[Asterisk-Users] asterisk outbound to SIP provider problems

[w fm3]

Hi

I am having alot of difficulty connecting to SIP providers (I am trying 3) 
and can't seem to find anything similar in the wiki or on the lists.....I 
can receive inbound calls fine however on placing an outbound call the 
calling phone never gets 'connected' but 2 way audio is passed for about 
20secs before some sort of timeout.

Anything suggestions as to what I could try appreciated.

Many thanks

Walt.

--

The call goes like this:

caller: dial
caller: SIP code 100
destination: ring
caller: 1-2 second delay
caller: SIP code 183
caller: ring
destination: pickup
caller: 2 way audio ok
destination: 2 way audio ok
caller: Sip code 183 (Never 200)
caller: some sort of call timout, audio stops
destination: chooses to hang up
caller: chooses to hang up

sip debug peer of a provider:
http://www.walt.9k.com/sip/1_SIP_Provider.html

sip debug peer of phone placing the call
http://www.walt.9k.com/sip/1_cisco_phone.html

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