similar to: Problem with chan_iax.c implimentation causes bad audio?

We upgraded all five servers to 1.2.4. We tried trunking/notrunking. End users use an IAX2 softphone on their desktop PCs. Agents are VLANed and all IAX2 traffic is QoS'd on all LAN and WAN legs. Calls flow from the agents to the local Asterisk server as IAX2/ulaw. Then they went over the IP-VPN as IAX2/g729 (we tried ilbc and straight ulaw as well). Calls get to the PSTN from the

We have three remote call center Asterisk servers communicating with two central Asterisk boxes over a private IP-VPN with QoS. All systems were running Asterisk 1.0.7 communicating via IAX2 with little or no quality issues at all. Once we upgraded to Asterisk 1.2.4 call quality with IAX2 was horrific. We tried with/without jitterbuffer. We messed with every jitterbuffer parameter. We tried

Asterisk 1.2.7 Redhat 9 I have DiDs from two different ITSP both set up as IAX2. Each one works when it's the only one in my iax.conf, but when I have them both defined in iax.conf at the same time, only one will work. My iax.conf is provided below. Any ideas how to fix? I'd like to use both DiDs! Thanks, H My iax.conf is below. When I dial the DiD provided by ITSP_B, the other

I was using G729 with Asterisk 1.07. With the new ability to do packet loss correction with ILBC, I felt I'd give it a try. The new PLC does not work with G729. I don't use Speex because my softphone does not support it. This is a 1.5mb IP-VPN connection with prioritized QOS for port 4569 (IAX2). I've never really stressed the bandwidth. Typically, only 10-20 concurrent calls.

Hi, this is the scenario: One * is placed in a central location with more than enough up/down bandwidth. One * is placed behind a DSL 3000/384. Both * are linked via IAX trunking. Everything is fine until the upload channel of the remote site is filled with a download, then heavy voice distortion starts. Well of course this is expected. So I fooled around with HFSC QoS scheduling on the remote

I've got IAX2 setup between two servers with this config: I have two servers on a switch: asteriskm is 192.168.0.160 and asterisk1 is 192.168.0.161 asteriskm has a Sangoma T1 card in it. I want to route calls from asteriskm to asterisk1 which will run an AGI IVR for the call. Config is below, but my problem is that 90-95% of the time when I start asterisk on the two servers I get the

Howdy, I recently saw something strange with a call between *'s over IAX2. There are actually 3 *'s involved. The setup is like this: SIP phone ------(ulaw over LAN)------ *1 -------- IAX2 (ulaw over Internet) ---------*2--------(GSM over Internet) -----------*3--------(ulaw over LAN)------ SIP phone Now what is shown below is the Asterisk in the middle, that is doing the

Hello all, I tried to compile Asterisk from CVS yesterday. I would like to try gnophone as IAX client on the same Asterisk server with a sound card. I didn't have any problem of compilation. Here are the config files in /etc/asterisk: /etc/asterisk/asterisk.conf /etc/asterisk/extensions.conf /etc/asterisk/iax.conf /etc/asterisk/indications.conf /etc/asterisk/logger.conf

All, I have an issue with IAX that I can't comprehend. Approximately every eight minutes my servers go unreachable. They stay unreachable for exactly 10ms. I have two servers running IAX and it happens on both servers simultaneously. I have searched the archives and see similar issues, but not the exact same one. I am on the current CVS stable version of *. Also, during IAX calls,

Is there any way to have more specific disposition codes in the CDR? Currently there are only 3 values: ANSWER, NO ANSWER, BUSY. In this way, when i call a cell phone that is switched off i get "NO ANSWER", while i would like to be able to log that the call is not answered because "The customer you have dialed is unavailable at the moment". The same for "non

Hi My first post get no answer :=<, i post new with new elements. I have two Asterisk server, running on Asterisk 1.6: SRV1 = 192.168.0.5 on Asterisk 1.6.1.4 SRV2 = 192.168.0.20 on Asterisk 1.6.1.8 I want create a link for exchange call. on Srv1: iax.conf: [general] bindport=4569 bindaddr=0.0.0.0 language=fr bandwidth=low jitterbuffer=no forcejitterbuffer=no autokill=yes

All switches and routers give highest priority to traffic on IAX2 port 4569. We use DSCB values over the IP-VPN to prioritize it as well. This did not change with the upgrade, as we can still see proper packet coding. The softphone is provided by our vendor Aheeva. It is the same IAX2 softphone they use in their own call centers. Funny thing is that they say that moving to Asterisk 1.2.4

Hi guys, i've been using iax2 show netstats and i wonder if someone could explain what all these means, just in case i have them wrong. Because i am looking for something that tells me that there is delay , and/or packet loss. -------- LOCAL --------------------- -------- REMOTE -------------------- Channel RTT Jit Del Lost % Drop OOO Kpkts

Jean-Marc Valin wrote: >>Heh. I guess after playing with different jitter buffers long enough, >>I've realized that there's always situations that you haven't properly >>accounted for when designing one. >> >> > >For example? :-) > > I have a bunch of examples listed on the wiki page where I had written initial specifications:

I have looked all over the site(s) for help. But heres the problem. Im missing something. In coming works fine from FreeWorld via IAX. But when Dialing out i get: May 13 13:42:01 WARNING[1150495040]: chan_iax2.c:5256 socket_read: I don't know how to authenticate iaxtel to 65.39.205.121 my IAX.conf if as follows [general] port=5036 register => ######:xxxxxxxxxxxxx@iax2.fwdnet.net

Hello, I understand that only the destination of a call should do jitter buffering. So, if IAX2/PhoneA calls IAX2/PhoneB through my server (no transfers), PhoneA and PhoneB need to perform their own jitter buffering, and Asterisk will just forward the frames, correct? What happens if the peer does not support jitter buffering, but is close by so there's no need for jitter buffering? My

Hi, all Please help me. My platform is RedHat Linux 9.0. I have a wildcard x100p. I just installed asterisk by following step: # cd ../zaptel # make clean ; make install # cd ../libpri # make clean ; make install # cd ../asterisk # make clean ; make install # make samples When I test Asterisk typing # asterisk –vvvvc I get one error and one warning: [chan_iax.so] => (Inter Asterisk

Lets talk about SIP and IAX2 1. The good and bad of both 2. What is the better one and why 3. and any other information that maybe use full -- Best regards, Al Bochter Bochter Services (Voip PBX) Toll Free: 866-638-1254 EXT: 250 (Voip PBX) Free World DialUp: 780217 EXT: 250 (Voip) Cellular: 712-432-5401 http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold

Hello, I have searched the lists, and have seen this message posted numerous times, however there never seem to be any replies to it. I was curious if anyone had figured out the WARNING: File chan_iax2.c, Line 5466 (set_config): Ignoring port for now Everything seems to work just fine, however when I try to change iax.conf port to 4569 it doesn't seem to like that either. Thanks, Brent