similar to: video support over iax

I'm having a problem with IAX running through an intermediate asterisk box. Perhaps a small diagram will explain the situation better: *A ------- [cloud (public internet)] ------- *B --------[cloud (private network)]----------- *C Asterisk server's A, B, and C, are all connected together with IAX All asterisk servers are 1.6.0.6 Server A and B are geographically close, but connected over

I am experiencing choppy sound when I bridge from a SIP peer to an IAX peer. I am running Asterisk 1.4.13 on a 2.6.22.9 kernel (Fedora). I am experiencing choppy sound from the SIP peer to the IAX peer but not vice-versa. I know that this is not a bandwidth issue because I don't have choppy sound (with the same codec) when bridging IAX->IAX peers or SIP->SIP peers. My timing source is

We have been, unsuccessful, trying to implement inter-asterisk communications using IAX. I was wondering if anyone is having similar problems, or would share information about their succes/ Our environment consists of two servers with public IP addresses and DNS entries that resolve. (for the example we use "domain.net") Based on the information available to us, we have tried the

Hi all, ------ I tried to use * over satellite, but all my effort did not succeed. The Asterisk is behind the VSAT and is resposibel for alle the SIP clients in a field location. The clients are notebooks and PDA's running SJPhoen for Windows and PocketPC. Unfortunately I could not find any Linux Client wich worked satisfying. SJ LAbs promised a Linux Version at the end of August but they

Hi there, slowly I am starting to think that there is an IAX issue in the Oct.5 CVS snapshot that I use to run Asterisk: I can register server2 to server1, but not vice versa, no matter what I do - the setup of both servers is identical, and I even removed the password for the server1 in server2's iax.conf file. Still if I check with "iax2 show registry" in server1 I see the

Hi, I have 2 asterisk servers link via IAX. and i'm trying to do a video call. if 2 sip users are registered on the same server, the video works fine. but if 1 sip user is on server 1 and sip user 2 on sip server 2. there's no video at all. is it because call from sip server 1 goes to sip server 2 via IAX? hope my question is clear enough, thanks in advanced Ron

Thanks for the feedback ! I know the IP-address of my Asterisk-server. The WAN-interface of my Asterisk-box is set manually (ifcfg-eth1). I have port 4569 forwarded on my NAT/firewall. Strangely I have the same 'notice' when being attached directly to the internet (so no firewall in between). And set my WAN-interface to the IP I get from my ISP or even when obtained by DHCP. Doesn't

Hi All, if i do this setup: |---[ext 100] |--[router/nat gw]--| | |---[ext 101] | [asterisk]--[internet]---| | | |---[ext 200] |--[router/nat gw]--|

Hi, I have setup 2 asterisk talking? a single mysql cluster. I'm also using realtime db. I've setup sip peering between the two asterisk servers. [asterisk-1] insecure=port,invite type=peer host=201.202.203.204 context=from-asterisk-1 [asterisk-2] insecure=port,invite type=peer host=201.202.203.205 context=from-asterisk-2 scenario: ext 100 registers on Asterisk 1 ext 200

<div><FONT size=2>Hello,</FONT></div> <div><FONT size=2>two questions: </FONT></div> <div><FONT size=2></FONT>&nbsp;</div> <div><STRONG><FONT size=2>1: How can I open/enable network connection to B?</FONT></STRONG></div> <div><FONT

Hi I had installed and configured 2 IAX server , users from 1'st server can dial to the second server and vice versa But when I make calls to users in other server , on my client , I get the caller if as asterisk@192.168.20.99 , the same I get when I try reverse , ie I get on my cleint caller id as astersik@192.168.20.32 Please guide me what

I have an application where we receive calls on an inbound PRI. After hours, our Asterisk box dials our answering service on an outbound PRI and then bridges the caller to the answering service. The flow looks like this: (CALLER)INBOUND_PRI --> CONTEXT --> GOSUB(Incoming) --> GOSUB(bridge-to-anssrv) --> DIAL(answering_service) --> OUTBOUND_PRI(service) This has been working

Hi I'm browsing around the internet looking for signs that the IAX client library and app_meetme support video. I stumbled across this post by SteveK on the 27th of Feb 2006. "My company is looking to hire a full-time developer, who will be working about 25-50% of the time on iaxclient; in particular to finally integrate, build, polish and enhance video in iaxclient, add video

Hi, We had 2 asterisk 1.4 connected together in iax, all was fine. One of them was upgraded (server and Asterisk) in 1.6.2.15, the other end is in 1.4.38 When calling to 1.4 to 1.6.2 -remember, it's iax- all is good. But calling from 1.6.2 to 1.4 give a bad audio to calling party (words are cutted, you can't understand the words). On callee party it's still good. We replace

Hi all, I have 2 remote asterisk servers (1.2.5 version) , both of them locally supporting both iax and sip video. I connected the two asterisk via Internet / IAX2 trunk (two way connection) When I dial from one sip video phone on the first server a sip video phone on the second server, the voice is ok; when one of the two phones try to start the video, the call is suddendly ended. Moreover,

I'm working on getting two SIP servers to talk. An Asterisk box and a Zultys MX250 system. So far, things have been working pretty well. I can call from one of my Asterisk-managed phones to an MX250-managed phone and vice versa. However, there are some strange issues. If I call from an MX250 phone to an Asterisk phone, the conversation is ok, but there is a noticeable delay in the voice

HI ALL. Please help. Problem: video calls drop after 15-20 seconds all the time. Use * latest cvs. from sip.conf [1102] type=friend username=1102 host=dynamic callerid=Veo webcam<1102> canreinvite=no disallow=all allow=gsm ;allow=ulaw allow=h261 allow=h263 from iax.conf [peer2] ; 192.168.0.7 type=friend port=4569 auth=md5 secret=second2 context=local host=dynamic qualify=yes trunk=yes

Hello, I am having horrible echo problems when using the FireFly client on both the caller and callee sides of the call. When I use another IAX soft client like IAXcomm or IAXPhone I do not have the same echo problems. Has anyone else experienced this and do you know what might be the problem? Thanks, dj -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi... I have what is, I am sure, a relatively common & straightforward problem (no, NOT that kind of problem!)... I'm trying to hook two asterisk servers together so I can make a "distributed" PBX. Here's the scenario: [MASTER] is in the office. It has unrestricted access to the internet, and a fixed IP address. It has 3 SIP hardphones configured & working, plus a

Hello, Would like to use IAX /IAX2 to transport 30 channels inter Asterisk. I don't have any experience with that, so can someone help ?? How much bw do I need for simultaneous calls and is there any latency for SIP G711 to IAX2 and vice-versa , ... etc ? Thanks in advance for any info, Geo ___________________________________________________________________________