[asterisk-users] Help with Avaya integration

[Steve Totaro]

Answer() is the cure to most problems like this...

While not very intuitive, when you think about the various "legs" of
the call, it starts to make "some sense" although the exact reason has
always eluded me.

My guess it is solves or at least gets you closer to solving your issues

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)



On Thu, Jan 22, 2009 at 8:34 PM, Steven J. Douglas <stevend at moij.biz> wrote:
> Hi David,
>
> Thanks for your suggestion, I will give it a try at the Avaya at my
> client's place.
>
> Here is the set up for the failed call.
>
> Caller -> PSTN ---[Analog DID]---> Avaya ---[IP Trunk (H.323)]--->
> Asterisk ---[SIP]---> SIP Phone.
>
>  From my packet capture on the IP Trunk of the failed call, I can see
> the following call flow.
>
> Avaya -> SETUP -> Asterisk
> Asterisk -> CALL PROCEEDING -> Avaya
> Asterisk -> ALERTING + OpenLogicalChannel -> Avaya
> Asterisk -> CONNECT + OpenLogicalChannel -> Avaya (this was sent when I
> picked up the call from sip phone on asterisk)
>
> Asterisk <- RTP -> Avaya (This is where I still hear ringing from PSTN
> (via Avaya) end and modem-like sounds from the Asterisk end)
>
> Avaya -> RELEASE COMPLETE -> Asterisk
>
> It seems that the answer (and the resulting CONNECT) was already being
> sent to Avaya. But the Avaya did not recognize this and stop the ringing
> on the PSTN side. I'll give your suggestion a try and see if it makes a
> difference.
>
> Thanks.
>
> -Steve
>
> David fire wrote:
>> try a answer() before the dial(sip/xxx)
>> and if you are using originate try local/.... and start whit and answer()
>>
>>
>> 2009/1/22 Steven J. Douglas <stevend at moij.biz <mailto:stevend at moij.biz>>
>>
>>     Hi,
>>
>>     I'm trying to integrate my asterisk PBX to an Avaya PBX. I am using
>>     chan_ooh323 from asterisk-addons.
>>
>>     I am able to make a call from SIP Phone -> Asterisk -> Avaya ->
>>     Station
>>     (phone) and vice versa.
>>     I am also able to make a call from SIP Phone -> Asterisk -> Avaya
>>     -> PSTN.
>>
>>     However I face problems when I make DID calls from the PSTN. The DID
>>     calls are made through analog DID lines to the TN753 on the Avaya.
>>     When
>>     I make the call, I can hear ringing on the caller phone (PSTN) and the
>>     SIP Phone rings. But when I pick up the SIP Phone, the caller phone
>>     remains in ringing mode. On the SIP Phone, I hear random sound.
>>
>>     I did a packet capture and on the Q.931 setup information header,
>>     under
>>     Progress Indicator, the call is not end-to-end ISDN. So it seems that
>>     the SIP answer message is not being communicated properly to the Avaya
>>     PBX. Can this be the cause of the problem?
>>
>>     Has anyone encountered this problem and what is your solution?
>>
>>     Thanks in advance.
>>
>>     Regards,
>>     Steve
>>
>>
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>
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