[Asterisk-Users] SPA-3k outbound calls...

[Mike Benoit]

Interesting. I wonder if thats a way to work around the issue I ran in
to with Asterisk v1.0+. I sent an email to the list about it a while
back, here it is again:

I recently upgraded from a few month old CVS version of Asterisk to
v1.0.1, and dialing out through my SPA-3000 stopped working. 

Notice right after INVITE, in the old CVS version, it includes the
number I'm trying to dial (8019596) which works fine, however in v1.0.1,
it doesn't include the number and of course the dial fails. 

Did a config option change out from underneath me or something?

Old CVS version of Asterisk: (works fine)
--------------------------------
Oct  4 23:18:07 192.168.1.190 INVITE sip:8019596 at 192.168.1.190:5061
SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.3:5060
;branch=z9hG4bK522738f1^M From: "asterisk"
<sip:asterisk at 192.168.1.3>;tag=as52daeb2d^M To: <sip:8019596 at 192.168
.1.190:5061>^M Contact: <sip:asterisk at 192.168.1.3>^M Call-ID:
2d02c0cc392a99264f5f09666c3ff875 at 192.168.1.3^M CS
eq: 102 INVITE^M User-Agent: Asterisk PBX^M Date: Tue, 05 Oct 2004
06:18:07 GMT^M Allow: INVITE, ACK, CANCEL, O
PTIONS, BYE, REFER^M Content-Type: application/sdp^M Content-Length:
214^M ^M v=0^M o=root 27838 27838 IN IP4 1
92.168.1.3^M s=session^M c=IN IP4 192.168.1.3^M t=0 0^M m=audio 13232
RTP/AVP 0 101^M a=rtpmap:0 PCMU/8000^M a=
rtpmap:101 telephone-event/8000^M a=fmtp:101 0-16^M a=silenceSupp:off -
- - -^M


Asterisk v1.0.1: (doesn't work)
---------------------------------
Sep 30 20:38:45 192.168.1.190 INVITE sip:500 at 192.168.1.190:5061
SIP/2.0^M Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK36cac9c5^M
From: "2508019596" <sip:2508019596 at 192.168.1.3>;tag=as7f1bd067^M To:
<sip:500 at 192.168.1.190:5061>^M Contact: <sip:2508019596 at 192.168.1.3>^M
Call-ID: 0701aef72bda06da6e3fb4593dc78e31 at 192.168.1.3^M CSeq: 102
INVITE^M User-Agent: Asterisk PBX^M Date: Fri, 01 Oct 2004 03:38:45
GMT^M Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER^M Content-Type:
application/sdp^M Content-Length: 214^M ^M v=0^M o=root 22051 22051 IN
IP4 192.168.1.3^M s=session^M c=IN IP4 192.168.1.3^M t=0 0^M m=audio
14996 RTP/AVP 0 101^M a=rtpmap:0 PCMU/8000^M a=rtpmap:101 telephone-
event/8000^M a=fmtp:101 0-16^M a=silenceSupp:off - - - -^M


On Sat, 2004-10-09 at 23:16 -0700, Ray wrote:
> On Sat, Oct 09, 2004 at 11:41:41PM -0400, Jeff owen wrote:
> > Ok, now since I have inbound working properly the outbound seemed to have
> > gotten hosed.
> > 
> >  
> > 
> > In the extensions.conf I have it setup as:
> > 
> >  
> > 
> > exten => _9NXXXXXXXXX,1,Dial(SIP/${EXTEN:1}@pstn)
> 
> FWIW I was fighting with this today and had to make this line like:
> 
> exten => _9NXXXXXXXXX,1,Dial(SIP/${EXTEN:1}@ipaddress_of_sipura:5061)
> 
> Maybe if the host was specified in sip.conf rather than being listed as
> dynamic this wouldn't be necessary.
> 
-- 
Mike Benoit <ipso at snappymail.ca>