Does anyone currently use Vonage with Asterisk? I've tried to set
it up but it looks like Asterisk (at least the version that I have)
does not handle well the SIP call dialog, sending a BYE with the wrong
tag. As a result, when I hang up, Vonage sends back a 400 Bad
Request and the call on the PSTN side does not hang up. <br>
I know that Vonage does a lot of nasty stuff which impacts UA's but
Xten Eyebeam handles it correctly at least. I have tried
pedantic=yes but no difference.<br>
Here is the sip.conf and the BYE dialog with numbers replaced:<br>
<br>
[vonage]<br>
type=friend<br>
secret=<pass><br>
username=<no><br>
host=<a href="http://sphone.vopr.vonage.net">sphone.vopr.vonage.net</a><br>
dtmfmode=rfc2833<br>
port=5061<br>
fromuser=<no><br>
fromdomain=<a href="http://sphone.vopr.vonage.net">sphone.vopr.vonage.net</a><br>
canreinvite=no<br>
context=context<br>
insecure=very<br>
<br>
<br>
BYE sip:(PSTN Number)@<a href="http://216.115.20.171:5060">216.115.20.171:5060</a> SIP/2.0<br>
Via: SIP/2.0/UDP (Asterisk IP):5070;branch=z9hG4bK1dc3ea2d<br>
Route: <sip:(PSTN Number)@<a href="http://216.115.20.171:5060">216.115.20.171:5060</a>><br>
From: "Adrian" <sip:(Vonage No)@<a href="http://sphone.vopr.vonage.net">sphone.vopr.vonage.net</a>>;tag=as74d54cec<br>
To: <sip:(PSTN Number)@<a href="http://sphone.vopr.vonage.net:5061">sphone.vopr.vonage.net:5061</a>>;tag=2067764114<br>
Contact: <sip:(Vonage No)@(Asterisk IP):5070><br>
Call-ID: <a href="mailto:307c340e67a787f202ed0ed55572c4ee@sphone.vopr.vonage.net">307c340e67a787f202ed0ed55572c4ee@sphone.vopr.vonage.net</a><br>
CSeq: 104 BYE<br>
Proxy-Authorization: Digest username="(Vonage No)",
realm="<a href="http://216.115.25.198">216.115.25.198</a>", algorithm=MD5, uri="sip:<a href="http://216.115.25.198">216.115.25.198</a>",
nonce="18861432149", response="5de1aaac0fa9db87sdfb074a1fe324b",
opaque=""<br>
Content-Length: 0<br>
<br>
<br>
---<br>
== Spawn extension (default, 8(PSTN Number), 3) exited non-zero on 'SIP/370-29aa'<br>
Destroying call '<a href="mailto:2751229817@192.168.0.150">2751229817@192.168.0.150</a>'<br>
<br>
<-- SIP read from <a href="http://216.115.25.198:5061">216.115.25.198:5061</a>: <br>
SIP/2.0 400 Bad Request<br>
Via: SIP/2.0/UDP (Asterisk IP):5070;branch=z9hG4bK1dc3ea2d<br>
From: "Adrian" <sip:(Vonage No)@<a href="http://sphone.vopr.vonage.net">sphone.vopr.vonage.net</a>>;tag=as74d54cec<br>
To: <sip:(PSTN Number)@<a href="http://sphone.vopr.vonage.net:5061">sphone.vopr.vonage.net:5061</a>>;tag=2067764114<br>
Call-ID: <a href="mailto:307c340e67a787f202ed0ed55572c4ee@sphone.vopr.vonage.net">307c340e67a787f202ed0ed55572c4ee@sphone.vopr.vonage.net</a><br>
CSeq: 104 BYE<br>
Max-Forwards: 15<br>
Content-Length: 0<br>
<br>
The issue I think is that Asterisk uses the "To" tag from the 183
Session Progress instead of the tag from the 200 OK that Vonage sends.<br>
If anyone uses Vonage with Asterisk and it works fine for you (ie.
landline hangs up when you hang up), can you please let me know which
version you're using? (I'm using CVS HEAD from a couple of months ago
and would like to know if an upgrade may fix the issue.)<br>
<br>