[Asterisk-Users] Asterisk/Metaswitch trunk, no inbound RTP stream on inbound calls

[Clint Sharp]

I've been racking my brain for the last two days to try to figure out
what I could possibly be doing wrong in my configuration for a SIP trunk
that's setup through my local ISPs Metaswitch.  I've setup a very simple
SIP Peer, which I've played around with a lot in the past two days but
still comes back to the following basic setup:

 

[provider-fireball]

type=friend

insecure=very

host=1.2.3.4

context=keysystem

nat=yes

canreinvite=no

username=1235551212

fromuser=1235551212

secret=mysecret

disallow=all

allow=ulaw

 

For whatever reason, on inbound calls, the RTP stream coming from the
provider never initiates.  The RTP stream on my side starts as soon as
my dialplan is set to Answer() and we send a 200 OK back to the
Metaswitch.  However, after the 200 OK, we never receive an inbound RTP
stream.  There are no known configuration changes on their side that
would cause this nor any configuration changes on my side.  It's a very
strange problem.  Does anyone out there have any experience with interop
between Asterisk and Metaswitch?  More importantly, has anyone ever seen
an issue where inbound SIP signaling works fine but no RTP inbound and
it's definitely not a firewall issues (verified with multiple packet
traces before and after the firewall).  Outbound calls work fine with
the inbound and outbound RTP streams both good.  I have plenty of packet
traces available for people who are interested.  Interestingly enough, a
Linksys PAP2 on the same network works fine with the same switch.  Any
ideas?

 

Clint

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060605/3c5010cb/attachment.htm