Getting hold music after remote hangup on outgoing calls

Asterisk Asterisk Support
1

When I make an outgoing call and the remote side answers, if the remote side hangsup I get music on hold for a few seconds before the call ends. I am using Asterisk 1.6.2.11. All calls are SIP calls. The calls go through the network to a router, across a T1 to the SIP trunk provider.

Here is the Asterisk CLI

localhostCLI>
== Using SIP RTP CoS mark 5
– Executing [1003@internal:1] Dial(“SIP/6000-0000003a”, “SIP/204.xx.xx.xx/0019056689999”) in new stack
== Using SIP RTP CoS mark 5
– Called 204.xx.xx.xx/0019056689999
– SIP/204.xx.xx.xx-0000003b is ringing
– SIP/204.xx.xx.xx-0000003b is making progress passing it to SIP/6000-0000003a
– SIP/204.xx.xx.xx-0000003b answered SIP/6000-0000003a
– Packet2Packet bridging SIP/6000-0000003a and SIP/204.xx.xx.xx-0000003b
– Started music on hold, class ‘default’, on SIP/6000-0000003a
== Spawn extension (internal, 1003, 1) exited non-zero on ‘SIP/6000-0000003a’
– Stopped music on hold on SIP/6000-0000003a
localhostCLI>

Any help with this would be greatly appreciated. Thanks.

2

Is there any weird setup with the dialplan?

try posting the output of:

dialplan show 1003@internal
OR
dialplan show internal

3

Thanks for the reply.
I am using extension 1003 like a speed dial for testing. It calls a local number which terminates from the PSTN to a phone on my desk. I answer the call, check for audio and then hangup. The phone on my desk is connected to a toshiba pbx which is being replaced by the asterisk box.

localhost*CLI> dialplan show 1003@internal
[ Context ‘internal’ created by ‘pbx_config’ ]
‘1003’ => 1. Dial(SIP/204.xx.xxx.xx/xxxxxxx9056689197) [pbx_config]
2. Hangup() [pbx_config]

I am beginning to suspect that it could be the Toshiba system. I called my cell phone and it doesn’t give me the music on hold it just hangsup correctly.

Will.

4

My guess is that the network doesn’t do called party clearing - fairly common for landlines - but your main PBX has common channel signalling and is seeing the clear signal from the network, passing it on as a SIP hold, then initiating a release, but not sending SIP BYE until the release is confirmed by the network.

clear can be cancelled by re-answer, but release finishes the call for good. The typical re-answer case is where the callee puts down one phone then picks up another one on the same line.

5

Our system is up and live now and the problem has gone away. The old Toshiba system must have been the source of the problem. Thanks for the replies.