[Asterisk-Users] asterisk outbound to SIP provider problems (still)

[w fm3]

Hi

I am using cvs and updating it every couple of days Unfortunately I am still 
getting a 20 second timeout on  sip calls placed to various providers, can 
anyone see anything wrong from sip debugs? Or have any ideas what the 
problem might be?

Cheers

Walt

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


The call goes like this:

caller: dial
caller: SIP code 100
destination: ring
caller: 1-2 second delay
caller: SIP code 183 (this is what it says on the cisco phone)
caller: ring
destination: pickup

caller: 2 way audio ok
destination: 2 way audio ok
caller: Sip code 183 (Never 200 connected etc)
caller: audio stops

destination: chooses to hang up
caller: chooses to hang up

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