[Asterisk-Users] SIP VoIP Provider problems

[w fm3]

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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