Re: [asterisk-users] loosing audio from one end after 5 min.

1) Does it happen every time at the 5 minute work? 2) Have you done a dump on the client side to see if the NAT device is dropping the packets? 3) Is the phone behind a load balance internet connection and is the RTP port changing? On Thu, Aug 11, 2016 at 5:33 PM, Jonas Christoffersen

[asterisk-users] loosing audio from one end after 5 min.

Hi all, Just installed Asterisk 13 on CentOS 7 and have run into a problem. The Scenario is this: Asterisk is on the internet the Phone, a D40, is behind NAT So someone calls the number and Asterisk routes the call to the D40 Everything works fine and the call is established, but then after 5

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

On 11-08-16 18:03, Matt Fredrickson wrote: On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens wrote: My main reason not to upgrade to Ast 13 is because I'm afraid of losing functionality as there are certain functions deprecated/replaced. This can also cause headache :-)

Re: [asterisk-users] PJSIP not detected

On Thu, Aug 11, 2016 at 1:08 PM, Saint Michael wrote: > I installed PJSIP from the project > git clone https://github.com/asterisk/pjproject pjproject > cd pjproject > make uninstall & make distclean > ./configure --libdir=/usr/lib64 --prefix=/ --enable-shared --disable-sound

Re: [asterisk-users] PJSIP not detected

On Thu, Aug 11, 2016 at 2:08 PM, Saint Michael wrote: > I installed PJSIP from the project > git clone https://github.com/asterisk/pjproject pjproject > cd pjproject > make uninstall & make distclean > ./configure --libdir=/usr/lib64 --prefix=/ --enable-shared --disable-sound >

Re: [asterisk-users] PJSIP not detected

Why don't you use the bundle option in Asterisk compilation ? ./configure --with-pjproject-bundled Regards, Marcelo H. Terres IM: mhter...@jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On

[asterisk-users] PJSIP not detected

I installed PJSIP from the project git clone https://github.com/asterisk/pjproject pjproject cd pjproject make uninstall & make distclean ./configure --libdir=/usr/lib64 --prefix=/ --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr --with-external-srtp make

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

On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens wrote: > My main reason not to upgrade to Ast 13 is because I'm afraid of losing > functionality as there are certain functions deprecated/replaced. This can > also cause headache :-) What in particular? Any longer,

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

On Thu, Aug 11, 2016 at 9:04 AM, Tammy Firefly wrote: > my bad, both sides are generating re-invites. Vitelity ignores any > inbound invites to continue call flow. to keep the call going our pbx > has to deal with their re-invites otherwise the call terminates at 30 >

Re: [asterisk-users] Replacement for phpagi?

Hi I would recommend PAMI - its object oriented and well structured On 10 August 2016 at 19:49, Alex Villací­s Lasso wrote: > El 10/08/16 a las 12:06, Carlos Chavez escribió: > >> Anyone know a good replacement for phpagi? Unfortunately development >> stalled long

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

On Thu, Aug 11, 2016 at 9:40 AM, Jonas Kellens wrote: > My main reason not to upgrade to Ast 13 is because I'm afraid of losing > functionality as there are certain functions deprecated/replaced. This can > also cause headache :-) > > I will do so if there is no other

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

I'm genuinely fascinated why you are insisting on using a version of Asterisk almost 3 years old, for which EOL support ended last year. Is there any particular reason you cannot or will not use the current version as others have suggested? Also, I see you are using Doubango and WebRTC, but in

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

Hello Using Asterisk 12.8.2. I now have the "via ICE" messages in the RTP debug (see below). If you look in the SIP debug (see below), you also now see the "ice-ufrag" and "ice-pwd" in the 200 OK SIP-message from Asterisk to the webRTC client. But still no audio ! None at all ! In both

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

my bad, both sides are generating re-invites. Vitelity ignores any inbound invites to continue call flow. to keep the call going our pbx has to deal with their re-invites otherwise the call terminates at 30 minutes on the dot. Our side is ignoring the inbound invites from vitelity and that

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

My main reason not to upgrade to Ast 13 is because I'm afraid of losing functionality as there are certain functions deprecated/replaced. This can also cause headache :-) I will do so if there is no other option. But still, I don't see why Ast 13 would differ so much in this case ? If ICE