asterisk users - Oct 2011 - Queue announcements and MOH blanking out on calls from PSTN over IAX2

Hi,

Problem with Asterisk 1.6.2.9 on Debian Squeeze.

* Infrastructure

We have two servers, SIP and Dial.

The SIP server handles SIP clients; it also receives incoming PSTN calls 
from the Dial server and makes outgoing PSTN calls on the Dial server.

The Dial server is connected to multiple 4-port Redfone devices for 
handling PSTN incoming and outgoing calls.  Outgoing calls always 
originate from and incoming calls always terminate at the SIP server.  
SIP and Dial servers are connected over IAX2.

Normal incoming and outgoing have been working fine for many months now 
on this setup.

* Problem

We recently enabled caller queues on the SIP server.  Queue functions 
are working fine.  Local (on SIP server itself) callers get periodic 
position announcements and MOH while they wait for the call to be picked 
up.  Once an agent picks up the call, the caller and agent can talk 
normally.  Tried with an IAX2 client (instead of SIP) and that works 
fine too.

Callers coming in from the PSTN (through the Dial server, over IAX2) can 
also talk normally after an agent has picked up the call.  However, 
callers from the PSTN get the announcement and/or MOH blanked out after 
a random period of time, typically 5-10 seconds.  This often happens in 
the middle of the queue position or thank-you announcement.

After the blanking out, the call is still alive, queue functions are 
working, and if an agent picks up the calls s/he can talk normally to 
the caller.  However, blanking out of the MOH/announcement makes the 
caller think that the call has been dropped, and they hang up before an 
agent answers.

Debug logs show that Asterisk is playing the MOH and announcement files 
continuously even though the caller cannot hear them.

Unable to figure out why the blanking happens ONLY on incoming calls 
from the PSTN.  Any help appreciated.

Regards,

-- Raj
-- 
Raj Mathur                raju at kandalaya.org      http://kandalaya.org/
       GPG: 78D4 FC67 367F 40E2 0DD5  0FEF C968 D0EF CC68 D17F
PsyTrance & Chill: http://schizoid.in/   ||   It is the mind that moves

On Saturday 29 Oct 2011, Raj Mathur (??? ?????) wrote:
> [snip]
> Callers coming in from the PSTN (through the Dial server, over IAX2)
> can also talk normally after an agent has picked up the call. 
> However, callers from the PSTN get the announcement and/or MOH
> blanked out after a random period of time, typically 5-10 seconds. 
> This often happens in the middle of the queue position or thank-you
> announcement.
> 
> After the blanking out, the call is still alive, queue functions are
> working, and if an agent picks up the calls s/he can talk normally to
> the caller.  However, blanking out of the MOH/announcement makes the
> caller think that the call has been dropped, and they hang up before
> an agent answers.
> 
> Debug logs show that Asterisk is playing the MOH and announcement
> files continuously even though the caller cannot hear them.
> 
> Unable to figure out why the blanking happens ONLY on incoming calls
> from the PSTN.  Any help appreciated.
Further simplified the issue to an extension that just does:

... Answer()
... MusicOnHold(default)

When called from the PSTN, the musiconhold blanks out after a few 
seconds, while it plays fine when the extension is called locally.

Regards,

-- Raj
-- 
Raj Mathur                raju at kandalaya.org      http://kandalaya.org/
       GPG: 78D4 FC67 367F 40E2 0DD5  0FEF C968 D0EF CC68 D17F
PsyTrance & Chill: http://schizoid.in/   ||   It is the mind that moves

Try turning on the Sip debug for the PSTN call as well as RTP debug. Paste
the output here.
>
> The Dial server is connected to multiple 4-port Redfone devices for
> handling PSTN incoming and outgoing calls.  Outgoing calls always
> originate from and incoming calls always terminate at the SIP server.
> SIP and Dial servers are connected over IAX2.

Explain the above abit as well..couldnt get the clear picture of what it
looks like. Seems to me that you guys are using two servers and call-audio
gets lost in between the servers OR in between the Dial-Server and redfone
device for Queue Calls.


2011/10/29 Raj Mathur (??? ?????) <raju at linux-delhi.org>
> On Saturday 29 Oct 2011, Raj Mathur (??? ?????) wrote:
> > [snip]
> > Callers coming in from the PSTN (through the Dial server, over IAX2)
> > can also talk normally after an agent has picked up the call.
> > However, callers from the PSTN get the announcement and/or MOH
> > blanked out after a random period of time, typically 5-10 seconds.
> > This often happens in the middle of the queue position or thank-you
> > announcement.
> >
> > After the blanking out, the call is still alive, queue functions are
> > working, and if an agent picks up the calls s/he can talk normally to
> > the caller.  However, blanking out of the MOH/announcement makes the
> > caller think that the call has been dropped, and they hang up before
> > an agent answers.
> >
> > Debug logs show that Asterisk is playing the MOH and announcement
> > files continuously even though the caller cannot hear them.
> >
> > Unable to figure out why the blanking happens ONLY on incoming calls
> > from the PSTN.  Any help appreciated.
>
> Further simplified the issue to an extension that just does:
>
> ... Answer()
> ... MusicOnHold(default)
>
> When called from the PSTN, the musiconhold blanks out after a few
> seconds, while it plays fine when the extension is called locally.
>
> Regards,
>
> -- Raj
> --
> Raj Mathur                raju at kandalaya.org      http://kandalaya.org/
>       GPG: 78D4 FC67 367F 40E2 0DD5  0FEF C968 D0EF CC68 D17F
> PsyTrance & Chill: http://schizoid.in/   ||   It is the mind that moves
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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hmmm so  IAX channel is playing with you guys.

1- Cant you guys use SIP, does this happen with SIP trunk as well !?
2- Which version of asterisk are there on both servers.
3- See the output of the command "core show file versions" in your
both
asterisk servers. Mainly lookout for IAX channel version.

Also try enabling IAX debug and paste the output on console.



2011/10/30 Raj Mathur (??? ?????) <raju at linux-delhi.org>
> On Sunday 30 Oct 2011, Raj Mathur (??? ?????) wrote:
> > After looking further, the problem seems to be purely in playing
> > recorded messages over IAX2.  Looking at the debug logs on the SIP
> > server (which is playing the recorded messages) shows that it stops
> > playing one of the messages at some point in the flow, and then never
> > plays anything again.
>
> This seems to be very similar to:
>
>  https://issues.asterisk.org/view.php?id=17232
>
> except there is no virtualisation involved in the process -- everything
> is working on native hardware.  It /is/ amd64 Debian Squeeze running on
> Intel, though.
>
> Regards,
>
> -- Raj
> --
> Raj Mathur                raju at kandalaya.org      http://kandalaya.org/
>       GPG: 78D4 FC67 367F 40E2 0DD5  0FEF C968 D0EF CC68 D17F
> PsyTrance & Chill: http://schizoid.in/   ||   It is the mind that moves
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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