30 minute cutoff

[dswartz]

Well, there may be a bug where it is not always getting set. On 1.4 I had the same experience as you (sometimes yes and sometimes no.) With 1.6 I never do.

[scott2020]

That is an option I suppose. I can't believe no one else has this problem, especially since the talk was the 30 minute timers were taken away by VOIPo. There must me someone else here running Asterisk 1.4. Maybe they just don't talk as long as I do!

[fisamo]

I run Asterisk 1.4, but calls on the line are generally short. For now, it's mainly my 'hobby' line, because I got in on the Beta and have good pricing on the line. However, as my girls get older, the line will definitely start getting heavy use.

In the meantime, I'll have to think of a good test number to call that would keep the call open without me having to do anything. Any ideas?

Edit: (It may be obvious, but I don't want to burn cell minutes or tie up my regular house line, and I'd like the call to be off-voipo network...)

[bubbanc]

Call Dell tech support, if they come on quickly tell them you have to find your service tag number and to give you a minute or perhaps an IRS hot-line.

[scott2020]

I know some other VOIP providers that would keep me on hold until I was old and grey, but I won't mention them here! Dell seemed to be good for that when I called them too.

Anyway, my wife talked to her mom for 55 minutes without getting cut off, so it is very random. Perhaps it is something from an upstream provider or something? Depending on the call route, maybe one of VOIPo's providers does something with the session timers.

Somewhere in this forum I think I remember a patch someone created, a backport from Asterisk 1.6 to 1.4 to support the session timers. I'll dig for that.

[zcnkac]

I have attached a zip file that contains the modified chan_sip.c and the compiled chan_sip.so, which supports session timers. These are for asterisk version 1.4.21.2.

!!! Backup the files /usr/lib/asterisk/modules/chan_sip.so and /usr/src/asterisk/channels/chan_sip.c !!!

Since there is a 97 KB limitation on attachments I have split chan_sip.c into two volumes. To compile it, overwrite /usr/src/asterisk/channels/chan_sip.c with the one from the archive and do a 'make all' in the /usr/src/asterisk directory. This will create a chan_sip.so in the /usr/src/asterisk/channels directory. Next do an 'amportal stop'. Overwrite /usr/lib/asterisk/modules/chan_sip.so with /usr/src/asterisk/channels/chan_sip.so. Do a 'amportal start' ...

For configuring this, disable timers globally by adding the line
session-timers=refuse
to /etc/asterisk/sip_general_custom.conf. Enable timers for voipo, by adding
session-timers=accept (or session-timers=originate) and session-refresher=uac
to PEER Details of the voipo trunk in FreePBX.

I have been using this for the last few weeks without any issue. YMMV. Use at your own risk.

Somewhere in this forum I think I remember a patch someone created, a backport from Asterisk 1.6 to 1.4 to support the session timers. I'll dig for that.

[scott2020]

Great, thanks! I'll have to give that a try.

PS

Seems like 97k is pretty small for the attachment limit.