[SR-Users] B2BUA issues

[Alex Balashov]

The short answer to your latter question is: yes. Cisco media and PSTN gateways have never hairpinned SIP-to-SIP calls well, even when officially supported. 

Asterisk has a lower learning curve due to the abundance of information and tutorials, but SEMS would make more sense, since all you need is a signaling B2BUA.




-- Alex

--
Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. 

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/phillman25 <phillman25 at gmail.com> wrote:Hi Alex

Thanks for your prompt reply.

The PGW 2200 solution is used as our core PSTN gateway where its currently handling many SS7, H.323 and SIP interconnections. However, there are a few scenarios like the example described below, that the call is originating from Kamailio being sent to the PGW and then back to Kamailio for termination and this scenario doesn't seem to work.

Do you think that by implementing SEMS or Asterisk in between the PGW and Kamailio could resolve this issue for these specific scenarios?
From your experience what do you think would be a better solution?

Thanks again!
Phillip

========================
Message: 2
Date: Mon, 06 Aug 2012 04:26:28 -0400
From: Alex Balashov <abalashov at evaristesys.com>
Subject: Re: [SR-Users] B2BUA issues
To: sr-users at lists.sip-router.org
Message-ID: <501F7FB4.8040700 at evaristesys.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

What is the larger objective?  Are you using the PGW purely as a B2BUA?
  If so, that's a colossally overblown waste of resources;  just use
something like SEMS or Asterisk.

On 08/06/2012 04:24 AM, phillman25 wrote:

> Dear List
>
> I am trying to accomplish the following:
>
>
> Asterisk PABX (192.168.10.189) ==> Kamailio(xxx.xxx.xxx.xxx)  ==> Cisco
> PGW 2200 (PSTN gateway) (yyy.yyy.yyy.yyy) ==>
> Kamailio(xxx.xxx.xxx.xxx) ==> Asterisk PABX (192.168.10.189)
>
> When trying the above scenario, the call is silent and drops after a few
> seconds. In syslog i observe the following error:
>
> *ERROR: <core> [parser/parse_rr.c:84]: parse_rr(): Error while parsing
> name-addr (sip:22030305 at 192.168.10.189:5060
> <http://sip:22030305@192.168.10.189:5060>>)*
>
> Looking at the sip trace i see that his might be caused by the ACK
> message received from the ASTERISK PABX? :
>
> ACK sip:22030305 at 192.168.10.189:5060
> <http://sip:22030305@192.168.10.189:5060> SIP/2.0
> Via: SIP/2.0/UDP 192.168.10.189:5060;branch=z9hG4bK3c80f516;rport
> Route:
> <sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea;did=97b.66623da5>,<sip:22030305@
> yyy.yyy.yyy.yyy;pgw-call=call-55bc4>,<sip:xxx.xxx.xxx.xxx;lr=on;ftag=as166b1eea>
> Max-Forwards: 70
> From: "22498045" <sip:22498045 at 192.168.10.189
> <mailto:sip%3A22498045 at 192.168.10.189>>;tag=as166b1eea
> To: <sip:22030305 at xxx.xxx.xxx.xxx>;tag=as6d578713
> Contact: <sip:22498045 at 192.168.10.189:5060
> <http://sip:22498045@192.168.10.189:5060>>
> Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778 at 192.168.10.189:5060
> <http://5e2d61160bd1bec9214e2d7d04e5a778@192.168.10.189:5060>
> CSeq: 102 ACK
> User-Agent: FPBX-2.8.1(1.8.12.0)
> Content-Length: 0
>
>
> After contacting Cisco they informed us that issue is cause by B2BUA
> from our current version of Cisco PGW 2200 that doesn't support this
> feature. Is there a module, solution that i can implement on Kamailio
> that could temporarily resolve this issue?
>
> Thanking you in advance.
>
> Phillip
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