[asterisk-dev] SIP trunking

[John Martin]

Anton,
  We've been using Asterisk to trunk to a Huawei switch for over 12
months now, you may have a later build of the Huawei code but here's
what we found. 

  You can certainly get odd behaviour - denied was quite a common
response I seem to remember. We found it doesn't like a number of things
being sent to it:
- SIP Fast Update Requests using XML, it needs to see them as old
fashioned RTCP/H.261
- silence suppression in SDP, you have to remove:

	if(!p->owner || !ast_internal_timing_enabled(p->owner))
		ast_build_string(&a_audio_next, &a_audio_left,
"a=silenceSupp:off - - - -\r\n");

from chan_sip.c

Not sure it'll help you, but it worked for us when hooking up to a
Huawei switch.

There were other valid SIP messages that the Huawei didn't like at the
time, but nothing that Asterisk used in our installation.

Olle, I think we had a talk about this in Paris and we decided it had to
stay in the code, but for Huawei it made sense for us. I should also
probably be on the users forum by now :-)

John

> -----Original Message-----
> From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-
> bounces at lists.digium.com] On Behalf Of Anton
> Sent: 06 March 2007 16:55
> To: Asterisk Developers Mailing List
> Subject: Re: [asterisk-dev] SIP trunking
> 
> Olle,
> 
> I'll try to find out from the Huawei what is their SIP
> trunking is. As I currently know they do use SIP-T as
> transport, but in specs it's also stated that plain SIP is
> supported too. Current interconnect behaviour is quite
> funny, when I try to call their phone - it rings, but if
> someone pick the phone up, my asterisk starting playing me
> MOH, and remote side insists that I send them "denied".
> 
> On 6 March 2007 15:35, Olle E Johansson wrote:
> > 1 mar 2007 kl. 17.52 skrev Anton:
> > > any plans to support SIP Trunking?
> >
> > As previously said, it's a very generic question. The
> > version of SIP trunking I want to support
> > is the SIP Forum's SIPconnect specification. It's part of
> > my work with codename pineapple
> > (http://www.codename-pineapple.org) to implement an
> > object called "trunk" in Asterisk.
> >
> > Having said that, with proper configuration we can set up
> > proper SIP trunks with most
> > equipment today. If you have any specific issue or
> > variant we do not support, let us
> > know more details about it.
> >
> > Best regards,
> > /Olle
> >
> >
> > ---
> > Olle E. Johansson * Asterisk Evangelist, developer * VOOP
> > A/S olle at voop.com
> >
> >
> >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-dev mailing list
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-dev
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
> 
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-dev
> 
>
________________________________________________________________________
> This e-mail has been scanned for all viruses by Star. The
> service is powered by MessageLabs. For more information on a proactive
> anti-virus service working around the clock, around the globe, visit:
> http://www.star.net.uk
>
________________________________________________________________________