We are running SIP trunks on a PBX behind a Cisco ASA. Inbound calls come through without a hitch but outbound calls only last 30 seconds before being disconnected.
Things we have tried:
Adding local networks of phones to Asterisk SIP Settings
Configuring NAT settings in Asterisk SIP Settings
Updating Cisco ASA to disable SIP-Alg and increased udp timeout
In Asterisk SIP Settings, check that External Address and Local Networks are correctly set. If you have overridden these in chan_pjsip or chan_sip settings, make sure they are also correct.
If no luck, at the Asterisk command prompt, type pjsip set logger on
or sip set debug on
(according to trunk type) make a failing test call, paste the relevant section of the Asterisk log at https://pastebin.freepbx.org and post the link here.
Thanks for the quick reply. A new development reported by users is that they are getting the same results on extension to extension dialing as well. Attached is the pjsip log, I’ll post another post with sip debug.
[2020-07-09 16:36:45] VERBOSE res_pjsip_logger.c: <— Received SIP response (503 bytes) from UDP:10.40.8.76:49049 —>
I’m not finding anything that says INVITE SIP, is there a different log command I should be running? Here’s the standard asterisk log for the full length of a call attempt. Hopefully its helpful. Thanks again for the help.
When you do pjsip set logger on
or sip set debug on
(depending on trunk type) the SIP trace should appear in the regular Asterisk log, along with the normal entries. Note that when you Apply Config or otherwise reload Asterisk, the logger/debug commands are canceled and need to be issued again.
Yes, trunk is disconnected on 29 seconds, because it execute hangup
Executing [h@restrictedroute-c4ca4238a0b923820dcc509a6f75849b:1] Hangup("PJSIP/4238-00000082", "") in new stack.
How about by disabling reinvite on PJSIP trunks by setting timer=no?
Since the are post with similar issue, but with different disconnected time.
May be it could help.
Workaround via «Admin»→«Config Edit»→pjsip.endpoint_custom_post.conf (replace myTrunkX with name of your pjsip trunk from «Connectivity»→«Trunks» or CLI “pjsip show endpoints”… note the (+) which indicates add this to an already present config.