Call ends after 32 seconds

I use a generated call file to setup a call. The call is setup between a local channel (actually an extension) that dials the external number and an Asterisk extension that start a recording after the beep, using Record()

I have two different sip trunks available and one trunk works fine but is mend for my private phone calls. The other one, mend to be used for the service offered, is giving problems because the connection ends after 32 seconds.

I enabled sip debug and “core set debug 10” and captured the output with #asterisk -r > /home/erik/sipdebug

The line showing the problem is:

Scheduling destruction of SIP dialog '7f34e9117e68dad470385cab45a5e064@xx.xx.241.180’ in 32000 ms (Method: REGISTER)

the complete SIP debug info can be found here

I found some info. … 90552.html Jared Smith wrote that it isn’t a problem because it is just a simple an informative message from the SIP channel driver that can be safely ignored. But the connection really ends after 32 seconds and that is hard to ignore.

Google presents, doing a search on “Scheduling destruction of SIP dialog 32000”, more search results but no real explanation or solution.

I’m wondering if this is a ITSP issue (one of the two siptrunks is working without a problem) or if it is an Asterisk related issue that can be solved changing settings. Any suggestion or pointer is very welcome.

The register timeout message is completely normal and is cancelled by

222.Really destroying SIP dialog '2175985755b07ecd1cd804db4a708021@xx.xx.241.180’ Method: REGISTER

The called party has cleared the call, normally. You will have to look at its logs, although it might be a lack of RTP due bad NAT configuration.

The problem is that the called party is an ITSP (internet telephone service provider) and I don’t have access to the sip debug info of this ITSP. The ITSP takes care for the interconnection to the pstn network. It happens when the system is calling out. Before I call the support desk of the ITSP I want to be sure that the problem isn’t caused by the Asterisk server that is under my control.

My Asterisk server isn’t behind nat and the ITSP is also not behind nat so it is unlikely that there is a nat issue involved.

Am I right with the conclusion that the ITSP is causing the problem because of bad response to a SIP message send by the Asterisk server? Or is it something else and is there anything I can fix myself? Thanks in advance!


There is no evidence of anything wrong on those traces. The ITSP is clearing the call completey normally. You need their logs to understand why they are clearing the call, but it would have to be either a failure to receive the ACK (unlikely) or a failure to receive RTP.