asterisk users - Mar 2005 - asterisk outbound to SIP provider problems

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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> 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@mundivox.com

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

>I'm having the same problem over here, but with both, inbound/outbound 
>calls, I use a SER server >auth my users, and when I need to use a VoIP 
>line that is not at my server, I use Asterisk to auth >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
I don't use SER but as far as inbound Provider->asterisk it 'just
worked out
of the box' without any problems for me. Making sure that

- authentication parameters are correct with sip.conf,
- you are registering with them fine
- registration line has a  /internalextension after it so it knows where to 
put the inbound call
- corresponding entry in extensions.conf relating to the above - internal 
extension above dialing a physical phone / group or whatever.
- asterisk ports are open / accessible from your provider
- codecs are compatible

also

- if you are making config changes to try and get things to work u r  better 
to stop and restart asterisk rather than 'sip reload'
- try both cvs and an older version, sometimes things break
- if you have multiple cards in the machine use nat=yes for the outbound 
provider to force asterisk to use the correct IP address in the sip 
messages.
- I would use asterisk for a test setup and once that is working  add SER to 
the equation.
- try a few different providers to make sure any problems are not provider 
specific.

NOTE these are notes from someone who CAN'T get SIP outbound calls to be 
placed correctly so take it with a bag of salt.... lol ... Good luck.

Walt

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