PAP2T, sip.voipwelcome, local call.Originally Posted by voxabox
Senior Member
Re: 30 minute cutoff
PAP2T, sip.voipwelcome, local call.
Member
Re: 30 minute cutoff
Session timers are enabled only on incoming calls. Please test incoming calls to be sure of the cutoff.
I use Asterisk 1.4, and I manually backported session timer support from the 1.6 branch (digium bug id 10665). If people are interested, I can provide the compiled chan_sip.so and also the merged source file. Overwriting chan_sip.so in /usr/lib/asterisk/modules/ is enough (a restart of asterisk or the system is not required).
Senior Member
Re: 30 minute cutoff
Local call does not count. This is voip...
Is it sip to sip or sip to pots?
my wish list:
- Intelligent Call Forwarding that detects the incoming call originated from the "forwarded" phone and rings the original destination instead
Call History that makes use of CallerID/Custom CallerID+Location. Call History only shows Custom CallerId+Loc. No CNAM look up; Albeitly,it's a step in the right direction!
- Scheduled sim. ring with a twist (see wish #1)
Senior Member
Re: 30 minute cutoff
Do we really know for sure anymore ? People calling me assume it's a POTS number.......
I did not dial a sip address, so do we assume it went to POTS ?
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