[Asterisk-Users] SIP Problem - Asterisk to Provider Gateway

[Gavin Adams]

On Mar 3, 2006, at 1:46 PM, Gavin Adams wrote:

> Hi All,
>
> I'm stumped on a weird problem. I have an * server working fine for  
> local
> SIP phones and IAX2 connections. We just provisioned a second Ethernet
> port to attach to a local SIP provider.
>
> PSTN calls incoming work fine:
>
> PSTN -> SIP Provider -> SIP -> *
>
> but outgoing calls are not. Call setup takes place and the caller  
> can hear
> about 1-2 seconds of audio before the SIP provider cancels the call  
> and
> sends back a BYE message. They haven't made any changes on their end
> (metaswitch).
>

[snip]

Okay, by changing the sip.conf entry to an IP address instead of a / 
etc/host entry has resolved the problem. I'll do further research  
next week to see if it's * or the remote SIP gateway choking on the  
entry.

Regards,

--- Gavin