Re: [asterisk-users] PJSIP Stun/ICE

2016-01-29 Thread George Joseph
On Fri, Jan 29, 2016 at 5:11 AM, Bryant Zimmerman wrote: > George > > Reloading transports is one critical part and it sounds like you are > making headway on that. I have yet to be able to get transports to load > from a real-time table using sorcery.conf > ​Patch up for this part which allows

Re: [asterisk-users] PJSIP RTP Timeout - Calls not ending

2016-01-29 Thread Richard Mudgett
On Fri, Jan 29, 2016 at 3:23 PM, John Roth wrote: > I’m running FreePBX 13.0.49 (Asterisk 13.5.0) with PJSIP and running into > a problem when my endpoint disconnects form the network while the call is > in progress. I was able to set RTP timeouts on the endpoint so that it > recognizes loss of c

[asterisk-users] PJSIP RTP Timeout - Calls not ending

2016-01-29 Thread John Roth
I'm running FreePBX 13.0.49 (Asterisk 13.5.0) with PJSIP and running into a problem when my endpoint disconnects form the network while the call is in progress. I was able to set RTP timeouts on the endpoint so that it recognizes loss of connectivity and hangs up, but the call on the Asterisk se

Re: [asterisk-users] 11.21.0 : echo woes : can't install canceller (sean darcy)

2016-01-29 Thread Mc GRATH Ricardo
Hi Sean Darcy Question about "the remote party always hears an echo on it's side", strange because eco suppression circuit is for local side. Mc GRATH Ricardo -- _ -- Bandwidth and Colocation Provided by http://www.api-digita

Re: [asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API

2016-01-29 Thread Matthew Jordan
On Fri, Jan 29, 2016 at 6:15 AM, Bryant Zimmerman wrote: > Sonny > > We use a real-time database for adding pjsip users. If you want to do it > from the pjsip.conf you would have to write to the file from a script of > some sort and then trigger a reload. There is a real-time implementation > f

Re: [asterisk-users] Asterisk 13.6.0: Is there a way to create PJSIP users and dialplans programmatically using API

2016-01-29 Thread Bryant Zimmerman
Sonny We use a real-time database for adding pjsip users. If you want to do it from the pjsip.conf you would have to write to the file from a script of some sort and then trigger a reload. There is a real-time implementation for the extensions.conf as well. I personally use scripts for mos

Re: [asterisk-users] PJSIP Stun/ICE

2016-01-29 Thread Bryant Zimmerman
George Reloading transports is one critical part and it sounds like you are making headway on that. I have yet to be able to get transports to load from a real-time table using sorcery.conf If I would get the transports pulling from real-time as the (documentation says is possible but I have

Re: [asterisk-users] asterisk 13 mixmonitor - random missing syllables

2016-01-29 Thread Brian ::
12 calls isn't under any type of load. Someone with better understanding of Asterisk internals may chime in here. Could it be vmware timing? Is timing critical when using mixmonitor? I've seen > 100 concurrent calls being recorded wtihout issue. On Fri, Jan 29, 2016 at 10:39 AM, Marek Červenk

Re: [asterisk-users] asterisk 13 mixmonitor - random missing syllables

2016-01-29 Thread Marek Červenka
Dne 28.1.2016 v 13:37 Brian :: napsal(a): when you say load - how many concurrent calls? Is there transcoding happening? sip / PRIs ? what load? 12 concurrent calls no transcoding SIP under 1.5 with 4x 1Ghz vcpus (its vmware VPS) On Thu, Jan 28, 2016 at 9:57 AM, Marek Červenka