Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN

You were right. I had non-default rtp ports open in iptables. Edited rtp.conf et voila. Everything seems to be working. Thanks so much for your patience and guidance! Have a lovely eening. -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN

Chirag Desai wrote: So I see: EXTERNAL_SNOM_IP -> EXTERNAL_ASTERISK_IP (UDP, length 218, src: 60798, dst 11128) EXTERNAL_ASTERISK_IP -> INTERNAL_SNOM_IP (UDP, length 218, src: 11128 dst 60478 So i see udp from the phone, but there's no audio. If "rtp set debug on" shows no packets

Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN

So I see: EXTERNAL_SNOM_IP -> EXTERNAL_ASTERISK_IP (UDP, length 218, src: 60798, dst 11128) EXTERNAL_ASTERISK_IP -> INTERNAL_SNOM_IP (UDP, length 218, src: 11128 dst 60478 So i see udp from the phone, but there's no audio. I do also see some packets :: EXTERNAL_ASTERISK_IP ->

Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN

Chirag Desai wrote: In the PCAP I can see asterisk sending UDP packets to my local IP 192.168.0.5 If you don't see anything arriving from the remote side and we've told them the right IP address and ICE is not actually negotiated... then that leans more towards something remote unless

Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN

In the PCAP I can see asterisk sending UDP packets to my local IP 192.168.0.5 It's funny, when I switch to TCP on 5060 audio seems to work fine. The moment I go to 5063 on TLS everything goes a bit awry. Any further input is greatly appreciated. --

Re: [asterisk-users] Asterisk now available with bundled pjproject!

On Mon, Mar 7, 2016 at 2:53 PM, Jean-Denis Girard wrote: > Hi, > > Le 07/03/2016 09:28, George Joseph a écrit : > > PLEASE TRY THIS!! I'd love some feedback BEFORE 13.8.0 is released. > > I have tried GIT-master-ee5a944M on my Fedora 23 test server, and got: > > [pjproject]

[asterisk-users] Tapping into an existing audio stream rather than starting a new mp3Player?

>From what I can tell from the Wiki page at https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_MP3Player, if someone dials in and starts playing a stream, mp3player will load up the URL and inject it into the current call. But what about if 20 or 30 people call in, and it's firing

Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN

Chirag Desai wrote: I'm dialling from the snom and every few calls asterisk sends media to the phones external IP and it works! And then now and again it sends the media to the phones internal IP and I hear nothing. I'm really at a loss. In the non-working case check the IP address in the

Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN

I'm dialling from the snom and every few calls asterisk sends media to the phones external IP and it works! And then now and again it sends the media to the phones internal IP and I hear nothing. I'm really at a loss. -- _ --

Re: [asterisk-users] Asterisk now available with bundled pjproject!

Hi, Le 07/03/2016 09:28, George Joseph a écrit : > PLEASE TRY THIS!! I'd love some feedback BEFORE 13.8.0 is released. I have tried GIT-master-ee5a944M on my Fedora 23 test server, and got: [pjproject] Unpacking /tmp/pjproject-2.4.5.tar.bz2 [pjproject] Applying patches and custom files

Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN

Chirag Desai wrote: Joshua Colp wrote: Have you done a packet capture to see if the RTP from the remote device is hitting the machine to narrow things down? Nope. When I run with RTP encryption on it seems that rewrite_contact does not work in PJSIP. When I turn

Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN

> Joshua Colp wrote: >> >> Have you done a packet capture to see if the RTP from the remote device >> is hitting the machine to narrow things down? >> >> >> Nope. When I run with RTP encryption on it seems that rewrite_contact does not work in PJSIP. When I turn off RTP some calls get media, some

Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN

> Joshua Colp wrote: > > There should be nothing different, except for how you configure things. > What is the full PJSIP configuration? What is the environment where > Asterisk is running? Is ICE actually in use on the other side? What is > the full SIP trace? > The full configuration is here:

[asterisk-users] Asterisk now available with bundled pjproject!

The current Asterisk 13 and master git branches have a new feature that will be included in 13.8.0: The ability to compile and run Asterisk with a bundled version of pjproject. ​​ Why would you want to do this? Several reasons: - Predictability: When built with the ​bundled

Re: [asterisk-users] Pass variable to voicemail script

>If you are talking about the 'externnotify' parameter in voicemail.conf, the variables are passed simply as @ARGV. I'm referring to the mailcmd= setting in voicemail.conf. Asterisk runs this when emailing a voicemail (with attachment) --

Re: [asterisk-users] Pass variable to voicemail script

Hello; If you are talking about the 'externnotify' parameter in voicemail.conf, the variables are passed simply as @ARGV. Regards; John V. Tech Support Tech Support VoIP Business Solutions 240-215-3479 x325 supp...@voipbusiness.us From:

Re: [asterisk-users] Pass variable to voicemail script

March 6 2016 1:06 AM, "Michelle Dupuis" wrote: > I have a custom voicemail script which reformats and forwards the attached > voicemail wav file to > the recipient. > > I would like to make use of a channel variable in my script; is there a way > to pass a channel > variable to

Re: [asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN

Chirag Desai wrote: I have two accounts on Asterisk 13. One uses chan_sip and the other pjsip. In my snom 760 the setup for these two accounts is identical. When I call echo test from the account using chan_sip audio comes through fine. When I call echo test from the account using pjsip there