[asterisk-users] Sippy Cup

2014-11-14 Thread Murthy Gandikota
If you have used sippy_cup by Ben Klang and Will Drexler please comment. Please note, I know there is a Sipp users mailing list. I am trying to catch the attention of the developers and users who work with asterisk as well. I have a scenario where I expect field0 and field1 to be injected to the x

Re: [asterisk-users] E1 - Cisco - Asterisk and vice verso

2014-11-14 Thread Bruce Ferrell
On 11/14/2014 02:57 PM, Kevin Larsen wrote: > > > Hi, > > > > > > Change my Dynastar E1 gateway to Cisco with E1 module, but can't make > > > easiest dialplan. All my routing i made on asterisk, so i need that cisco > > > all calls from E1 send via sip to Asterisk and all calls came from > > > Ast

Re: [asterisk-users] Como unir webrtc con asterisk???

2014-11-14 Thread symack
Hahahah Rusty! Love it​. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users

Re: [asterisk-users] E1 - Cisco - Asterisk and vice verso

2014-11-14 Thread Kevin Larsen
> > Hi, > > > > Change my Dynastar E1 gateway to Cisco with E1 module, but can't make > > easiest dialplan. All my routing i made on asterisk, so i need that cisco > > all calls from E1 send via sip to Asterisk and all calls came from Asterisk > > by sip send to E1. From E1 to Asterisk already wo

Re: [asterisk-users] E1 - Cisco - Asterisk and vice verso

2014-11-14 Thread Rusty Newton
On Wed, Nov 12, 2014 at 10:02 AM, Dmitriy Sirant wrote: > Hi, > > Change my Dynastar E1 gateway to Cisco with E1 module, but can't make > easiest dialplan. All my routing i made on asterisk, so i need that cisco > all calls from E1 send via sip to Asterisk and all calls came from Asterisk > by sip

Re: [asterisk-users] Erratic calls through NAT-ed server

2014-11-14 Thread Rusty Newton
On Thu, Nov 13, 2014 at 8:23 AM, Norman Laidla wrote: > Morning, > > We recently pushed our Asterisk video bridge into a DMZ and since then, > local calls have been unreliable to say the least. While offsite calls work > nicely, calls on our internal server usually fail to ring the far end. Two >

Re: [asterisk-users] Como unir webrtc con asterisk???

2014-11-14 Thread Rusty Newton
2014-11-12 16:24 GMT-06:00 Dario Estupinan : > tengo la siguiente pagina pero no se como seguir despues del punto 22 > > http://highsecurity.blogspot.com/2012/12/webrtc-and-asterisk-11-using-sipml5.html > > gracias! You haven't described your problem and I'm relying mostly on Google translate, so

[asterisk-users] Asterisk 13 confbridge recordings not working

2014-11-14 Thread Bill Barron
We upgraded from asterisk 11 to asterisk 13. Recordings were working fine in 11 but nothing is being written on 13. Here is the dialplan segment same => n,ExecIF($["${TL_PHONE_CALL_RECORD}"="TRUE"]?SET(CONFBRIDGE(bridge,record_conference)=yes)) same => n,ExecIF($["${TL_PHONE_CALL_REC

[asterisk-users] SLA (Shared Line Appearance) and realtime

2014-11-14 Thread Leandro Dardini
Hello, do you know if it is possible to define the SLA configuration in the database for realtime usage with asterisk? Leandro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us fo