Calls disconnect after 15 minutes

Hi all,

I am using Asterisk 1.6.2.13 on Elastix 2.0.2. I am in the UK using Sipgate as my VoIP provider. All calls that proceed for 15 minutes are disconnected at the 15 minute milestone, incoming and outgoing. Looking at the Asterisk logs I see this:-
[Aug 4 09:45:37] VERBOSE[4347] pbx.c: – Executing [s@macro-dial:7] Dial(“SIP/sipgate_jonny-000000e5”, “SIP/202,15,tr”) in new stack
[Aug 4 09:45:37] VERBOSE[4347] netsock.c: == Using SIP RTP TOS bits 184
[Aug 4 09:45:37] VERBOSE[4347] netsock.c: == Using SIP RTP CoS mark 5
[Aug 4 09:45:37] VERBOSE[4347] app_dial.c: – Called 202
[Aug 4 09:45:37] VERBOSE[4347] app_dial.c: – SIP/202-000000e6 is ringing
[Aug 4 09:45:46] VERBOSE[4347] app_dial.c: – SIP/202-000000e6 answered SIP/sipgate_jonny-000000e5
[Aug 4 09:50:01] VERBOSE[2649] asterisk.c: – Remote UNIX connection
[Aug 4 09:50:01] VERBOSE[4365] asterisk.c: – Remote UNIX connection disconnected
[Aug 4 09:55:01] VERBOSE[2649] asterisk.c: – Remote UNIX connection
[Aug 4 09:55:01] VERBOSE[4374] asterisk.c: – Remote UNIX connection disconnected
[Aug 4 10:00:01] VERBOSE[2649] asterisk.c: – Remote UNIX connection
[Aug 4 10:00:01] VERBOSE[4384] asterisk.c: – Remote UNIX connection disconnected
[Aug 4 10:00:37] VERBOSE[2841] chan_sip.c: – Got SIP response 420 “Option Disabled” back from 217.10.79.23
[Aug 4 10:00:37] VERBOSE[4347] pbx.c: – Executing [h@macro-dial:1] Macro(“SIP/sipgate_jonny-000000e5”, “hangupcall”) in new stack
[Aug 4 10:00:37] VERBOSE[4347] pbx.c: – Executing [s@macro-hangupcall:1] GotoIf(“SIP/sipgate_jonny-000000e5”, “1?noautomon”) in new stack

The line that interests me most is "Got SIP response 420 “Option Disabled”. So, I found this bug report

issues.asterisk.org/view.php?id=17005

And this is where I get lost. From what I can tell Asterisk 1.6.2.13 includes this fix. Is this bug the same problem I am encountering? Also I have no session-timer settings in any of the configuration files as far as I can tell.

Any guidance would be appreciated. Thanks. Kevin

Alltough You’re thinking, that You’ve got no session-timers settings in Your conf it seems to be related to this according to the default settings for this feature.

You may check the actual settings with asterisk -rx “sip show settings”. Session-Timers settings will appear in the section Global Signalling Settings: If the param Session Timers: is something other than Refuse add the following line to Your sip.conf in the [general]-chapter:

and the problem should disappear after a sip reload.

I agree with abw1oim on the cause. His solution will work.

You may also check if there is a firewall that has closed port talking to port 5060 since call establishement, and reparameter so the session timer will be passed through, keeping the protection for lost sessions

Thanks for the replies. I examined the sip settings and it was set to “accept”. So I amended the config file and did a sip reload. The sip settings now show “refuse”. I shall see if this sorts it. I am more confident in my firewall skills so lean more towards an Asterisk issue, which hopefully is now resolved.

Thanks again.