similar to: Asterisk 1.6 and windows RTC

Hi, I'm using Asterisk 13.x and have defined a pjsip TCP IPv6 transport: [transport-tcp-ipv6] type=transport protocol=tcp bind=[2001:1234:5678:abcd::2]:5060 I also have an IPv4 version of that: [transport-tcp-ipv4] type=transport protocol=tcp bind=10.75.22.8:5060 I've then configured an endpoint to use it: [outgoing] type = endpoint context = default dtmf_mode = none disallow = all

Hi All, This may be an off topic but I'm not sure who else would know the answer. I'm playing around with Asterisk and Polycom phones. I see Polycom supports quite a few codec. The usual ones and these: G722 Siren14.24kbps Siren22.32kbps Siren14.32kbps Siren22.48kbps Siren14.48kbps Siren22.64kbps G7221.16kbps

I am not sure what you mean by the ful SIP signaling? Here is the trace for the sip.conf which works successfully. Below that, I will include the trace for the pjsip.conf which it seems Vitelity isn't accepting the ACK in response to the OK ---- SIP --- <--- Transmitting SIP request (1004 bytes) to UDP:64.2.142.93:5060 ---> INVITE sip:8005555555 at 64.2.142.93 SIP/2.0 Via: SIP/2.0/UDP

I've got a fresh (1 day old) svn trunk release SVN-trunk-r225360 of Asterisk and I'm trying to get it to accept a SIREN14 call from Polycom's softphone. Having trouble with SDP negotiation, I want to only allow SIREN14 and nothing else. I also want to record and playback files, any tips on what the Record function parameters should be? In sip.conf I have: disallow=all

Hi, I am facing an issue while testing the video mail service of Asterisk. I have two different setup on one setup client being used is Mercuro while on the other client is Android based. On the Mercuro setup video mail is stored and retrieved properly while with Android based setup video?mail is not stored (audio is through). Both the client?use H.264 codec with following sdp information:

Sorry these haven't been regularly scheduled :) --------- <p><p>Ogg Traffic for Wednesday, December 12, 2001 Jack Moffitt <[1]jack@xiph.org> _________________________________________________________________ Table of Contents 1. [2]Support The Ogg Project 2. [3]Status Updates 2.1. [4]Monty 2.2. [5]Stan Seibert 2.3. [6]Jack

Hello List; I am facing a trouble with a sip trunk on asterisk 1.4 and asterisk 1.8 and I am getting the following debug, can someone advise me about the solution: <--- SIP read from Provider_IP_Address:5083 --->INVITE sip:22021782 at Asterisk_IP_Address:5060 SIP/2.0?Via: SIP/2.0/UDP Provider_IP_Address:5083;branch=z9hG4bKn1va9h109091cms8h5a0.1?From: "1828444" <sip:1828444 at

Hi, I've a problem configuring my Asterisk. What I try to reach is to interconnect a Tandberg Visioconference (SIP) world with my Asterisk (SIP) with 1 constraint I can't change : "every RTP flow needs to pass THROUGH Asterisk, and are NOT nated" What I observe : - a call made from a SIP Phone registred in Asterisk to Tandberg works (voice and video bidirectionnal) - a call

On Thu, Feb 20, 2020 at 9:38 PM Patrick Wakano <pwakano at gmail.com> wrote: > Hello list, > Hope you are all doing well! > > I am facing a problem when compiling Asterisk 16.8.0 in a CentOS 6 box and > I wonder if someone can put some light on it. > Log history short, install_prereq fails to install the packages (not sure > how important they actually are....):

Hi all. Just as a quote note, can I thank everyone on this list. I find my self finding pretty much every answer I am looking for on here. And a big thanks to all thoughs helping us out. Mass Respect :) Ok, I'm using a SIP provider (SipGate UK) to do my international dialing etc, working great from extension 8 on phones. However some more friends/contacts have started using SipGate also, and

Hello list, Hope you are all doing well! I am facing a problem when compiling Asterisk 16.8.0 in a CentOS 6 box and I wonder if someone can put some light on it. Log history short, install_prereq fails to install the packages (not sure how important they actually are....): speexdsp-devel, gmime-devel, uriparser-devel, iksemel-devel, uw-imap-devel, hoard Then, I am running the following commands

When I try call to net2pohe sip service in my debug I look next: ---------------------------------------------------- We're at 192.0.0.0 port 27916 Answering with preferred capability 1 Answering with preferred capability 2 Answering with preferred capability 256 Answering with capability 4 Answering with capability 8 Answering with capability 16 Answering with capability 32 Answering with

When I activate the equalizer in WinAmp, QCD, and Sonic Foundry SIREN Jukebox, I get a funny "washed out" sound to it. Vorbis sounds fine without it, but I miss the extra bass and what-not. I was wondering why this happens? --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to

> A Siren codec is not currently available and the one for 12 will not > work. I have no timeframe for when this might change. So the only option is to build one from the Polycom sources? I'm already doing this for Siren14 (I forget why).

Anyone have a status update on when Sonic Foundry plans on supporting RC2 ogg files? A while back on this list someone said they had a good contact at SF and was supposed to chekc up on it. I have contacted Ahead Software about including Vorbis support with Nero and they asked that a developer contact them about including it. Any chance a developer can contact them about support for Vorbis

On Wed, Oct 29, 2014 at 7:10 PM, sean darcy <seandarcy2 at gmail.com> wrote: > On 10/29/2014 08:06 PM, Matthew Jordan wrote: > >> On Wed, Oct 29, 2014 at 5:16 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> >>> Can we expect a SILK codec for 13 ? Or does the one for 12 work for 13? >>> >>> >> codec_silk for Asterisk 12 will most

Hi Kevin! Thanks very much for your reply! Much appreciated! So I just have a remaining question from this, if the with-ssl is not mandatory to have the encryption support, what is it actually used for? Maybe it is some old flag which is not needed anymore and so can be ignored for now and possibly removed from the configure/makefile stuff for future releases? Kind regards, Patrick Wakano On

Hi all, I've included copies of the relavent files below. I'm having a bit of a problem with Samba in that when I try to access files on its shared resources, I can browse to the shares fine, but the moment I try to open a file, the client's file browser hangs. I've verified that this is a problem on both my Mac (OS X 10.3.4), my Windows XP SP2 and WIndows XP SP1 machines. I'm

> All patches need to go into JIRA with a license agreement to be > accepted. Understood, but I was using it as an illustration. Note, however, that, from a legal perspective, a patch such as this has no protectable IP (you can't copyright the only way of doing something) and the GNU projects have a formal rule that sufficiently-small patches need no assignments for that reason, which

I'm connecting a Polycom SoundStation IP 7000 and trying to use siren14. I downloaded the codecs and now it will properly transcode to connect to other phones and play any files that are in .wav format. But when it tries to play any files with .siren14 extensions, I get complete noise coming out. Here's the negotiated SDP: v=0 o=root 1668560220 1668560220 IN IP4 207.10.184.50 s=Asterisk