For your relatively narrow, specific applications in your topology, no. You'd be better off installing SEMS per se.
-- 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@gmail.com wrote:Hello Alex
Will try with SEMS first, found something called sip:provider CE v2.4 from http://www.sipwise.com/news/announcements/spce-v2_4-release/%C2%A0if i'm not mistaken, this seems to combine Kamailio with SEMS? Do you think that this might be an easier installation rather than installing SEMS on its own as it seems to provide more documentation?
Thanks again!
On Mon, Aug 6, 2012 at 1:33 PM, Alex Balashov abalashov@evaristesys.com wrote: 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@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@evaristesys.com Subject: Re: [SR-Users] B2BUA issues To: sr-users@lists.sip-router.org Message-ID: 501F7FB4.8040700@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@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@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@192.168.10.189 mailto:sip%3A22498045@192.168.10.189>;tag=as166b1eea To: sip:22030305@xxx.xxx.xxx.xxx;tag=as6d578713 Contact: <sip:22498045@192.168.10.189:5060 http://sip:22498045@192.168.10.189:5060> Call-ID: 5e2d61160bd1bec9214e2d7d04e5a778@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