Since Voipo is now shipping PAP2T's because of shortage of GS devices, you would think they would have disabled the timers.
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.
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
Last edited by scott2020; 02-15-2009 at 11:52 AM. Reason: Additional info
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.
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