Since Voipo is now shipping PAP2T's because of shortage of GS devices, you would think they would have disabled the timers.
Printable View
Since Voipo is now shipping PAP2T's because of shortage of GS devices, you would think they would have disabled the timers.
Tim said it was taken care of. Lets just take his word for it. ;)
I received one of the recently shipped PAP2Ts. I have received incoming calls longer than 30 minutes.
No problem, I must have missed where he said specifically the 30 minute session timers were removed. When I was using Asterisk sometimes I would get cut off at 30 and sometimes not, so I was wondering if it was more coincidence or if the timers were indeed taken off.Quote:
Originally Posted by Xponder1
I moved everything back over to my Asterisk box, since people have had good luck with it. I am running Asterisk 1.4, and I was talking with my brother this morning. At exactly the 30 minute mark, my call got cut off. I got a fast busy on my end. So I don't know what people are doing to get around it or whatever, but I got cut off at exactly 30 minutes. This makes me think the session timers are on somewhere and that is why I wanted an official word whether they were enabled, disabled for some, or disabled across the board. My other trunks always work over 30 minutes so I'm not sure if it is on my end or not.
Edit: I called him back, and talked to him for 35 minutes without getting cut off. So I have no idea what is going on.
Scott
Was the cutoff call initiated by you or him? I remember speculation that which end originated it mattered as far as the timers being on or not. I got irritated enough several months ago to roll out asterisk 1.6.0 and it's been stable for me.
I called out to him both times, which makes it even more frustrating. I am on the phone with my dad, who called me, at the 32 minute mark and no cutoff so far. Seems very random.
Sorry for your pain :( All I can suggest is switching to 1.6 :(
I patched 1.4.21.2's chan_sip with session timers. I have been using it for the last few weeks without problems. I can post the modified chan_sip.c and chan_sip.so ...
I don't get it. The timers are either on or off for everyone I would think. Sometimes I get cut at 30 minutes and sometimes not. I am lost.
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.
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!
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. :eek:
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...)
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.
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.
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.
Quote:
Originally Posted by scott2020
Great, thanks! I'll have to give that a try.
PS
Seems like 97k is pretty small for the attachment limit.