nov
(33) |
Dec
(92) |
|---|---|---|---|---|---|---|---|---|---|---|---|---|
| 2008 |
Jan
(31) |
Feb
(24) |
Mar
(54) |
Apr
(59) |
May
(31) |
Jun
(22) |
Jul
(32) |
Aug
(19) |
Sep
(49) |
Oct
(41) |
Nov
(84) |
Dec
(19) |
| 2009 |
Jan
(64) |
Feb
(37) |
Mar
(20) |
Apr
(5) |
May
(2) |
Jun
|
Jul
(3) |
Aug
(7) |
Sep
(3) |
Oct
|
Nov
|
Dec
|
| 2011 |
Jan
|
Feb
|
Mar
|
Apr
(1) |
May
|
Jun
|
Jul
|
Aug
|
Sep
|
Oct
|
Nov
|
Dec
|
| 2012 |
Jan
|
Feb
|
Mar
|
Apr
|
May
(1) |
Jun
|
Jul
|
Aug
|
Sep
|
Oct
|
Nov
|
Dec
|
|
From: OpenSBC F. <ope...@op...> - 2008-12-23 06:26:17
|
I've just put OpenSBC 1.1.5 RC3 between sipXecs 3.10.2 and our SIP Trunk because the Trunk won't handle REFERs -- causing the auto-attendent to not function properly for calls originating from the PSTN (transfers fail). With OpenSBC in place, set to handle REFERs locally, attended transfers now work. But once OpenSBC gets the REFER and attempts to connect to the new number, it doesn't generate a "ringing" sound, even though its receiving 180 Ringing messages from sipXecs. Is this an expected behavior for OpenSBC, or is there something I haven't configured? I've looked around for a media file that would be played by OpenSBC to produce the ring, but I couldn't find anything. Any thoughts would be appreciated! |
|
From: OpenSBC F. <ope...@op...> - 2008-12-15 02:21:51
|
Yes, It works. Thanks so much! :p |
|
From: OpenSBC F. <ope...@op...> - 2008-12-13 03:16:15
|
Yest, most definitely. Start with a fresh config of OpenSBC (just delete your OpenSBC_data/opensbc.ini) and follow these steps. 1. Set General Parameters/SBC-Application-Mode to "B2BUpperReg Mode" then restart OpenSBC after you click Update 2. Add [ sip:*@ims.domain.com ] sip:10.0.0.3 in Outbound-Proxies/Outbound-Proxies settings. (This will force all requests bound for ims.domain.com to traverse your proxy) 3. Put a check on B2BUA-Routes/Route-By-Request-URI That's it. If your invite looks as follows, it will reach the IMS network. This assumes that OpenSBC is treated as if its a normal OB proxy by the calling UA. You can't treat it as if it's a GW or this scheme won't work. INVITE sip:us...@im... SIP/2.0 To: <us...@im...>;tag=123 Route: <10.0.0.2;lr> HTH, Joegen > {quote:title=baolovebao wrote:}{quote} > > Hi all, > > I have a use case that require the OSBC as a proxy between the UA and the other IMS network. The diagram is like below: > > UA <--------->OSBC<------------------>Outbound Proxy<-------------->IMS network.(ims.domain.com) > > 10.0.0.1 10.0.0.2 10.0.0.3 anything else > > As I think the PROXY mode can meet this use case, but unlucky it didn't. > > some configurations: > > UA: Server: ims.domain.com > > outbound proxy: 10.0.0.2 > > OSBC: Proxy Mode > > Proxy-Relay-Routes: [sip:*@ims.domain.com:*]sip:ims.domain.com:5060, 10.0.0.3:5060 > > Internal-DNS-Mapping : [sip:ims.domain.com]sip:10.0.0.3 > > but after the UA send the REGISTER to the OSBC, it doesn't relay it to the 10.0.0.3. > > Anything can be done more? Or any advices and proposal? Thanks in advance! > > {quote:title=baolovebao wrote:}{quote} > > Hi all, > > I have a use case that require the OSBC as a proxy between the UA and the other IMS network. The diagram is like below: > > UA <--------->OSBC<------------------>Outbound Proxy<-------------->IMS network.(ims.domain.com) > > 10.0.0.1 10.0.0.2 10.0.0.3 anything else > > As I think the PROXY mode can meet this use case, but unlucky it didn't. > > some configurations: > > UA: Server: ims.domain.com > > outbound proxy: 10.0.0.2 > > OSBC: Proxy Mode > > Proxy-Relay-Routes: [sip:*@ims.domain.com:*]sip:ims.domain.com:5060, 10.0.0.3:5060 > > Internal-DNS-Mapping : [sip:ims.domain.com]sip:10.0.0.3 > > > but after the UA send the REGISTER to the OSBC, it doesn't relay it to the 10.0.0.3. > > > Anything can be done more? Or any advices and proposal? Thanks in advance! > > |
|
From: OpenSBC F. <ope...@op...> - 2008-12-12 09:33:30
|
Hi all, I have a use case that require the OSBC as a proxy between the UA and the other IMS network. The diagram is like below: UA <--------->OSBC<------------------>Outbound Proxy<-------------->IMS network.(ims.domain.com) 10.0.0.1 10.0.0.2 10.0.0.3 anything else As I think the PROXY mode can meet this use case, but unlucky it didn't. some configurations: UA: Server: ims.domain.com outbound proxy: 10.0.0.2 OSBC: Proxy Mode Proxy-Relay-Routes: [sip:*@ims.domain.com:*]sip:ims.domain.com:5060, 10.0.0.3:5060 Internal-DNS-Mapping : [sip:ims.domain.com]sip:10.0.0.3 but after the UA send the REGISTER to the OSBC, it doesn't relay it to the 10.0.0.3. Anything can be done more? Or any advices and proposal? Thanks in advance! |
|
From: Erik L. (ext) <eri...@aa...> - 2008-12-04 13:44:21
|
Hi! Thanks for the answer. When it is planned to be implemented? /Erik ________________________________ Från: Joegen E. Baclor [mailto:joe...@gm...] Skickat: to 2008-12-04 13:49 Till: ope...@li... Ämne: Re: [OpenSBC] 3PCC in OpenSBC Erik, Sorry, 3PCC support is not complete in OpenSBC. Joegen Erik Larsson (ext) wrote: > Hi! > How does it work to implement 3PCC in openSBC? How do I set up call > legs? I am trying to make a test application that reads phone numbers > from a database and let openSBC set up the calls. > > /Erik > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > Opensipstack-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > ------------------------------------------------------------------------ > > > No virus found in this incoming message. > Checked by AVG - http://www.avg.com <http://www.avg.com/> > Version: 8.0.176 / Virus Database: 270.9.13/1827 - Release Date: 12/3/2008 5:41 PM > > ------------------------------------------------------------------------- This SF.Net email is sponsored by the Moblin Your Move Developer's challenge Build the coolest Linux based applications with Moblin SDK & win great prizes Grand prize is a trip for two to an Open Source event anywhere in the world http://moblin-contest.org/redirect.php?banner_id=100&url=/ _______________________________________________ Opensipstack-osbcdevel mailing list Ope...@li... https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel |
|
From: Joegen E. B. <joe...@gm...> - 2008-12-04 12:49:42
|
Erik, Sorry, 3PCC support is not complete in OpenSBC. Joegen Erik Larsson (ext) wrote: > Hi! > How does it work to implement 3PCC in openSBC? How do I set up call > legs? I am trying to make a test application that reads phone numbers > from a database and let openSBC set up the calls. > > /Erik > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > This SF.Net email is sponsored by the Moblin Your Move Developer's challenge > Build the coolest Linux based applications with Moblin SDK & win great prizes > Grand prize is a trip for two to an Open Source event anywhere in the world > http://moblin-contest.org/redirect.php?banner_id=100&url=/ > ------------------------------------------------------------------------ > > _______________________________________________ > Opensipstack-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > ------------------------------------------------------------------------ > > > No virus found in this incoming message. > Checked by AVG - http://www.avg.com > Version: 8.0.176 / Virus Database: 270.9.13/1827 - Release Date: 12/3/2008 5:41 PM > > |
|
From: Erik L. (ext) <eri...@aa...> - 2008-12-04 09:23:48
|
Hi! How does it work to implement 3PCC in openSBC? How do I set up call legs? I am trying to make a test application that reads phone numbers from a database and let openSBC set up the calls. /Erik |
|
From: OpenSBC F. <ope...@op...> - 2008-12-04 05:19:56
|
hi, I would assume that this is from PSTN --> OSBC --> Sipx, if this is the case, point your DID number to your OSBC then from OSBC create a route going to sipx. ex. DID number(123456789) --> OSBC (route would be, [sip:123456789*] sip:10...@si... ) -->SipX with the example above, when an incoming call is received by the DID number, it will go to osbc then applied the specified route,which is going to sipx AA (as implied by 100 ext), raymund |
|
From: OpenSBC F. <ope...@op...> - 2008-12-04 01:44:00
|
Not sure I got you correctly so feel free to correct me if I am wrong. Your SoftSwitch complains about CRLF header termination? If so then it's not compliant to the specs. As per 3261 ABNF for SIP Messages
---
generic-message = start-line
*message-header
CRLF
[ message-body ]
start-line = Request-Line / Status-Line
The start-line, each message-header line, and the empty line MUST be
terminated by a carriage-return line-feed sequence (CRLF). Note that
the empty line MUST be present even if the message-body is not.
---
What terminating character is your SoftSwitch expecting?
Joegen
> {quote:title=smitrob wrote:}{quote}
>
> I am having a problem with the content in the remote party ID. For a 180 ringing message my system sends the following for Remote-Party-ID to the OpenSBC
>
> bq. {color:#0000ff}Remote-Party-ID:"Test 2"<sip:3035550996@198.133.219.25;user=phone>;screen=yes;party=called;privacy=off;id-type=subscriber{color}
>
> However the OpenSBC sends the following.
>
> bq. {color:#0000ff}Remote-Party-ID:"360 Test 2"<sip:3033300996@64.15.100.52;user=phone>;screen=yes;party=called;privacy=off;id-type=subscriber{color}{color:#ff0000}\r\n{color}
>
> This is according to a tshark trace on the SBC. The same is true for all messages sent in the Message Body (i.e. RPID-Privacy, Content-Length)
>
>
> Is there a reason for this?
>
>
> I am assuming that the \r\n is a Carrage Return & Line Feed is this correct? Is there a way to change this? The SBC that I am sending to see this as a malformed message.
>
>
> This is running on a Linux System (CentOS) and is running OpenSBC v1.1.5-36
>
|
|
From: OpenSBC F. <ope...@op...> - 2008-12-03 22:17:58
|
I am having a problem with the content in the remote party ID. For a 180 ringing message my system sends the following for Remote-Party-ID to the OpenSBC
bq. {color:#0000ff}Remote-Party-ID:"Test 2"<sip:3035550996@198.133.219.25;user=phone>;screen=yes;party=called;privacy=off;id-type=subscriber{color}
However the OpenSBC sends the following.
bq. {color:#0000ff}Remote-Party-ID:"360 Test 2"<sip:3033300996@64.15.100.52;user=phone>;screen=yes;party=called;privacy=off;id-type=subscriber{color}{color:#ff0000}\r\n{color}
This is according to a tshark trace on the SBC. The same is true for all messages sent in the Message Body (i.e. RPID-Privacy, Content-Length)
Is there a reason for this?
I am assuming that the \r\n is a Carrage Return & Line Feed is this correct? Is there a way to change this? The SBC that I am sending to see this as a malformed message.
This is running on a Linux System (CentOS) and is running OpenSBC v1.1.5-36
|
|
From: OpenSBC F. <ope...@op...> - 2008-11-25 22:29:53
|
Greetings - How do I configure things to get inbound calls on a SipXecs/OpenSBC box? I've got SipXecs and OpenSBC on the same box with one NIC using Upper Registration and I am able to make outgoing calls. However, I'm not able to take incoming calls. Judging from the logs, the incoming calls never touch OpenSBC and go straight to SipXecs looking like 999...@ex...dress and SipXecs is unable to handle them properly. Is there something I can do to force the calls to OpenSBC first? Thanks. radp1399 |
|
From: Joegen E. B. <joe...@gm...> - 2008-11-25 16:22:04
|
Hi Vitaly, The error you are seeing is SIP/2.0 500 Maximum Concurrent Connection Reached is pretty much misleading. I'll change this in the future. The real meaning of this error for trunk calls is that OpenSBC was not able to map an account to the inbound call. Based on this error response, the domain is not correct. SIP/2.0 500 Maximum Concurrent Connection Reached From: <sip:044...@21...>;tag=ffcc5d70113083cd38912c3472e27a3a To: <sip:414...@21...>;tag=44b43a0372b9dd11870fd91503730f87 As you can see, it is sending to 414...@21.... if you configured your trunk like this SIP-Trunk-Config=<siptrunk trunk-name="voipgateway.org" route-set="voipgateway.org" sip-domain="voipgateway.org" expires="20"> Then the INVITE should be INVITE sip:41445551800ip.add.of.sbc:5066 SIP/2.0 From: <sip:044...@vo...>;tag=ffcc5d70113083cd38912c3472e27a3a To: <sip:414...@vo...>;tag=44b43a0372b9dd11870fd91503730f87 HTH, Joegen OpenSBC Forum wrote: > Hi, Joegen > > > I've installed OpenSBC v.1.1.5 RC2 on another server with ext.ip. > > > When I call from PSTN number 448755540 to number 41445551800 I got the next result > > > 2008/11/25 17:18:42.257 ERR: [CID=0x0000] GC: CallSessionManager.cxx:535 CallSessionManager::OnCreateServerSession::CallSession Attempt to CreateReference() a NULL Pointer or none descendant of PObject!!! > 2008/11/25 17:18:42.257 DBG: [CID=0x06cb] *** MESSAGE ARRIVAL *** No Session available to handle INVITE sip:41445001800@my_ext_ip:9006;transport=udp SIP/2.0 > 2008/11/25 17:18:42.258 DTL: [CID=0x0bfd] IST(1227626196313) Event(SIPMessage) - SIP/2.0 500 Maximum Concurrent Connection Reached > 2008/11/25 17:18:42.258 DBG: [CID=0x0bfd] TRANSACTION: (IST) SIP/2.0 500 Maximum Concurrent Connection Reached State: 2 > 2008/11/25 17:18:42.258 DTL: [CID=0x0bfd] IST(1227626196313) StateProceeding->StateCompleted(SIP/2.0 500 Maximum Concurrent Connection Reached) > 2008/11/25 17:18:42.258 DBG: [CID=0x0bfd] IST(1227626196313) Timer H( 32000 ms ) STARTED > 2008/11/25 17:18:42.258 DBG: [CID=0x0bfd] IST(1227626196313) Timer G( 500 ms ) STARTED > 2008/11/25 17:18:42.258 DBG: [CID=0x0bfd] Added ACK Transaction 279DF7A7-BA3B11DD-9734EA9F-DB957415@212.117.200.79~1o44b43a0372b9dd11870fd91503730f87|z9hG4bKc9c4.2b3ee7949af4ab8a4cc33ef47f04bf57.0|ACK > 2008/11/25 17:18:42.259 INF: [CID=0x0bfd] >>> SIP/2.0 500 Maximum Concurrent Connection Reached DST: 212.117.200.148:5060:UDP src=my_ext_ip:9007 enc=0 bytes=608 > 2008/11/25 17:18:42.259 DBG: [CID=0x0bfd] > 2008/11/25 17:18:42.259 DBG: [CID=0x0bfd] SIP/2.0 500 Maximum Concurrent Connection Reached > 2008/11/25 17:18:42.259 DBG: [CID=0x0bfd] From: <sip:0448755540@212.117.200.148>;tag=ffcc5d70113083cd38912c3472e27a3a > 2008/11/25 17:18:42.259 DBG: [CID=0x0bfd] To: <sip:41445551800@212.117.200.148>;tag=44b43a0372b9dd11870fd91503730f87 > 2008/11/25 17:18:42.259 DBG: [CID=0x0bfd] Via: SIP/2.0/UDP 212.117.200.148;branch=z9hG4bKc9c4.2b3ee7949af4ab8a4cc33ef47f04bf57.0;rport=5060;received=212.117.200.148 > 2008/11/25 17:18:42.259 DBG: [CID=0x0bfd] Via: SIP/2.0/UDP 212.117.200.148:5061;branch=z9hG4bKb14092bed380e45e3f6003f979886bf3;rport=5061 > 2008/11/25 17:18:42.259 DBG: [CID=0x0bfd] CSeq: 200 INVITE > 2008/11/25 17:18:42.259 DBG: [CID=0x0bfd] Call-ID: 279DF7A7-BA3B11DD-9734EA9F-DB957415@212.117.200.79~1o > 2008/11/25 17:18:42.259 DBG: [CID=0x0bfd] Record-Route: <sip:212.117.200.148;ftag=ffcc5d70113083cd38912c3472e27a3a;lr> > 2008/11/25 17:18:42.259 DBG: [CID=0x0bfd] Content-Length: 0 > > > What did I confuse? > |
|
From: OpenSBC F. <ope...@op...> - 2008-11-25 15:42:31
|
Hi, Joegen I've installed OpenSBC v.1.1.5 RC2 on another server with ext.ip. When I call from PSTN number 448755540 to number 41445551800 I got the next result 2008/11/25 17:18:42.257 ERR: [CID=0x0000] GC: CallSessionManager.cxx:535 CallSessionManager::OnCreateServerSession::CallSession Attempt to CreateReference() a NULL Pointer or none descendant of PObject!!! 2008/11/25 17:18:42.257 DBG: [CID=0x06cb] *** MESSAGE ARRIVAL *** No Session available to handle INVITE sip:41445001800@my_ext_ip:9006;transport=udp SIP/2.0 2008/11/25 17:18:42.258 DTL: [CID=0x0bfd] IST(1227626196313) Event(SIPMessage) - SIP/2.0 500 Maximum Concurrent Connection Reached 2008/11/25 17:18:42.258 DBG: [CID=0x0bfd] TRANSACTION: (IST) SIP/2.0 500 Maximum Concurrent Connection Reached State: 2 2008/11/25 17:18:42.258 DTL: [CID=0x0bfd] IST(1227626196313) StateProceeding->StateCompleted(SIP/2.0 500 Maximum Concurrent Connection Reached) 2008/11/25 17:18:42.258 DBG: [CID=0x0bfd] IST(1227626196313) Timer H( 32000 ms ) STARTED 2008/11/25 17:18:42.258 DBG: [CID=0x0bfd] IST(1227626196313) Timer G( 500 ms ) STARTED 2008/11/25 17:18:42.258 DBG: [CID=0x0bfd] Added ACK Transaction 279DF7A7-BA3B11DD-9734EA9F-DB957415@212.117.200.79~1o44b43a0372b9dd11870fd91503730f87|z9hG4bKc9c4.2b3ee7949af4ab8a4cc33ef47f04bf57.0|ACK 2008/11/25 17:18:42.259 INF: [CID=0x0bfd] >>> SIP/2.0 500 Maximum Concurrent Connection Reached DST: 212.117.200.148:5060:UDP src=my_ext_ip:9007 enc=0 bytes=608 2008/11/25 17:18:42.259 DBG: [CID=0x0bfd] 2008/11/25 17:18:42.259 DBG: [CID=0x0bfd] SIP/2.0 500 Maximum Concurrent Connection Reached 2008/11/25 17:18:42.259 DBG: [CID=0x0bfd] From: <sip:0448755540@212.117.200.148>;tag=ffcc5d70113083cd38912c3472e27a3a 2008/11/25 17:18:42.259 DBG: [CID=0x0bfd] To: <sip:41445551800@212.117.200.148>;tag=44b43a0372b9dd11870fd91503730f87 2008/11/25 17:18:42.259 DBG: [CID=0x0bfd] Via: SIP/2.0/UDP 212.117.200.148;branch=z9hG4bKc9c4.2b3ee7949af4ab8a4cc33ef47f04bf57.0;rport=5060;received=212.117.200.148 2008/11/25 17:18:42.259 DBG: [CID=0x0bfd] Via: SIP/2.0/UDP 212.117.200.148:5061;branch=z9hG4bKb14092bed380e45e3f6003f979886bf3;rport=5061 2008/11/25 17:18:42.259 DBG: [CID=0x0bfd] CSeq: 200 INVITE 2008/11/25 17:18:42.259 DBG: [CID=0x0bfd] Call-ID: 279DF7A7-BA3B11DD-9734EA9F-DB957415@212.117.200.79~1o 2008/11/25 17:18:42.259 DBG: [CID=0x0bfd] Record-Route: <sip:212.117.200.148;ftag=ffcc5d70113083cd38912c3472e27a3a;lr> 2008/11/25 17:18:42.259 DBG: [CID=0x0bfd] Content-Length: 0 What did I confuse? -- Regard, Vitaly |
|
From: OpenSBC F. <ope...@op...> - 2008-11-25 12:57:36
|
> {quote:title=Guest wrote:}{quote}
> Hi Joegen
>
> I am running 1.1.5-36 Compile date 21 Nov 2008. Is there a newer version of
> OpenSBC?
Yes there is a new version in CVS. I actually just committed the patch as a reaction to your query. Current version is 1.1.5-39
>
> Following the install CVS nethod from your website, I was only able to get
> the OpenSipStack NOT the Opensbc. I wrote you about this, asking how do i
> get the CVS of OpenSBC. Is there another website or method to get the code?
>
What's the actual problem? You should be able to get OpenSBC using.
cvs -z3 -d:pserver:ano...@op...:/cvsroot/opensipstack co -P opensbc
> r
>
> > ----- Original Message -----
> > From: "OpenSBC Forum" <ope...@op...>
> > To: <ope...@li...>
> > Sent: Monday, November 24, 2008 7:50 PM
> > Subject: Re: [OpenSBC] Fw: SipX<=>OpenSBC: Sip Trunk config
> >
> >
> > > username/password not required
> > > MIME-Version: 1.0
> > > Content-Type: text/plain; charset=ISO-8859-1
> > > Content-Transfer-Encoding: 7bit
> > >
> > > > {quote:title=Guest wrote:}{quote}
> > > > Hi Joegen
> > > >
> > > > I am using Broadvox for SIP Trunk ITSP orgination and termination. I
> > don't
> > > > need to send them a username and passwd. Can i still us Sip Trunk
> > > > configuration. It seems that the OSBC config requires this setup. Is
> > there
> > > > a way to turn off registration?
> > > >
> > > > r
> > > >
> > > >
> > >
> > > Robert,
> > >
> > > There is now. I have introduced a new attribute for sip-trunk named
> > send-reg. Just set it to "no" in you trunk account config.
> > >
> > > send-reg="no"
> > > auth-user-name="1003"
> > > auth-password="1003"
> > > inbound-route="sip:90...@wi..."
> > >
> > > Please upgrade using CVS head and let me know if the patch works for
> you.
> > >
> > > Joegen
> > >
> >
|
|
From: OpenSBC F. <ope...@op...> - 2008-11-25 12:50:16
|
> {quote:title=atomic wrote:}{quote}
>
> Hi Joegen,
>
>
> I can't understand what means "+*fails to refresh registration with tge upper-reg overtime*+". Can you point me to a discussion about this bug?
>
>
I am sorry, I was quick to conclude that you are suffering from the race condition in upper reg that causes refresh to lock up.
>
> I traced the call with wireshark and I see that opensbc sends the INVITE to the private IP and not to the public IP.
>
I doubt this very much. We have OpenSBC in production serving a multitude of NATed UAs. Are you using RFC 1918 compliant addressing like 10.0.0.1 or 192.168.0.1? If not then that's what causing OpenSBC to send the INVITE directly to the UA.
>
> If you need logs or wireshark trace I'll send to you.
>
Yes you can send the wireshark capture of the REGISTRATION and the INVITE attempts to jo...@op...
>
> Thank you very much.
>
>
> Regards.
>
>
> Antonio.
>
>
>
>
> > {quote:title=joegen wrote:}{quote}Hi Antonio,
> >
> > Yes, OpenSBC shows the actual binding sent in the contact in the registration status but it does send the INVITE to the public IP in cases where the phone is behind a NAT. There is currently a known bug in OpenSBC where it fails to refresh registrations with tge upper-reg overtime. I am currently developing a registration test tool just for this purpose. I will announce it as soon as a bug fix is committed.
> >
> > Joegen
> >
|
|
From: OpenSBC F. <ope...@op...> - 2008-11-25 09:14:58
|
Hi Joegen,
I can't understand what means "+*fails to refresh registration with tge upper-reg overtime*+". Can you point me to a discussion about this bug?
I traced the call with wireshark and I see that opensbc sends the INVITE to the private IP and not to the public IP.
If you need logs or wireshark trace I'll send to you.
Thank you very much.
Regards.
Antonio.
> {quote:title=joegen wrote:}{quote}Hi Antonio,
>
> Yes, OpenSBC shows the actual binding sent in the contact in the registration status but it does send the INVITE to the public IP in cases where the phone is behind a NAT. There is currently a known bug in OpenSBC where it fails to refresh registrations with tge upper-reg overtime. I am currently developing a registration test tool just for this purpose. I will announce it as soon as a bug fix is committed.
>
> Joegen
>
|
|
From: OpenSBC F. <ope...@op...> - 2008-11-25 01:50:49
|
username/password not required
MIME-Version: 1.0
Content-Type: text/plain; charset=ISO-8859-1
Content-Transfer-Encoding: 7bit
> {quote:title=Guest wrote:}{quote}
> Hi Joegen
>
> I am using Broadvox for SIP Trunk ITSP orgination and termination. I don't
> need to send them a username and passwd. Can i still us Sip Trunk
> configuration. It seems that the OSBC config requires this setup. Is there
> a way to turn off registration?
>
> r
>
>
Robert,
There is now. I have introduced a new attribute for sip-trunk named send-reg. Just set it to "no" in you trunk account config.
send-reg="no"
auth-user-name="1003"
auth-password="1003"
inbound-route="sip:90...@wi..."
Please upgrade using CVS head and let me know if the patch works for you.
Joegen
|
|
From: OpenSBC F. <ope...@op...> - 2008-11-24 23:38:13
|
Hi Antonio,
Yes, OpenSBC shows the actual binding sent in the contact in the registration status but it does send the INVITE to the public IP in cases where the phone is behind a NAT. There is currently a known bug in OpenSBC where it fails to refresh registrations with tge upper-reg overtime. I am currently developing a registration test tool just for this purpose. I will announce it as soon as a bug fix is committed.
Joegen
> {quote:title=atomic wrote:}{quote}
> Hi all,
>
> I configured opensbc in B2BUpperReg mode. No problem with registration and outgoing calls, but I have problem with incoming calls.
>
> In _OpenSBC Registration Status_ I see in the binding column the private IP address of the NATted client, so opensbc tries to send the incoming invite to that IP. Why opensbc doesn't rewrite the IP address? Is this the correct behaviour? I have missing something in the configuration?
>
> Thanks. Regards.
>
>
>
> Antonio.
|
|
From: OpenSBC F. <ope...@op...> - 2008-11-24 17:12:05
|
Hi all, I configured opensbc in B2BUpperReg mode. No problem with registration and outgoing calls, but I have problem with incoming calls. In [OpenSBC Registration Status|http://81.31.219.9:9999/OpenSBC%20Registration%20Status] I see in the binding column the private IP address of the NATted client, so opensbc tries to send the incoming invite to that IP. Why opensbc doesn't rewrite the IP address? Is this the correct behaviour? I have missing something in the configuration? Thanks. Regards. Antonio. |
|
From: OpenSBC F. <ope...@op...> - 2008-11-21 21:51:41
|
Joegen, I sent the information as requested. Any further news regarding this? regards, Miced |
|
From: OpenSBC F. <ope...@op...> - 2008-11-21 02:34:09
|
Since OpenSBC does not transcode, there is no need for it to identify which codec is in effect for the session. There is also no such thing as enabling codecs in OpenSBC. It simply relays whatever codec is specified in SDP.
Joegen
> {quote:title=miced wrote:}{quote}
> Hello folks,
>
> Is there anyway I can check what codecs are enabled in OSBC and also which one is used during the call?
>
> regards,
>
> Miced
>
|
|
From: OpenSBC F. <ope...@op...> - 2008-11-21 02:30:57
|
Sorry, CPS throttling is not supported in OpenSBC. The nearest thing to that would be setting the value of Max-Concurrent-Session in General-Parameters
Joegen
> {quote:title=Imran wrote:}{quote}
> Hi Eric & Joegen,
>
>
> I created a SIP URI blocking list. Thanks for helping me in the right direction. We are setting up OpenSBC in an testenviroment to test its capabilities in front of a mediaservice (we are a SP).
> I figured out the most of the functionality that we need to test in OpenSBC. One functionalitiy i could not find tough.
>
> We need to filter the INVITE/sec and REGISTER/sec in an network on the OSSBC. Thus, users are not allowed to send, for example, more then 30 REGISTER/sec attempts to our SIP service (TROUGH the OPENSBC) to protect it for overload.
> Could one tell me if this functionalitiy is in OpenSBC and if yes where I could configure it?
>
> TiA.
>
> Regards,
> Imran
|
|
From: OpenSBC F. <ope...@op...> - 2008-11-20 23:21:27
|
Hello folks, Is there anyway I can check what codecs are enabled in OSBC and also which one is used during the call? regards, Miced |
|
From: OpenSBC F. <ope...@op...> - 2008-11-20 13:38:11
|
Hi Eric & Joegen, I created a SIP URI blocking list. Thanks for helping me in the right direction. We are setting up OpenSBC in an testenviroment to test its capabilities in front of a mediaservice (we are a SP). I figured out the most of the functionality that we need to test in OpenSBC. One functionalitiy i could not find tough. We need to filter the INVITE/sec and REGISTER/sec in an network on the OSSBC. Thus, users are not allowed to send, for example, more then 30 REGISTER/sec attempts to our SIP service (TROUGH the OPENSBC) to protect it for overload. Could one tell me if this functionalitiy is in OpenSBC and if yes where I could configure it? TiA. Regards, Imran |
|
From: OpenSBC F. <ope...@op...> - 2008-11-19 22:59:32
|
Hey Miced,
Can you grep the REGISTER session for this and send it to me by mail? Just grep the CID for the REGISTER. I have received a similar report about this behavior from a client of ours. It would be good if I can have separate info as to why wand when this occurs.
grep 0x0a90 b2bua.log > dest.log
You can send the log to me by mail at jo...@op...
Joegen
> {quote:title=miced wrote:}{quote}
> {quote:title=joegen wrote:}{quote}
> > > 2) I have 'B2BUpperRegMode' enabled and 'Accept-All-Registration' under Local-Domain-Accounts checked. Is this how you meant I should configure OSBC for normal users? The phone is showing as registered under OpenSBC Registration Status and also I can see it in Brekeke as below.
> > That should work.
> >
> > Joegen
> >
> I noticed that after an hour or so my phone seems to drop off the Registered Users page on Brekeke but I am able to make calls. I can see the phone is registered on the OSBC registration page. If I kill and restart OSBC and reboot then phone it shows again in Brekeke. Is there anything I can do to resolve this?
>
> regards,
>
> Miced
|
14 messages has been excluded from this view by a project administrator.
