[SR-Users] B2BUA issues

phillman25 phillman25 at gmail.com
Mon Aug 6 14:59:57 CEST 2012


Ok Alex thanks for the info!

On Mon, Aug 6, 2012 at 3:05 PM, Alex Balashov <abalashov at evaristesys.com>wrote:

> 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 at 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/ if 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 at 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 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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