[Asterisk-Users] SIP VoIP Provider problems

[Pedro]

Sounds like you are having a codec issue with 2 of  your providers. 
Make sure you find out what codecs are supported and that your config
is set up accordingly.


On Sun, 06 Mar 2005 00:14:05 +0000, w fm3 <wfm3 at hotmail.com> wrote:
> Hi
> 
> Hope someone can help :)
> 
> I am testing 4 PSTN termination providers. 3 SIP and 1 IAX
> 
> IAX and 1 of the SIP providers work fine.
> 
> Now the wierdness:
> 
> 2 SIP providers I can only get oubound calls to ring at the destination and
> then nothing more. 1 gets as far as SIP code 183 (and ringing on the src
> handset ...yay) the other doesn't get past 100.
> 
> Added to this inbound calls (PSTN->provider->asterisk->handset) work fine
> 100% of the time.
> 
> I have tried alot of config options from the wiki and lists but can't seem
> to get any further.  AFAIK from sip debug and the console it looks like
> that the call is placed  and then no further  communication. Looks like they
> might be using SER / CISCO GW at the VOIP Provider end.
> Don't think it a open UDP port type thing.
> 
> Cheers
> 
> Walt
> 
> PS Newbie
> 
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