Asterisk question/ SIP Native Bridging

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Asterisk question/ SIP Native Bridging

Michael Rosario

Hello.  I am a newbie in the world of asterisk.   I am currently
researching a problem where calls may be dropped.  I am trying to
determine if Asterisk could be at fault.   The call life time operates
in the following manner.
1) The asterisk server calls an internal extension on our phone
system.
2) The asterisk server calls an external destination.
3) The calls are bridged.

In this segment, the Asterisk server is connecting a channel to a
local or internal extension.  In this case, extension 621.

    -- Executing [621@local_click_to_call:1] Dial("Local/
621@local_click_to_call-cda1,2", "SIP/621@avaya") in new stack
    -- Called 621@avaya
    -- SIP/avaya-09236488 is ringing
    -- SIP/avaya-09236488 is making progress passing it to Local/
621@local_click_to_call-cda1,2
    -- SIP/avaya-09236488 answered Local/621@local_click_to_call-
cda1,2
       > Channel Local/621@local_click_to_call-cda1,1 was answered.
    -- Executing [005194075732000@destination_click_to_call:1] AGI
("Local/621@local_click_to_call-cda1,1", "agi://localhost") in new
stack
  == Spawn extension (local_click_to_call, 621, 1) exited non-zero on
'Local/621@local_click_to_call-cda1,2'
    -- AGI Script Executing Application: (SipAddHeader) Options: (P-
Asserted-Identity:BUMSCIS<sip:[hidden email]>)
    -- AGI Script agi://localhost completed, returning 0

In this segment, the Asterisk server is connecting a channel to a
destination.  Our main CKG phone number.

-- Executing [005194075732000@destination_click_to_call:2] Dial("SIP/
avaya-09236488", "SIP/94075732000@avaya||r")
in                                      new stack
    -- Called 94075732000@avaya
    -- SIP/avaya-0935d310 is ringing
    -- SIP/avaya-0935d310 is making progress passing it to SIP/
avaya-09236488

The final steps bridges the channels together.

    -- Native bridging SIP/avaya-09236488 and SIP/avaya-0935d310






Here are my questions:
1) I can't seem to find a solid definition of "native bridging."  The
best one I could find is here: http://www.asterisk.org/doxygen/1.2/Def_Channel.html
Does anyone else have simple way to explain this feature of asterisk?


2) Could unknown instability in our asterisk server result in dropped
calls after the two SIP end points are bridged?  In other words, after
two SIP end points are bridged, technology wise is the Asterisk server
still in the picture and a point of failure??


Thanks so much!



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Re: Asterisk question/ SIP Native Bridging

JasonGoecke

You may actually be better talking to the folks on the #asterisk
irc.freenode.net channel or @ http://forums.digium.com/, as this is a
more Asterisk specific issue than an Adhearsion one.

Putting my dCAP hat on now, and being familiar with your site, I don't
believe the Asterisk bridging is the culprit here. I do believe there
may be some instability between the Asterisk and the Avaya SES server
that may be creating some dropped calls. I recall this was one of the
core issues with a previous Asterisk based system that you had in
place. If this is now happening with a vanilla Asterisk system, I
would suspect the Avaya SES as the culprit.

What you might do, to try and test this, is to take another SIP
endpoint besides Asterisk, like maybe some SJPhones, or something like
this. Have each of them do a SIP URI dial to one station and a
conference of a SIP URI dial to another station. If the SJPhones drop
as well, then you have eliminated Asterisk and may look towards the
Avaya SES more definitively.

If the Avaya SES is no longer in the middle and you are still having
this problem, then my recommendation is the various Asterisk forums.

Good luck!

On Mar 11, 2:18 pm, Michael Rosario <[hidden email]>
wrote:

> Hello.  I am a newbie in the world of asterisk.   I am currently
> researching a problem where calls may be dropped.  I am trying to
> determine if Asterisk could be at fault.   The call life time operates
> in the following manner.
> 1) The asterisk server calls an internal extension on our phone
> system.
> 2) The asterisk server calls an external destination.
> 3) The calls are bridged.
>
> In this segment, the Asterisk server is connecting a channel to a
> local or internal extension.  In this case, extension 621.
>
>     -- Executing [621@local_click_to_call:1] Dial("Local/
> 621@local_click_to_call-cda1,2", "SIP/621@avaya") in new stack
>     -- Called 621@avaya
>     -- SIP/avaya-09236488 is ringing
>     -- SIP/avaya-09236488 is making progress passing it to Local/
> 621@local_click_to_call-cda1,2
>     -- SIP/avaya-09236488 answered Local/621@local_click_to_call-
> cda1,2
>        > Channel Local/621@local_click_to_call-cda1,1 was answered.
>     -- Executing [005194075732000@destination_click_to_call:1] AGI
> ("Local/621@local_click_to_call-cda1,1", "agi://localhost") in new
> stack
>   == Spawn extension (local_click_to_call, 621, 1) exited non-zero on
> 'Local/621@local_click_to_call-cda1,2'
>     -- AGI Script Executing Application: (SipAddHeader) Options: (P-
> Asserted-Identity:BUMSCIS<sip:[hidden email]>)
>     -- AGI Script agi://localhost completed, returning 0
>
> In this segment, the Asterisk server is connecting a channel to a
> destination.  Our main CKG phone number.
>
> -- Executing [005194075732000@destination_click_to_call:2] Dial("SIP/
> avaya-09236488", "SIP/94075732000@avaya||r")
> in                                      new stack
>     -- Called 94075732000@avaya
>     -- SIP/avaya-0935d310 is ringing
>     -- SIP/avaya-0935d310 is making progress passing it to SIP/
> avaya-09236488
>
> The final steps bridges the channels together.
>
>     -- Native bridging SIP/avaya-09236488 and SIP/avaya-0935d310
>
> Here are my questions:
> 1) I can't seem to find a solid definition of "native bridging."  The
> best one I could find is here:http://www.asterisk.org/doxygen/1.2/Def_Channel.html
> Does anyone else have simple way to explain this feature of asterisk?
>
> 2) Could unknown instability in our asterisk server result in dropped
> calls after the two SIP end points are bridged?  In other words, after
> two SIP end points are bridged, technology wise is the Asterisk server
> still in the picture and a point of failure??
>
> Thanks so much!
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