I'm very happy to report that I made a 45 minute call today !! No more 30 minute cutoff
Is Voipo a great company or what !?!?
Senior Member
Re: 30 minute cutoff
I'm very happy to report that I made a 45 minute call today !! No more 30 minute cutoff
Is Voipo a great company or what !?!?
Senior Member
Re: 30 minute cutoff
Good news, but
Need to further elaborate on equipment, call, server, etc
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
Yes they are. I think the service is great.Originally Posted by Montano
I'm very happy to report that I made a 45 minute call today !! No more 30 minute cutoff
Is Voipo a great company or what !?!?
I can not wait to see what happens next. I have had the service less than a month and despite a few minor problems I am really happy with the product. I can not wait to see what happens next. Even in my short time here I have seen things continue to change for the better.
Thank you Tim and all the staff for your hard work and dedication.
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
I must disagree. All the calls I got cut off on, were initiated by me.
Member
Re: 30 minute cutoff
Outgoing calls cutting off might have been because of something else. I have since switched to central01. Before that, outgoing calls would drop sometimes because asterisk could not "re-authenticate". It might also be because I was using codeblue in fromdomain.
I can definitely say that RFC 4028 session-timers are being enabled only for incoming calls.
Senior Member
Re: 30 minute cutoff
Hmmm, that's weird. I could swear I had a number of calls dropped at the 30 minute mark when I was calling other people (that was on asterisk 1.4 before I moved to 1.6).
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
Guys, read the rfc again
Sst is to prevent zombies
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)
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