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 'firstname.lastname@example.org’ in 32000 ms (Method: REGISTER)
the complete SIP debug info can be found here
I found some info.
asteriskguru.com/archives/im … 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.