[Asterisk-Users] asterisk outbound to SIP provider problems

Felipe Martins fmartins at mundivox.com
Mon Mar 14 05:28:34 MST 2005


> Hi

Good Morning,
 
> 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.
> 

I'm having the same problem over here, but with both, inbound/outbound calls, I use a SER server to auth my users, and when I need to use a VoIP line that is not at my server, I use Asterisk to auth the line outside my server at my Foreign Voip server then when I get the line I can dial, but none of them, incoming/outgoing, calls are working fine. How did you configure your incoming call ?


> 
> 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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-- 
Felipe Martins
Mundivox Communications
Tecnologia e Projetos
fmartins at mundivox.com

Tel.: +55 +21 +3820 8839
Cel.: +55 +21 +9823 8602
Fax.: +55 +21 +3820 8844
www.mundivox.com





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