[asterisk-users] Internal calls without voice transport

[martin f krafft]

Hey,

we're experiencing a weird problem with Asterisk 1.8.13.1
(1:1.8.13.1~dfsg1-3+deb7). Calls that leave and enter Asterisk via
a PBX (sipgate.de) work perfectly fine, almost 100% of the time.

However, calls that are routed to sipgate.de, which then routes the
call back to our Asterisk instance are "silent" most of the time.

What I mean with that is that even though RTP traffic flows, neither
side can hear anything from the other.

This problem happens when people at site A dial someone at site
B using the number provided by sipgate.de, but also if people call
each other within a site through the external number, i.e. if I dial
089-1234567-100 from 089-1234567-200.

I have not been able to reproduce this problem with purely internal
calls, i.e. calling ext. 100 directly, so I am assuming there's
a problem due to sipgate's involvement. However, as far as
I understand, once the call is established (and both parties' phones
suggest that), the traffic flows only via Asterisk (directmedia
= update,nonat), so the problem is likely to be found there, no?

Before I shower you with debug logs and traces, I am wondering if
this sounds familiar to anyone…?

Thanks,

-- 
martin | http://madduck.net/ | http://two.sentenc.es/
 
if god had meant for us to be naked,
we would have been born that way.
 
spamtraps: madduck.bogus at madduck.net
-------------- next part --------------
A non-text attachment was scrubbed...
Name: digital_signature_gpg.asc
Type: application/pgp-signature
Size: 1107 bytes
Desc: Digital signature (see http://martin-krafft.net/gpg/sig-policy/999bbcc4/current)
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140728/a7d0e89a/attachment.pgp>