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From: Mohan A. K. <mk...@ve...> - 2008-03-17 04:56:23
|
Hi Joegen, B2BUA is functioning fine and it has no problem in this issue. Only problem is with the proxy mode. Regards, P. Mohan Arun Kumar -----Original Message----- From: ope...@li... [mailto:ope...@li...] On Behalf Of Mohan Arun Kumar Sent: Friday, March 14, 2008 7:12 PM To: ope...@li... Subject: Re: [OpenSBC] OpenSBC in Proxy Mode Hi Joegen, Thanks for your reply. I have seen in the logs about this behavior, it gives *** NO STATIC ROUTE DEFINED *** From: sip:alice@10.0.16.54 Target: sip:charles@10.0.16.54 so I have added a relay route in the configuration as [sip:*@10.0.16.54] sip:10.0.16.54:5060. But still it's the same behavior. I cannot send the log in attachment so I send it by link : http://www.fileden.com/files/2006/9/8/208261/proxy-2008-03-15-17473.log Regards, P. Mohan Arun Kumar -----Original Message----- From: ope...@li... [mailto:ope...@li...] On Behalf Of Joegen E. Baclor Sent: Friday, March 14, 2008 6:40 PM To: ope...@li... Subject: Re: [OpenSBC] OpenSBC in Proxy Mode Please send in the level 5 log so we can have more insight on what's happening. Is B2BOnly mode resultsing to 404 as well? Joegen Mohan Arun Kumar wrote: > > Hi, > > I am unable to establish calls between two users when opensbc runs in > proxy mode. The users are registered successfully but when I call > other user it gives me a response 404. Can anyone help me in this issue? > > This is the scenario. > > Alice openSBC in Proxy mode Bob > | | | > | | | > | | | > | | | > | | | > | | | > | | | > | | | > | | | > | | | > | | | > | | | > | | | > > So now alice should be able to call bob. I have configured everything correctly as shown below. > > [OpenSBC General Parameters] > SIP Log Level=3 > PTRACE Log Level=1 > Log File Prefix=proxy > SBC Mode=Proxy Only Mode > Interface Address Array Size=1 > Interface Address 1=sip:10.0.10.10:5060 > Always Proxy Media=True > Enable Local Refer=False > Encryption Mode=XOR > Encryption Key=GS > Transaction Thread Count=10 > Session Thread Count=10 > Alerting Timeout=30000 > Seize Timeout=60000 > > [Trusted Domains] > Accept All Calls=True > Domain Array Size=0 > > [Local Domain Accounts] > Accept All Registration=True > Account Array Size=0 > > > But when alice calls or vice-versa it gives 404. Can anyone solve this issue??? > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------ - > 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. > Version: 7.5.518 / Virus Database: 269.21.7/1327 - Release Date: 3/12/2008 1:27 PM > ------------------------------------------------------------------------ - 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: Mohan A. K. <mk...@ve...> - 2008-03-14 13:42:34
|
Hi Joegen, Thanks for your reply. I have seen in the logs about this behavior, it gives *** NO STATIC ROUTE DEFINED *** From: sip:alice@10.0.16.54 Target: sip:charles@10.0.16.54 so I have added a relay route in the configuration as [sip:*@10.0.16.54] sip:10.0.16.54:5060. But still it's the same behavior. I cannot send the log in attachment so I send it by link : http://www.fileden.com/files/2006/9/8/208261/proxy-2008-03-15-17473.log Regards, P. Mohan Arun Kumar -----Original Message----- From: ope...@li... [mailto:ope...@li...] On Behalf Of Joegen E. Baclor Sent: Friday, March 14, 2008 6:40 PM To: ope...@li... Subject: Re: [OpenSBC] OpenSBC in Proxy Mode Please send in the level 5 log so we can have more insight on what's happening. Is B2BOnly mode resultsing to 404 as well? Joegen Mohan Arun Kumar wrote: > > Hi, > > I am unable to establish calls between two users when opensbc runs in > proxy mode. The users are registered successfully but when I call > other user it gives me a response 404. Can anyone help me in this issue? > > This is the scenario. > > Alice openSBC in Proxy mode Bob > | | | > | | | > | | | > | | | > | | | > | | | > | | | > | | | > | | | > | | | > | | | > | | | > | | | > > So now alice should be able to call bob. I have configured everything correctly as shown below. > > [OpenSBC General Parameters] > SIP Log Level=3 > PTRACE Log Level=1 > Log File Prefix=proxy > SBC Mode=Proxy Only Mode > Interface Address Array Size=1 > Interface Address 1=sip:10.0.10.10:5060 > Always Proxy Media=True > Enable Local Refer=False > Encryption Mode=XOR > Encryption Key=GS > Transaction Thread Count=10 > Session Thread Count=10 > Alerting Timeout=30000 > Seize Timeout=60000 > > [Trusted Domains] > Accept All Calls=True > Domain Array Size=0 > > [Local Domain Accounts] > Accept All Registration=True > Account Array Size=0 > > > But when alice calls or vice-versa it gives 404. Can anyone solve this issue??? > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------ - > 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. > Version: 7.5.518 / Virus Database: 269.21.7/1327 - Release Date: 3/12/2008 1:27 PM > ------------------------------------------------------------------------ - 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...> - 2008-03-14 13:09:58
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Please send in the level 5 log so we can have more insight on what's happening. Is B2BOnly mode resultsing to 404 as well? Joegen Mohan Arun Kumar wrote: > > Hi, > > I am unable to establish calls between two users when opensbc runs in > proxy mode. The users are registered successfully but when I call > other user it gives me a response 404. Can anyone help me in this issue? > > This is the scenario. > > Alice openSBC in Proxy mode Bob > | | | > | | | > | | | > | | | > | | | > | | | > | | | > | | | > | | | > | | | > | | | > | | | > | | | > > So now alice should be able to call bob. I have configured everything correctly as shown below. > > [OpenSBC General Parameters] > SIP Log Level=3 > PTRACE Log Level=1 > Log File Prefix=proxy > SBC Mode=Proxy Only Mode > Interface Address Array Size=1 > Interface Address 1=sip:10.0.10.10:5060 > Always Proxy Media=True > Enable Local Refer=False > Encryption Mode=XOR > Encryption Key=GS > Transaction Thread Count=10 > Session Thread Count=10 > Alerting Timeout=30000 > Seize Timeout=60000 > > [Trusted Domains] > Accept All Calls=True > Domain Array Size=0 > > [Local Domain Accounts] > Accept All Registration=True > Account Array Size=0 > > > But when alice calls or vice-versa it gives 404. Can anyone solve this issue??? > ------------------------------------------------------------------------ > > ------------------------------------------------------------------------- > 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. > Version: 7.5.518 / Virus Database: 269.21.7/1327 - Release Date: 3/12/2008 1:27 PM > |
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From: Mohan A. K. <mk...@ve...> - 2008-03-14 10:06:40
|
Hi,
I am unable to establish calls between two users when opensbc runs in
proxy mode. The users are registered successfully but when I call other
user it gives me a response 404. Can anyone help me in this issue?
This is the scenario.
Alice openSBC in Proxy mode Bob
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So now alice should be able to call bob. I have configured everything
correctly as shown below.
[OpenSBC General Parameters]
SIP Log Level=3
PTRACE Log Level=1
Log File Prefix=proxy
SBC Mode=Proxy Only Mode
Interface Address Array Size=1
Interface Address 1=sip:10.0.10.10:5060
Always Proxy Media=True
Enable Local Refer=False
Encryption Mode=XOR
Encryption Key=GS
Transaction Thread Count=10
Session Thread Count=10
Alerting Timeout=30000
Seize Timeout=60000
[Trusted Domains]
Accept All Calls=True
Domain Array Size=0
[Local Domain Accounts]
Accept All Registration=True
Account Array Size=0
But when alice calls or vice-versa it gives 404. Can anyone solve this
issue???
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From: Joegen E. B. <joe...@gm...> - 2008-03-14 02:35:57
|
Request URI is first checked. If there is no match, then the To-URI is checked. Daniel wrote: > Hi people... > > > > > > I would like know how does work the B2BUA route in OSBC... For example: > > > If I have in my rules _*+[sip:*my.home.com*] sip:192.168.0.15:8060+*_, the OSBC search for "my.home.com" in all SIP Message or only in RURI??? > > > > > > 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: Joegen E. B. <joe...@gm...> - 2008-03-14 02:34:55
|
you will need to setup opensbc in B2BUpperReg Mode. You may then set it as the outbound proxy for you user agents and point your domain to the address of your trixbox. Registration call-flow. UA -> OpenSBC -> Trixbox (Your domain should point to the address of the trixbox and not SBC) All calls should also pass through opensbc as both ingress and egress proxy. OpenSBC will always attempt to proxy the media for NATted calls or audio will not traverese the NAT. Eric wrote: > My goal: to enable remote users access to my trixbox from behind their LANs... (PnP capabilty) > > > > > > Currently we have a trixbox on a public IP that is working great, but requires port forwarding on the remote user's routers to work. > > > We now have OpenSBC running on a box within the same LAN that also has it's own public IP. > > > >From what I understand we'll need to point remote users to the IP of the SBC to bypass the NAT issue... > > > Now I haven't the slightest clue how to configure the SBC... I would like the SBC to only manage the SIP traffic and not proxy the RTP if possible. > > > I really appreciate any input... Thanks in advance. > > > -Eric > > > > > > > ------------------------------------------------------------------------- > 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: Daniel <dca...@ya...> - 2008-03-13 21:57:04
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Hi people... I would like know how does work the B2BUA route in OSBC... For example: If I have in my rules _*+[sip:*my.home.com*] sip:192.168.0.15:8060+*_, the OSBC search for "my.home.com" in all SIP Message or only in RURI??? Thanks! |
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From: Eric <ope...@op...> - 2008-03-13 15:09:21
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Any input on which "mode" to use so I can get started? Thanks |
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From: Eric <ope...@op...> - 2008-03-13 04:35:31
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My goal: to enable remote users access to my trixbox from behind their LANs... (PnP capabilty) Currently we have a trixbox on a public IP that is working great, but requires port forwarding on the remote user's routers to work. We now have OpenSBC running on a box within the same LAN that also has it's own public IP. >From what I understand we'll need to point remote users to the IP of the SBC to bypass the NAT issue... Now I haven't the slightest clue how to configure the SBC... I would like the SBC to only manage the SIP traffic and not proxy the RTP if possible. I really appreciate any input... Thanks in advance. -Eric |
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From: Joegen E. B. <joe...@gm...> - 2008-03-12 09:52:54
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If OpenSBC is residing on 222.222.222.222 and has dual interface with routes to 192.168.1.10, then all you need to do is to run opensbc in B2BUA Mode and add the necessary routes in B2BUA routes to point dialing rules for your outbound and inbound traffic. OpenSBC will be a natural bridge between you and your provider. Example: [sip:1212*] sip:111.111.111.111 <-------- route to your provider [sip:?????] sip:192.168.1.10 <-------- routes all 5 digit dial stings to your server rmcfadzien wrote: > Hi All > > > I am wanting to use opensbc to allow our SIP server on our internal LAN to work with a SIP provider in a trunk fashion. Our internal SIP server is 192.168.1.10, our SIP provider is 111.111.111.111 and our outside address on our firewall is 222.222.222.222 with everything forwarded to 192.168.1.10. > > > Our internal number range is 5 digits eg. 2xxxxx, 3xxxxx, 4xxxxx, 5xxxxx, 6xxxxx, 7xxxxx > > > Our SIP provider requires no authentication, they are using IP restrictions so we can send any number to them and they will accept without a username and password. I want the opensbc to ensure that our SIP headers reach the provider with the address 222.222.222.222 (instead of 192.168.1.10) and I want the provider to be able to dial in to any of our number range and be forwarded to 192.168.1.10 with the header intact. > > > Would anyone be able to give me the config to make this work ? > > > Look forward to any suggestions, sorry I am admittedly clueless in this area but am willing to learn and try anything, > > > rob. > > > > ------------------------------------------------------------------------- > 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: rmcfadzien <ope...@op...> - 2008-03-12 07:01:46
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Hi All I am wanting to use opensbc to allow our SIP server on our internal LAN to work with a SIP provider in a trunk fashion. Our internal SIP server is 192.168.1.10, our SIP provider is 111.111.111.111 and our outside address on our firewall is 222.222.222.222 with everything forwarded to 192.168.1.10. Our internal number range is 5 digits eg. 2xxxxx, 3xxxxx, 4xxxxx, 5xxxxx, 6xxxxx, 7xxxxx Our SIP provider requires no authentication, they are using IP restrictions so we can send any number to them and they will accept without a username and password. I want the opensbc to ensure that our SIP headers reach the provider with the address 222.222.222.222 (instead of 192.168.1.10) and I want the provider to be able to dial in to any of our number range and be forwarded to 192.168.1.10 with the header intact. Would anyone be able to give me the config to make this work ? Look forward to any suggestions, sorry I am admittedly clueless in this area but am willing to learn and try anything, rob. |
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From: Joegen E. B. <joe...@gm...> - 2008-03-04 03:21:27
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Before anything else, please paste the entire compilation log here. Tell us what version of MSVC and Windows you are using and where you put the G.729 libraries. Needless to say, G.729 compiles fine here. Joegen Ilian Jeri C. Pinzon wrote: > Hi, > > That's strange. What project are you using? ATLSIP? Can you show me what > codecs appear in your codec list? > > Thanks. > > Adalicio wrote: > >> Hi Ilian, >> >> >> The filenames were changed and the rebuild was made but still not in the list. >> >> >> I don´t know if this g729 codec inclusion procedure have some additional steps. I mean, the not obvious ones. >> >> >> >> >> >> >> Located VOICEAGE at C:\Documents and Settings\junior\MeusProjetos\opensipstack\external\codecs\ >> >> >> (This line intrigates me) >> >> >> >> >> >> >> Thks for the Help >> >> >> >> ------------------------------------------------------------------------- >> 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: Ilian J. C. P. <ip...@so...> - 2008-03-04 03:09:10
|
Hi, That's strange. What project are you using? ATLSIP? Can you show me what codecs appear in your codec list? Thanks. Adalicio wrote: > Hi Ilian, > > > The filenames were changed and the rebuild was made but still not in the list. > > > I don´t know if this g729 codec inclusion procedure have some additional steps. I mean, the not obvious ones. > > > > > > > Located VOICEAGE at C:\Documents and Settings\junior\MeusProjetos\opensipstack\external\codecs\ > > > (This line intrigates me) > > > > > > > Thks for the Help > > > > ------------------------------------------------------------------------- > 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: Adalicio <ope...@op...> - 2008-03-03 21:12:38
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Hi Ilian, The filenames were changed and the rebuild was made but still not in the list. I don´t know if this g729 codec inclusion procedure have some additional steps. I mean, the not obvious ones. Located VOICEAGE at C:\Documents and Settings\junior\MeusProjetos\opensipstack\external\codecs\ (This line intrigates me) Thks for the Help |
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From: Marthin v. D. <mar...@gm...> - 2008-03-03 16:45:23
|
Hi, I am unable to find any information (docs, forum search, google etc.) that openSBC supports the generation of CDR's (call detail records) or any kind of traffic reporting functionality. Is this functionality available? Please point me in the right direction to set reporting and CDR generation for OpenSBC. Many thanks for your time. Marthin |
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From: Evelio <ope...@op...> - 2008-03-03 04:34:47
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I am interested in a variant of that... How about this??
Wholesale customer carrier --------------------> OSBC (Public IP) <-------------------------> Openser (Private IP)
I
I I
Gateway(Private IP)
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From: Joegen E. B. <joe...@gm...> - 2008-02-27 12:12:50
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Hi, I have not tried it but I am sure you will have problems with this setup. Number one road block you will hit would be SER will be inserting record-routes that are private which surely would screw-up routing. Joegen Daniel wrote: > Hi... > > > I was studing the OpenSBC and I would like know: is possible make this configuration? > > > > > > > > > endpoint --------------------> OSBC (Public IP) <-------------------------> Openser (Private IP) > > > | > > > | > > > | > > > V > > > Destination (Carrier, endpoint, etc...) > > > > > > > > > The endpoint can be a ATA or a Softphone... They will register in Openser through OSBC... > > > > All signaling must pass by OSBC, and the openser will do the analysis of routes and pass for the OSBC... > > > > > > I hope I have been clear... > > > > > > 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: Daniel <dca...@ya...> - 2008-02-25 22:00:47
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Hi...
I was studing the OpenSBC and I would like know: is possible make this configuration?
endpoint --------------------> OSBC (Public IP) <-------------------------> Openser (Private IP)
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V
Destination (Carrier, endpoint, etc...)
The endpoint can be a ATA or a Softphone... They will register in Openser through OSBC...
All signaling must pass by OSBC, and the openser will do the analysis of routes and pass for the OSBC...
I hope I have been clear...
Thanks...
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From: Joegen E. B. <joe...@gm...> - 2008-02-20 13:37:05
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We have tested MWI to work with sipX 3.8 where sipX is also the registrar. I am not familiar with asterisk and how it handles MWI but there shouldn't be any reason why it wouldn't work. Is asterisk sending unsolicited NOTIFYs or does it need a corresponding SUBSCRIBE? As always, if you think OpenSBC is misbehaving, please send in corresponding level 5 logs and a description of the problem so we could patch it. Joegen Erik Boyer wrote: > Does OpenSBC support MWI? And if it doesn't, are there plans to in the future or ways to get around that now? I'd like to have MWI messages coming from Asterisk make it through the SBC box. > > ------------------------------------------------------------------------- > 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...> - 2008-02-20 12:42:57
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Does OpenSBC support MWI? And if it doesn't, are there plans to in the future or ways to get around that now? I'd like to have MWI messages coming from Asterisk make it through the SBC box. |
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From: Joegen E. B. <joe...@gm...> - 2008-02-12 02:21:04
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Erik Boyer wrote:
>> {quote:title=Guest wrote:}{quote}
>> What do you exacly mean about "BC fails to acknowledge them" ? OpenSBC
>> does not send ACK? or do you claim OpenSBC ignores reINVITE?
>>
>
> According to the SIP traces I've done, the Metaswitch sends out the reINVITES to make sure the end connection is still alive, and after a certain number of missed ACKs from the other end it will end the call because it does not think the other device is there anymore.
If metaswitch send the reINVITE, then it should be the one sending the
ACK shouldn't it?
> In the trace I can see the INVITE from the Meta, and instead of a 200 OK coming back from the SBC I see two '100 Trying' messages. I have a copy of a good call and bad call SIP trace if you would like to see them.
>
OpenSBC will relay this reINVITE to the other UA. If you UA did not
respond with the reINVITE, then it's the UA which is at fault. If you
think OpenSBC is not relaying the INVITE, then that is another thing.
Can you confirm this further?
>
>> {quote:title=Guest wrote:}{quote}
>> This is certainly an interesting issue and I like to learn more about
>> the real cause of your problem. But I need your help to catch the
>> actual logs and more details what we should be looking for.
>>
>
> What do you need? The configuration of my OpenSBC, Level 5 logs, etc? I'm using the latest pull from CVS from February 4, 2008 for both OpenSipStack and OpenSBC.
>
> Thanks,
>
> 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...> - 2008-02-12 01:46:20
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> {quote:title=Guest wrote:}{quote}
> What do you exacly mean about "BC fails to acknowledge them" ? OpenSBC
> does not send ACK? or do you claim OpenSBC ignores reINVITE?
According to the SIP traces I've done, the Metaswitch sends out the reINVITES to make sure the end connection is still alive, and after a certain number of missed ACKs from the other end it will end the call because it does not think the other device is there anymore. In the trace I can see the INVITE from the Meta, and instead of a 200 OK coming back from the SBC I see two '100 Trying' messages. I have a copy of a good call and bad call SIP trace if you would like to see them.
> {quote:title=Guest wrote:}{quote}
> This is certainly an interesting issue and I like to learn more about
> the real cause of your problem. But I need your help to catch the
> actual logs and more details what we should be looking for.
What do you need? The configuration of my OpenSBC, Level 5 logs, etc? I'm using the latest pull from CVS from February 4, 2008 for both OpenSipStack and OpenSBC.
Thanks,
Erik
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From: Joegen E. B. <joe...@gm...> - 2008-02-12 01:04:02
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Erik Boyer wrote: > Hello again, > > > After realizing what the problem was with my SBC dropping the audio after 20 seconds, I've come upon a new interesting issue. Once again, calling into the UA is fine and the call lasts forever, but when calling out from the UA, sometimes the call ends after 1 minute 5 seconds. The reason for this is known: the Metaswitch is sending out reINVITEs to the SBC but the SBC fails to acknowledge them. What do you exacly mean about "BC fails to acknowledge them" ? OpenSBC does not send ACK? or do you claim OpenSBC ignores reINVITE? > The strange thing is that this does not happen all the time. So far, the only discernable pattern I've been able to see is between calls that would be long distance (an added 1 to the 10 digit number) versus local calls. It would appear that any time the UA calls out to a long distance number the SBC does not reply to the Metaswitch's reINVITES. > This is certainly an interesting issue and I like to learn more about the real cause of your problem. But I need your help to catch the actual logs and more details what we should be looking for. > > Any thoughts on why this would be? > > > 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: Joegen E. B. <joe...@gm...> - 2008-02-12 00:58:56
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No. OpenSBC is not a STUN server. Reg wrote: > Is OpenSBC also a STUN Server? > > > Thank you for response. > > > > ------------------------------------------------------------------------- > 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...> - 2008-02-11 21:00:08
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Hello again,
After realizing what the problem was with my SBC dropping the audio after 20 seconds, I've come upon a new interesting issue. Once again, calling into the UA is fine and the call lasts forever, but when calling out from the UA, sometimes the call ends after 1 minute 5 seconds. The reason for this is known: the Metaswitch is sending out reINVITEs to the SBC but the SBC fails to acknowledge them. The strange thing is that this does not happen all the time. So far, the only discernable pattern I've been able to see is between calls that would be long distance (an added 1 to the 10 digit number) versus local calls. It would appear that any time the UA calls out to a long distance number the SBC does not reply to the Metaswitch's reINVITES.
Any thoughts on why this would be?
Thanks in advance,
Erik
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