[Asterisk-Dev] sip outgoing proxy

[Karl Brose]

This seems a variation of a problem I encountered with SIP registration with
a certain provider who apparently wanted a certain formatted domainname.
It's a simply fix in chan_sip.c, there are only a few places where the
SIP Headers get formatted and you can replace them with what you need
depending on
circumstances.  I am running a modified version and it fixed the problem.
I am sure yours requires a similar fix.
Would this be Broadvoice?


> Message: 6
> Date: Tue, 16 Mar 2004 08:48:42 -0500
> From: "Thomas B. Clark" <digium at clark.durham.nc.us>
> To: asterisk-dev at lists.digium.com
> Subject: [Asterisk-Dev] sip outgoing proxy
> Reply-To: asterisk-dev at lists.digium.com
>
> I posted this message to asterisk-users but did not get a response. I am
> hoping somebody here may be interested in this problem.
>
> I have talked one of the major voip providers into giving me my userid
> and password to test with Asterisk. (I cannot say which one right now,
> although this would be very good for asterisk users--lots of rate
> centers, unlimited service.)
>
> Registration and receiving calls work pretty well.  However, I am
> unable to make calls because the provider requires that the sip URI say
> number at SIP.provider.com, but be sent to number at PROXY.provider.com.
>
> Sip bug 359
> http://bugs.digium.com/bug_view_page.php?bug_id=0000359
> appears to address this issue, but is not moving very quickly.
>
> I have tried working around by setting up my own DNS to be authoritative
> for provider.com, and providing a SRV record with a proxy in it, but
> Asterisk is ignoring it.
>
> Can anybody think of a different way to work around this problem?
>