[asterisk-users] Calls are dropped after 15 minutes

What version of asterisk are you on? Marlon Araujo > On Aug 10, 2016, at 13:00, asterisk-users-requ...@lists.digium.com wrote: > > Calls are dropped after 15 minutes -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] Replacement for phpagi?

Anyone know a good replacement for phpagi? Unfortunately development stalled long ago and it has not been updated. What is the best solution for AMI and AGI on PHP? Thanks. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)9116-91161 --

[asterisk-users] Realtime warnings for database structure

I keep getting messages like these in the cli: [Aug 10 12:20:17] WARNING[23411]: res_config_mysql.c:1162 require_mysql: Realtime table general@ps_contacts: column 'qualify_timeout' cannot be type 'int(10)' (need char) [Aug 10 12:20:17] WARNING[23411]: res_config_mysql.c:1246 require_mysql:

Re: [asterisk-users] Replacement for phpagi?

El 10/08/16 a las 12:06, Carlos Chavez escribió: Anyone know a good replacement for phpagi? Unfortunately development stalled long ago and it has not been updated. What is the best solution for AMI and AGI on PHP? Thanks. In the case of AMI, you could use the AMI client from the

Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

My suggestion is to verify and debug against Asterisk 13 first, and then you can try backing down versions, rather than reverse. WebRTC is a rapidly moving target, and has required ongoing changes that may not have made it into older and feature frozen versions of Asterisk. Matthew Fredrickson

Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

I don't see an ice-ufrag or ice-pwd line in the response from Asterisk, correlating with your suspicion that there is no ICE. Are you sure that the stun server you're using (the google one) still works? I haven't tried that server in a while, but I distantly seem to recall that maybe they shut

[asterisk-users] Original Callerid on transfer in asterisk 13

Hi Is there any configuration change in asterisk 13.9.1 to show original callerid on a transfer In asterisk 11.21 it works as expected Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to

Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

Hello thank you for your answer. I don't understand how there are many tutorials and examples on the web where every time the outcome is a working setup. Very strange I feel now after my personal experience with Asterisk 11 and webRTC. You also say Asterisk 13. How about Asterisk 12 then ??

Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

For WebRTC, I recommend you to use Asterisk 13+. Have a nice day. Ludovic Gasc (GMLudo) http://www.gmludo.eu/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live

Re: [asterisk-users] Original Callerid on transfer in asterisk 13

How are you attempting to view the original CallerId? Matthew Fredrickson On Wed, Aug 10, 2016 at 2:59 PM, Israel Gottlieb wrote: > Hi > Is there any configuration change in asterisk 13.9.1 to show original > callerid on a transfer > In asterisk 11.21 it works as expected > >

Re: [asterisk-users] Asterisk & Vitelity Invite issues

On 8/9/16 12:40 PM, Matt Fredrickson wrote: > On Mon, Aug 8, 2016 at 9:25 AM, Tammy Firefly wrote: >> Hi All, >> >> We have asterisk 11.23 running sip to vitelity and from there IAX trunks >> split off to where they need to go. We are having a problem getting >>

Re: [asterisk-users] Original Callerid on transfer in asterisk 13

In 11 setting trustrpid sendrpid is enough that phone getting the tranfered call shows the name and number of the caller and not the tranferer In 13 the same shows the transferrs info בתאריך 11 באוג׳ 2016 00:21,‏ "Matt Fredrickson" כתב: > How are you attempting to view the

Re: [asterisk-users] chan_pjsip ignoring endpoint device state (qualify) on dial

On 2016-08-09 10:06, Faheem Muhammad wrote: Jacek, This might be a bug or configuration issue, but you need to understand the SIP Session Timers. With Session Timers you can control the round trip time and Call Setup time of SIP Requests. I don't think you really mean SIP Session Timers

Re: [asterisk-users] chan_pjsip ignoring endpoint device state (qualify) on dial

On 2016-08-09 10:06, Faheem Muhammad wrote: trip time and Call Setup time of SIP Requests. In case of GSM Network with high delay you need to set the T1 timer a higher value like 1000ms (500 ms default). Similarly you can reduce the Call setup time by configuring 'T2' upto you choice as per you

Re: [asterisk-users] chan_pjsip ignoring endpoint device state (qualify) on dial

On 2016-08-10 11:53, Joshua Colp wrote: Jacek Konieczny wrote: On 2016-08-09 10:06, Faheem Muhammad wrote: trip time and Call Setup time of SIP Requests. In case of GSM Network with high delay you need to set the T1 timer a higher value like 1000ms (500 ms default). Similarly you can reduce

Re: [asterisk-users] Asterisk & Vitelity Invite issues

Wait a second, I thought in your original email that you said that Asterisk was generating reinvites. It sounds now like you're saying that the remote side is initiating reinvites instead. My understanding is that the canreinvite/directmedia option only influences Asterisk's behavior with

Re: [asterisk-users] chan_pjsip ignoring endpoint device state (qualify) on dial

Jacek Konieczny wrote: On 2016-08-09 10:06, Faheem Muhammad wrote: trip time and Call Setup time of SIP Requests. In case of GSM Network with high delay you need to set the T1 timer a higher value like 1000ms (500 ms default). Similarly you can reduce the Call setup time by configuring 'T2'

Re: [asterisk-users] Asterisk 11.23.0 on CentOS6 : how to get ICE support ?

On 10-08-16 08:52, Ludovic Gasc wrote: For WebRTC, I recommend you to use Asterisk 13+. Have a nice day. Ludovic Gasc (GMLudo) http://www.gmludo.eu/ Hello then why is there an option in sip.conf and rtp.conf " icesupport=yes" ?? This is no answer to my question. So again : what am I