Dears,

My scenario is to accept the call from user àAnswer the call -àplay mohà
dial(SIP/Trunk,XXXXX)

The problem is when the user send the bye the trunk call will not hangup

How to solve this issue

exten => 446696,1,Ringing

exten => 446696,n,Answer()

exten => 446696,n,Wait(2)

exten => 446696,n,Playback(Welcome)

exten => 446696,n,Dial(SIP/Trunk/${EXTEN},300)

exten => 446696,n,Hangup

How to solve such issue

Thanks in advance

winmail.dat

We have an Ast 1.6 installation which is connected to an Avaya using ooh323. Something is causing the log to fill with “In ooEndCall call state is – OO_CALL_CLEAR (incoming, ooh323c_1)” messages every 100ms. This causes the log to grow to 300MB in just 5 minutes, which eventually overloads the box.

Looking through the ooh323 log below, I suspect this stems from the “Error:Failed to enqueue ReleaseComplete message to outbound queue.(incoming, ooh323c_1)” message – but we don’t don’t see enough H323 installations to dig deeper. Can someone offer some suggested causes and resolutions?

Thanks!

On 12-06-03 03:56 PM, Matt Hamilton wrote:
>
> We would like to automate Shared Line Appearance testing (e.g. phoneA answers a call, puts in on hold, phoneB picks up the call on hold) in our lab. Are there any tools/SIP call generators/clients that may help us create such a scenario?
>
Check out the asterisk testsuite for some examples[1]. You could use a
combination of StarPy and pjsua (python bindings) to do this.

[1] http://svnview.digium.com/svn/testsuite/asterisk/trunk/

All,

We are having issues with one of our customers. They typically are
using remote sip clients on smart phones. For the purpose of allowing
the apps to work properly in the background we have to utilize TCP which
works fine.

The problem comes up when the softphone loses connectivity for some
reason. The session timers are not ending the call as they do on a UDP
session. Basically from the sip debug it sends the re-invite for the
session timer according to the sip debug and it appears all is fine
instead of not getting a response back from the client and disconnecting
the call as it does with udp. There is no way it is getting a response
back from the client however as the client has no network connectivity.

I have run some tcpdump’s on the server and when tracing the call I
actually never see those re-invites going out at all from the server.

We are running asterisk 1.8.7.0 currently.

I can reproduce the issue at will by establishing a call from a
softphone and then putting it into airplane mode to simulate the
connectivity loss.

Are session-timers expected to work with tcp? If so can anyone tell me
where to look to see what might be going on?

Thanks in Advance.

Hello,

I am trying to figure out how to make group call using DAHDI. I want to
make multiple call at once and conference among them. I know about meetme,
but that is for incoming conference call.

Please suggest.