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From: Whit T. <de...@wh...> - 2009-02-01 15:39:15
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Once you create a routing rule in OpenSBC, it will forward the register the request to Asterisk so the SIP client will be registered there. In this case the registration is handled by asterisk, so no client accounts need to be setup in OpenSBC. In this setup, OpenSBC is acting as a router, forwarding all packets to asterisk. Then you can utilize whatever trunks to the PSTN you have in Asterisk. So essentially its like having asterisk exposed on its own external IP address, but in this case OpenSBC is hiding the internal workings of your network. Hope that makes it more clear. The easiest thing to do is configure a route everything to asterisk rule and try it out.. you can create more complex routes once you get the hang of it.. Whit OpenSBC Forum wrote: > Thank you for your quick response. I just want to clear something do I have to create routes in the B2BUA-routes? With this setup my SIP clients will register to the IP address of my SBC box? and automatically SBC will forward all sip registration request to asterisk even if I don't create any local domain users. How about if I have IAX trunks in asterisk will my sip clients be able to use that trunk if they have registered in my SBC Box? Sorry for the many questions I am really really confused with this setup. Thank you again. > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > Opensipstack-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > > |
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From: OpenSBC F. <ope...@op...> - 2009-02-01 02:07:22
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Thank you for your quick response. I just want to clear something do I have to create routes in the B2BUA-routes? With this setup my SIP clients will register to the IP address of my SBC box? and automatically SBC will forward all sip registration request to asterisk even if I don't create any local domain users. How about if I have IAX trunks in asterisk will my sip clients be able to use that trunk if they have registered in my SBC Box? Sorry for the many questions I am really really confused with this setup. Thank you again. |
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From: Whit T. <de...@wh...> - 2009-01-31 15:38:01
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That configuration is actually quite simple. Set OpenSBC to B2BUAUpperReg mode along with the Always Proxy Media selection and set up routes to point any connection from OpenSBC to Asterisk. You can create as complex or simple routing rules as you want (look at the examples in the docs on your routes). The most simple is to add a wildcard route to point everything received to the upstream node(Asterisk). Or you can create simple domains in the config to then route clients to whichever asterisk instance you want (if there are multiple Asterisk boxes). Remember, if you create a domain structure in OpenSBC, you will have to include that information in the SIP client so OpenSBC knows where to send the traffic. By routing everything from the Border Controller to Asterisk the clients will actually register with Asterisk and you can then access the PSTN from there. I'd start with wild card entries and route everything at first till you get the hang of it. Regards, Whit OpenSBC Forum wrote: > Hi I am new to using OpenSBC. I want to implement this kind of setup. I want to ask if it is possible. If it is possible will anyone with a kind heart give me a headstart on how to set it up. Here is my kind of setup > > > SIP Clients-------Internet ------- OpenSBC ------- Asterisk---- PSTN > > > My Sip extensions are configured in my asterisk box. I want to hide Asterisk from the outside world so I want my SIP Clients to register in OpenSBC with the accounts configured in Asterisk. Can OpenSBC do this? Do I have to add local domain users in OpenSBC? or Can OpenSBC do the registration for Asterisk. > > > I hope someone would be able to help me with this I have search google for any information about this setup but to no avail. > > > Thank you very much in advance.. > > > > ------------------------------------------------------------------------------ > This SF.net email is sponsored by: > SourcForge Community > SourceForge wants to tell your story. > http://p.sf.net/sfu/sf-spreadtheword > _______________________________________________ > Opensipstack-osbcdevel mailing list > Ope...@li... > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > > |
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From: OpenSBC F. <ope...@op...> - 2009-01-31 15:11:03
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Hi I am new to using OpenSBC. I want to implement this kind of setup. I want to ask if it is possible. If it is possible will anyone with a kind heart give me a headstart on how to set it up. Here is my kind of setup SIP Clients-------Internet ------- OpenSBC ------- Asterisk---- PSTN My Sip extensions are configured in my asterisk box. I want to hide Asterisk from the outside world so I want my SIP Clients to register in OpenSBC with the accounts configured in Asterisk. Can OpenSBC do this? Do I have to add local domain users in OpenSBC? or Can OpenSBC do the registration for Asterisk. I hope someone would be able to help me with this I have search google for any information about this setup but to no avail. Thank you very much in advance.. |
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From: OpenSBC F. <ope...@op...> - 2009-01-30 12:32:22
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> {quote:title=Guest wrote:}{quote}
> Hi Joegen
>
> Do contributions to the project end up in the Open Source source or Solegy
> code? I have read past forum traffic about contributions but had not found
> any index or table of contents about them.
>
> r
Of course, all "accepted" contributions would go back as part of OpenSIPStack/OpenSBC. Acutally even enhancement work contracted via Solegy also goes back to opensource. Honestly there isn't any contributions yet I have approved so far from the community. There is one undergoing evaluation which concerns presence support in the softphone classes. For patches and even bug reports, i see to it that I at least mention the name of the person in CVS comments.
Joegen
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From: voice <vo...@ne...> - 2009-01-28 14:58:45
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Hi Joegen Do contributions to the project end up in the Open Source source or Solegy code? I have read past forum traffic about contributions but had not found any index or table of contents about them. r > > ----- Original Message ----- > From: "Joegen Baclor" <jb...@so...> > To: <ope...@op...>; > <ope...@li...> > Sent: Tuesday, January 27, 2009 11:00 PM > Subject: Re: [OpenSBC] Support for UDPTL (as used for T.38)? > > > > Your contribution to this project would be much welcome. For starters, > you > > should be looking at three important objects in OpenSIPStack if you are to > > implement T.38 proxy in OpenSIPStack. They are RTP_SessionManager, > RTP_UDP > > and B2BMediaInterface . Each B2BUACall has an RTP_SessionManager of it's > > own which maintains the RTP_UDP session responsible for each leg while > > B2BMediaInterface is the central location where parsing of the SDP Offer > and > > Answer is done. You can simply base your T38_SessionManager and T38_UDP > > classes from their RTP counterparts and modify B2BMediaInterface to parse > > T.38 m line as well. > > > > HTH > > > > Joegen > > > > > > > > Too bad. We might try to add it. Is there anything we should know? > > > > > > > > > > > -------------------------------------------------------------------------- > ---- > > This SF.net email is sponsored by: > > SourcForge Community > > SourceForge wants to tell your story. > > http://p.sf.net/sfu/sf-spreadtheword > > _______________________________________________ > > Opensipstack-osbcdevel mailing list > > Ope...@li... > > https://lists.sourceforge.net/lists/listinfo/opensipstack-osbcdevel > > > > |
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From: Joegen B. <jb...@so...> - 2009-01-28 05:00:37
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Your contribution to this project would be much welcome. For starters, you should be looking at three important objects in OpenSIPStack if you are to implement T.38 proxy in OpenSIPStack. They are RTP_SessionManager, RTP_UDP and B2BMediaInterface . Each B2BUACall has an RTP_SessionManager of it's own which maintains the RTP_UDP session responsible for each leg while B2BMediaInterface is the central location where parsing of the SDP Offer and Answer is done. You can simply base your T38_SessionManager and T38_UDP classes from their RTP counterparts and modify B2BMediaInterface to parse T.38 m line as well. HTH Joegen > > Too bad. We might try to add it. Is there anything we should know? > |
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From: OpenSBC F. <ope...@op...> - 2009-01-28 04:00:02
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Too bad. We might try to add it. Is there anything we should know? --Eric |
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From: Joegen B. <jb...@so...> - 2009-01-28 03:00:47
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Sorry, no OpenSBC does not support proxying of T.38. If you send T.38 via OpenSBC, it will not modify the m line for T.38 media. This would result to audio/video only proxy while T.38 remains end-to-end. Joegen > > I couldn't find any mention of it, but does OpenSBC currently support > proxying UDPTL media? > |
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From: OpenSBC F. <ope...@op...> - 2009-01-28 00:11:02
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I couldn't find any mention of it, but does OpenSBC currently support proxying UDPTL media? |
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From: Joegen B. <jb...@so...> - 2009-01-22 01:24:29
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Hi, This is very true and could be really frustrating if you need presense. OpenSBC currently do not process SUBSCRIBE messages and would respond with a bad extension by default. I am planning to work on this for version 1.1.6. The real problem is not really the incapability of OSBC to relay SUBSCRIBEs to upper-reg but the requirement to rewrite SUBSCRIBE messages the same way as its REGISTER counterpart. It's a bit tricky. Joegen -------------------------------------------------- From: "OpenSBC Forum" <ope...@op...> Sent: Thursday, January 22, 2009 8:02 AM To: <ope...@li...> Subject: [OpenSBC] OpenSBC Gives 420 Bad Extension on SIP Subscribe > I have OpenSBC 1.15 RC3 running in B2BUA Upper Reg Mode, sitting in front > of a sipXecs deployment. User registrations and calls are being > successfully routed through OpenSBC -- but I'm having a problem with > SUBSCRIBE events. I'm using X-Lite as my UA, and when X-Lite sends a > SUBSCRIBE to OpenSBC (to gather contact presence information), OpenSBC > sends a "420 Bad Extension" response. I've also seen the same behavior > when I've used Sip Communicator. > > Are there any settings in OpenSBC that could affect its ability to respond > to Sip SUBSCRIBE requests? I've included log entries below to demonstrate > the issue: > > 2009/01/21 09:00:55.997 INF: CID=0x0e8d <<< SUBSCRIBE sip:us...@si...main > SIP/2.0 SRC: 10.0.0.1:46288:UDP enc=0 bytes=673 > 2009/01/21 09:00:55.997 DBG: CID=0x0e8d > 2009/01/21 09:00:55.997 DBG: CID=0x0e8d SUBSCRIBE > sip:us...@si...main SIP/2.0 > 2009/01/21 09:00:55.997 DBG: CID=0x0e8d From: "usera" > <sip:us...@si...main>;tag=f449fc5d > 2009/01/21 09:00:55.997 DBG: CID=0x0e8d To: "User B" > <sip:us...@si...main> > 2009/01/21 09:00:55.997 DBG: CID=0x0e8d Via: SIP/2.0/UDP > 10.0.0.1:46288;branch=z9hG4bK-d8754z-5f50db05221c3b30-1---d8754z- > 2009/01/21 09:00:55.997 DBG: CID=0x0e8d CSeq: 1 SUBSCRIBE > 2009/01/21 09:00:55.997 DBG: CID=0x0e8d Call-ID: > YTRkMjc1NGUyNjg5MTBlNDk2ZDQ4OTUwYzU4NmQxZTU. > 2009/01/21 09:00:55.997 DBG: CID=0x0e8d Contact: > <sip:usera@10.0.0.1:46288> > 2009/01/21 09:00:55.997 DBG: CID=0x0e8d Event: presence > 2009/01/21 09:00:55.997 DBG: CID=0x0e8d Subject: > 2009/01/21 09:00:55.997 DBG: CID=0x0e8d User-Agent: X-Lite release > 1100l stamp 47546 > 2009/01/21 09:00:55.997 DBG: CID=0x0e8d Expires: 300 > 2009/01/21 09:00:55.997 DBG: CID=0x0e8d Max-Forwards: 70 > 2009/01/21 09:00:55.997 DBG: CID=0x0e8d Accept: multipart/related, > application/rlmi+xml, application/pidf+xml > 2009/01/21 09:00:55.997 DBG: CID=0x0e8d Allow: INVITE, ACK, CANCEL, > OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO > 2009/01/21 09:00:55.997 DBG: CID=0x0e8d Content-Length: 0 > 2009/01/21 09:00:55.997 DBG: CID=0x0e8d > 2009/01/21 09:00:55.997 DBG: CID=0x0e8d > 2009/01/21 09:00:55.998 DTL: CID=0x0e8d NIST(4156922636) *** CREATED > *** - > NIST|YTRkMjc1NGUyNjg5MTBlNDk2ZDQ4OTUwYzU4NmQxZTU.|z9hG4bK-d8754z-5f50db > 05221c3b30-1---d8754z-|SUBSCRIBE > 2009/01/21 09:00:55.998 DTL: CID=0x0e8d NIST(4156922636) > Event(SIPMessage) - SUBSCRIBE sip:us...@si...main SIP/2.0 > 2009/01/21 09:00:55.998 DBG: CID=0x0e8d TRANSACTION: (NIST) SUBSCRIBE > sip:us...@si...main SIP/2.0 State: 0 > 2009/01/21 09:00:55.998 DTL: CID=0x0e8d NIST(4156922636) > StateIdle->StateTrying > 2009/01/21 09:00:55.998 DTL: CID=0x0e8d NIST(4156922636) > Event(SIPMessage) - SIP/2.0 420 Bad Extension > 2009/01/21 09:00:55.998 DBG: CID=0x0e8d TRANSACTION: (NIST) SIP/2.0 420 > Bad Extension State: 1 > 2009/01/21 09:00:55.999 DTL: CID=0x0e8d > NIST(4156922636)HandleStateTrying->StateCompleted > 2009/01/21 09:00:55.999 INF: CID=0x0e8d >>> SIP/2.0 420 Bad Extension DST: > 10.0.0.1:46288:UDP SRC=10.0.0.20:5061 enc=0 bytes=436 > 2009/01/21 09:00:55.999 DBG: CID=0x0e8d > 2009/01/21 09:00:55.999 DBG: CID=0x0e8d SIP/2.0 420 Bad Extension > 2009/01/21 09:00:55.999 DBG: CID=0x0e8d From: "usera" > <sip:us...@si...main>;tag=f449fc5d > 2009/01/21 09:00:55.999 DBG: CID=0x0e8d To: "User B" > <sip:us...@si...main>;tag=ce21c5bc4ae6dd1196cdcd16fb6551b7 > 2009/01/21 09:00:55.999 DBG: CID=0x0e8d Via: SIP/2.0/UDP > 10.0.0.1:46288;branch=z9hG4bK-d8754z-5f50db05221c3b30-1---d8754z-;rport=46288;recei > ved=10.0.0.1 > 2009/01/21 09:00:55.999 DBG: CID=0x0e8d CSeq: 1 SUBSCRIBE > 2009/01/21 09:00:55.999 DBG: CID=0x0e8d Call-ID: > YTRkMjc1NGUyNjg5MTBlNDk2ZDQ4OTUwYzU4NmQxZTU. > 2009/01/21 09:00:55.999 DBG: CID=0x0e8d Server: OpenSBC v1.1.5-42 > 2009/01/21 09:00:55.999 DBG: CID=0x0e8d Content-Length: 0 > 2009/01/21 09:00:55.999 DBG: CID=0x0e8d > 2009/01/21 09:00:55.999 DBG: CID=0x0e8d > |
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From: OpenSBC F. <ope...@op...> - 2009-01-22 00:03:17
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I have OpenSBC 1.15 RC3 running in B2BUA Upper Reg Mode, sitting in front of a sipXecs deployment. User registrations and calls are being successfully routed through OpenSBC -- but I'm having a problem with SUBSCRIBE events. I'm using X-Lite as my UA, and when X-Lite sends a SUBSCRIBE to OpenSBC (to gather contact presence information), OpenSBC sends a "420 Bad Extension" response. I've also seen the same behavior when I've used Sip Communicator. Are there any settings in OpenSBC that could affect its ability to respond to Sip SUBSCRIBE requests? I've included log entries below to demonstrate the issue: 2009/01/21 09:00:55.997 INF: CID=0x0e8d <<< SUBSCRIBE sip:us...@si...main SIP/2.0 SRC: 10.0.0.1:46288:UDP enc=0 bytes=673 2009/01/21 09:00:55.997 DBG: CID=0x0e8d 2009/01/21 09:00:55.997 DBG: CID=0x0e8d SUBSCRIBE sip:us...@si...main SIP/2.0 2009/01/21 09:00:55.997 DBG: CID=0x0e8d From: "usera" <sip:us...@si...main>;tag=f449fc5d 2009/01/21 09:00:55.997 DBG: CID=0x0e8d To: "User B" <sip:us...@si...main> 2009/01/21 09:00:55.997 DBG: CID=0x0e8d Via: SIP/2.0/UDP 10.0.0.1:46288;branch=z9hG4bK-d8754z-5f50db05221c3b30-1---d8754z- 2009/01/21 09:00:55.997 DBG: CID=0x0e8d CSeq: 1 SUBSCRIBE 2009/01/21 09:00:55.997 DBG: CID=0x0e8d Call-ID: YTRkMjc1NGUyNjg5MTBlNDk2ZDQ4OTUwYzU4NmQxZTU. 2009/01/21 09:00:55.997 DBG: CID=0x0e8d Contact: <sip:usera@10.0.0.1:46288> 2009/01/21 09:00:55.997 DBG: CID=0x0e8d Event: presence 2009/01/21 09:00:55.997 DBG: CID=0x0e8d Subject: 2009/01/21 09:00:55.997 DBG: CID=0x0e8d User-Agent: X-Lite release 1100l stamp 47546 2009/01/21 09:00:55.997 DBG: CID=0x0e8d Expires: 300 2009/01/21 09:00:55.997 DBG: CID=0x0e8d Max-Forwards: 70 2009/01/21 09:00:55.997 DBG: CID=0x0e8d Accept: multipart/related, application/rlmi+xml, application/pidf+xml 2009/01/21 09:00:55.997 DBG: CID=0x0e8d Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO 2009/01/21 09:00:55.997 DBG: CID=0x0e8d Content-Length: 0 2009/01/21 09:00:55.997 DBG: CID=0x0e8d 2009/01/21 09:00:55.997 DBG: CID=0x0e8d 2009/01/21 09:00:55.998 DTL: CID=0x0e8d NIST(4156922636) *** CREATED *** - NIST|YTRkMjc1NGUyNjg5MTBlNDk2ZDQ4OTUwYzU4NmQxZTU.|z9hG4bK-d8754z-5f50db 05221c3b30-1---d8754z-|SUBSCRIBE 2009/01/21 09:00:55.998 DTL: CID=0x0e8d NIST(4156922636) Event(SIPMessage) - SUBSCRIBE sip:us...@si...main SIP/2.0 2009/01/21 09:00:55.998 DBG: CID=0x0e8d TRANSACTION: (NIST) SUBSCRIBE sip:us...@si...main SIP/2.0 State: 0 2009/01/21 09:00:55.998 DTL: CID=0x0e8d NIST(4156922636) StateIdle->StateTrying 2009/01/21 09:00:55.998 DTL: CID=0x0e8d NIST(4156922636) Event(SIPMessage) - SIP/2.0 420 Bad Extension 2009/01/21 09:00:55.998 DBG: CID=0x0e8d TRANSACTION: (NIST) SIP/2.0 420 Bad Extension State: 1 2009/01/21 09:00:55.999 DTL: CID=0x0e8d NIST(4156922636)HandleStateTrying->StateCompleted 2009/01/21 09:00:55.999 INF: CID=0x0e8d >>> SIP/2.0 420 Bad Extension DST: 10.0.0.1:46288:UDP SRC=10.0.0.20:5061 enc=0 bytes=436 2009/01/21 09:00:55.999 DBG: CID=0x0e8d 2009/01/21 09:00:55.999 DBG: CID=0x0e8d SIP/2.0 420 Bad Extension 2009/01/21 09:00:55.999 DBG: CID=0x0e8d From: "usera" <sip:us...@si...main>;tag=f449fc5d 2009/01/21 09:00:55.999 DBG: CID=0x0e8d To: "User B" <sip:us...@si...main>;tag=ce21c5bc4ae6dd1196cdcd16fb6551b7 2009/01/21 09:00:55.999 DBG: CID=0x0e8d Via: SIP/2.0/UDP 10.0.0.1:46288;branch=z9hG4bK-d8754z-5f50db05221c3b30-1---d8754z-;rport=46288;recei ved=10.0.0.1 2009/01/21 09:00:55.999 DBG: CID=0x0e8d CSeq: 1 SUBSCRIBE 2009/01/21 09:00:55.999 DBG: CID=0x0e8d Call-ID: YTRkMjc1NGUyNjg5MTBlNDk2ZDQ4OTUwYzU4NmQxZTU. 2009/01/21 09:00:55.999 DBG: CID=0x0e8d Server: OpenSBC v1.1.5-42 2009/01/21 09:00:55.999 DBG: CID=0x0e8d Content-Length: 0 2009/01/21 09:00:55.999 DBG: CID=0x0e8d 2009/01/21 09:00:55.999 DBG: CID=0x0e8d |
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From: OpenSBC F. <ope...@op...> - 2009-01-21 05:17:58
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Hi Joegen, Thanks for prompt response. 1) I'll look at that. Any plans to add this feature to the main stream? 2) Get registration notification in this or other way is 'MUST' for our project. We can consume either 3rd party registration or subscribe to events (reg or presence). I am sure it will beneficial to many applications that requre this information to make routing decision and implement service logic correctly. 3) We will defintely look at that, currently we use asterisk based billing solution. Thanks, Boris |
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From: Joegen B. <jb...@so...> - 2009-01-21 04:56:07
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Hi Boris, Inline .... > > I am looking for SBC solution and this project looks very promising. > > I want to get into the track fast, could you please help me clarify the > following: > > 1) > Can I manage subscriber accounts parameters from external system? > I guess that subscriber information is stored in the database, so I > probably > can modify records in this DB? > Not without modifying the existing OpenSBC code. It should be easy enough for an average C++ developer to fetch user auth information from an external source other than the built-in OpenSBC config. > > 2) > What is the way to get notification regarding registration > status (registered/unregistered). Ideally it will be 3^rd^ party > registration or registration event package support. > Registration event mechanism is already supported in OpenSIPStack so it should be easy to add it to OpenSBC in future builds. Is you event subscriber going to be Open Source? I would be willing to put sometime on the event package if it would benefit the community. > > 3) Are any billing solution available for integration with OpenSBC? > Solegy has a billing platform already integrated with OpenSBC. Joegen |
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From: OpenSBC F. <ope...@op...> - 2009-01-21 04:23:30
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I am looking for SBC solution and this project looks very promising. I want to get into the track fast, could you please help me clarify the following: 1) Can I manage subscriber accounts parameters from external system? I guess that subscriber information is stored in the database, so I probably can modify records in this DB? 2) What is the way to get notification regarding registration status (registered/unregistered). Ideally it will be 3^rd^ party registration or registration event package support. 3) Are any billing solution available for integration with OpenSBC? Thanks, Boris |
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From: OpenSBC F. <ope...@op...> - 2009-01-20 20:53:48
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thanks, resent... |
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From: OpenSBC F. <ope...@op...> - 2009-01-20 20:18:13
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Mike, Sorry, but I am not able to find the logs you sent. Please resend to ehernaez-at-gmail.com Tthanks |
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From: OpenSBC F. <ope...@op...> - 2009-01-20 19:19:18
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Eric, Any thoughts on this? Did you receive my logs? Thanks, Mike |
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From: OpenSBC F. <ope...@op...> - 2009-01-19 14:26:34
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Ok I figured out that you really mean adding <root> and </root> literally. Now ( operating in full mode, B2B modes make no difference ) I see that the trunks are getting registered. My current situation is having both internal users and external trunks correctly registered. I tried calling one of the registered trunks, I see the INVITE being sent to OpenSBC, I expected the inbound-route getting a relayed INVITE from OpenSBC, but in fact nothing happens! Is there anything more that must be set? Thanks |
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From: OpenSBC F. <ope...@op...> - 2009-01-19 12:41:07
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Your example is blank, nothing can be read in the example window! |
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From: OpenSBC F. <ope...@op...> - 2009-01-19 10:26:15
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Hi Paolo,
There is a typo in that documentation. the sample XML did not enclose the block in a ROOT ELEMENT causing parser error in the opensbc XML module. This causes the register not to be sent. Encode it like this:
{code:xml}<root>
<siptrunk trunk-name="opteron.opensipstack.org" route-set="opteron.opensipstack.org" sip-domain="opteron.opensipstack.org" expires="10">
<trunk-accounts>
<account user-name="1001" auth-user-name="1001" auth-password="1001" inbound-route="sip:90...@wi..." send-reg="yes" expires="3600" />
</trunk-accounts>
</root>{code}
Joegen
> {quote:title=Paolo wrote:}{quote}
>
> Nope.
>
>
> I had already noticed the document about trunks, there aren't too many documents to be read after all...
>
>
> I tried both B2B and B2B Upper, but nothing. I see no trace of registrations in the log.
>
>
> My configuration is (sorry for the long post) as follows.
>
>
> Help! Thanks everybody!
>
>
> [Solegy]
> RTTS-Client-Address=83.211.125.71
>
> [OpenSBC-General-Parameters]
> SIP-Log-Level=5
> PTRACE-Log-Level=1
> Log-File-Prefix=b2bua
> SBC-Application-Mode=B2BUpperReg Mode
> Interface-Address Array Size=0
> Enable-Backdoor-Port=True
> Enable-Trunk-Port=True
> Enable-Calea-Port=True
> RTP-Min-Port=10000
> RTP-Max-Port=20000
> Enable-Local-Refer=False
> Max-Forwards=70
> Encryption-Mode=XOR
> Encryption-Key=GS
> Transaction-Thread-Count=10
> Session-Thread-Count=10
> Alerting-Timeout=30000
> Seize-Timeout=60000
> SIP-Timer-B=Default
> SIP-Timer-H=Default
> Session-Keep-Alive=1800
> Session-Max-Life-Span=10800
> Max-Concurrent-Session=100
> Max-Call-Rate-Per-Second=10
>
> [Upper-Registration]
> Enable-Stateful-Reg=True
> Rewrite-TO-Domain=True
> Rewrite-FROM-Domain=True
> Route-List Array Size=0
>
> [Internal-DNS-Mapping]
> Internal-DNS-Map Array Size=0
>
> [Proxy-Relay-Routes]
> Drop-Routes-On-Ping-Timeout=False
> Proxy-Resolve-To-URI=True
> Route-List Array Size=0
>
> [B2BUA-Routes]
> Enable-Route-Scripting=False
> Route-Script=b2bua-route.cscript
> Route-List Array Size=1
> Route-List 1=[sip:*@voip.eutelia.it:*] sip:voip.eutelia.it:5060;sip-trunk=true
> Insert-Route-Header=True
> Rewrite-TO-URI=True
> Prepend-ISUP-OLI=False
> Route-By-Request-URI=True
> Route-By-To-URI=False
> Drop-Routes-On-Ping-Timeout=False
> Use-External-XML=False
> External-XML-File=b2bua-route.xml
> Route-List 2=[sip:030...@vo...] sip:030...@si...
>
> [Media-Server]
> Enable-Media-Server=False
> Media-Server-Number=5000
> Codec-List Array Size=0
> No-RTP-Proxy-On-All-Transfers=False
> Enable-Announcement-Service=False
> 4xx-Error-Map=prompts/basic/cant_complete.wav
> 5xx-Error-Map=prompts/basic/cant_complete.wav
> 6xx-Error-Map=prompts/basic/cant_complete.wav
> Announcement-Error-Map Array Size=0
>
> [Outbound-Proxies]
> Outbound-Proxies Array Size=0
>
> [Local-Domain-Accounts]
> Accept-All-Registration=False
> Account-List Array Size=3
> Account-List 1=sip:0302054741:xx...@si...
> Account-List 2=sip:0302054742:xx...@si...
> Account-List 3=sip:0302054681:xx...@si...
>
> [RTP-Proxy]
> Proxy-On-Private-Contact=False
> Proxy-On-via-received-vs-signaling-address=False
> Proxy-On-Private-Via=False
> Proxy-On-Different-RPORT=False
> Proxy-All-Media=False
>
> [SIP-Trunk-Config]
> SIP-Trunk-Config=<siptrunk trunk-name="voip.eutelia.it" route-set="voip.eutelia.it" sip-domain="voip.eutelia.it" expires="600">
> SIP-Trunk-Config=<trunk-accounts>
> SIP-Trunk-Config=<account user-name="0302054741" auth-user-name="0302054741" auth-password="xxx" inboundroute="sip:030...@si..." expires="3600" />
> SIP-Trunk-Config=</trunk-accounts>
> SIP-Trunk-Config=</siptrunk>
>
>
>
>
|
|
From: OpenSBC F. <ope...@op...> - 2009-01-19 09:41:22
|
Nope. I had already noticed the document about trunks, there aren't too many documents to be read after all... I tried both B2B and B2B Upper, but nothing. I see no trace of registrations in the log. My configuration is (sorry for the long post) as follows. Help! Thanks everybody! [Solegy] RTTS-Client-Address=83.211.125.71 [OpenSBC-General-Parameters] SIP-Log-Level=5 PTRACE-Log-Level=1 Log-File-Prefix=b2bua SBC-Application-Mode=B2BUpperReg Mode Interface-Address Array Size=0 Enable-Backdoor-Port=True Enable-Trunk-Port=True Enable-Calea-Port=True RTP-Min-Port=10000 RTP-Max-Port=20000 Enable-Local-Refer=False Max-Forwards=70 Encryption-Mode=XOR Encryption-Key=GS Transaction-Thread-Count=10 Session-Thread-Count=10 Alerting-Timeout=30000 Seize-Timeout=60000 SIP-Timer-B=Default SIP-Timer-H=Default Session-Keep-Alive=1800 Session-Max-Life-Span=10800 Max-Concurrent-Session=100 Max-Call-Rate-Per-Second=10 [Upper-Registration] Enable-Stateful-Reg=True Rewrite-TO-Domain=True Rewrite-FROM-Domain=True Route-List Array Size=0 [Internal-DNS-Mapping] Internal-DNS-Map Array Size=0 [Proxy-Relay-Routes] Drop-Routes-On-Ping-Timeout=False Proxy-Resolve-To-URI=True Route-List Array Size=0 [B2BUA-Routes] Enable-Route-Scripting=False Route-Script=b2bua-route.cscript Route-List Array Size=1 Route-List 1=[sip:*@voip.eutelia.it:*] sip:voip.eutelia.it:5060;sip-trunk=true Insert-Route-Header=True Rewrite-TO-URI=True Prepend-ISUP-OLI=False Route-By-Request-URI=True Route-By-To-URI=False Drop-Routes-On-Ping-Timeout=False Use-External-XML=False External-XML-File=b2bua-route.xml Route-List 2=[sip:030...@vo...] sip:030...@si... [Media-Server] Enable-Media-Server=False Media-Server-Number=5000 Codec-List Array Size=0 No-RTP-Proxy-On-All-Transfers=False Enable-Announcement-Service=False 4xx-Error-Map=prompts/basic/cant_complete.wav 5xx-Error-Map=prompts/basic/cant_complete.wav 6xx-Error-Map=prompts/basic/cant_complete.wav Announcement-Error-Map Array Size=0 [Outbound-Proxies] Outbound-Proxies Array Size=0 [Local-Domain-Accounts] Accept-All-Registration=False Account-List Array Size=3 Account-List 1=sip:0302054741:xx...@si... Account-List 2=sip:0302054742:xx...@si... Account-List 3=sip:0302054681:xx...@si... [RTP-Proxy] Proxy-On-Private-Contact=False Proxy-On-via-received-vs-signaling-address=False Proxy-On-Private-Via=False Proxy-On-Different-RPORT=False Proxy-All-Media=False [SIP-Trunk-Config] SIP-Trunk-Config=<siptrunk trunk-name="voip.eutelia.it" route-set="voip.eutelia.it" sip-domain="voip.eutelia.it" expires="600"> SIP-Trunk-Config=<trunk-accounts> SIP-Trunk-Config=<account user-name="0302054741" auth-user-name="0302054741" auth-password="xxx" inboundroute="sip:030...@si..." expires="3600" /> SIP-Trunk-Config=</trunk-accounts> SIP-Trunk-Config=</siptrunk> |
|
From: OpenSBC F. <ope...@op...> - 2009-01-19 03:28:13
|
Yes, OSBC will fit your needs using B2BUA Upper Reg mode. There are many configuration examples in this forum. |
|
From: OpenSBC F. <ope...@op...> - 2009-01-19 03:23:23
|
The following entry in the SIP-Trunk-Config section will map the registered UA 9001 to the external provider account 1001 <account user-name="1001" auth-user-name="1001" auth-password="1001" inbound-route="sip:90...@ex..." expires="3600" /> Please refer to this document for a complete example: http://www.opensourcesip.org:8080/clearspacex/docs/DOC-1040 Also, please note that the OSBC insatnce must be in B2BUA or B2BUA Upper Reg mode. |
|
From: OpenSBC F. <ope...@op...> - 2009-01-15 17:42:08
|
I have a pcx in a network 192.168.2.0, my sip provider is a public operator form Ecuador. So i have a adsl conection to the public operator, I got the static IP 172.29.6.34 Netmask 255.255.255.252. In this case I'am not able to set all the pcx ips on the operator subnet, so I'm looking a solution to NAT the network. We make some test with a Cisco router, and it works if we nat ports 5060 to 192.168.2.8 and the RTP port 30000-35000 to 192.168.2.7. But the router is a bit expensive. So we are looking for a PC solution, that's why I'm looking at OpenSBC. Maybee proxying is not the rigth word, I'm looking to smoething that can redirect (and rewrite sip headers) the sip and rtp as I explain before. If OpenSBC is able to solve my problem, could you give me an exemple of configuration. Thanks |
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