similar to: Audio delay w/ call forwarding

Hi all, I'm trying to get IAX2 trunking between two * boxes and am having extreme difficulty :) What happens is when the sending * server (the one initiating the call) receives the ACCEPT back from the receiving server it immediately replies with INVAL. I've checked the code and it seems to be not matching the accept packet with the relevant item in the iaxs array due to the following

I've been having 'gappy' audio problems with nufone for about a week now but I think I've nailed it down. Setup: office* - iax2 - colo* - iax2 - nufone office* and colo* are identical physical hardware (Xeon 2.8, dual ethernet, solely used for Asterisk) -- they are joined together through their second ethernet ports over a dedicated 2meg SDSL link. One hop between office* and

In my IAX config file I have: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) delayreject=yes disallow=all allow=ulaw allow=gsm jitterbuffer=yes forcejitterbuffer=yes mailboxdetail=yes dropcount=3 minexcessbuffer=80 jittershrinkrate=1 notransfer=yes allanrobertson- 209.23.224.97 (D) 255.255.255.255

Our * box is behind NAT with 4569 forwarded to it. I'm using stable 1-0 version. We're pure VoIP. Our SIP phones can use * to dial out, and can receive incoming calls, and conference in another SIP. However, if we try to conference 2 IAX connections together, we'll get one-way audio for one of the connected IAX devices. I found that this also happens if I use an IAXy to call out from *

Hi all, We keep getting these and all the calls between these two asterisk boxes get dropped. what is going on here, I have been trying to solve this problem on my own but maybe I don't have the trunk setup right. also I have posed the output of my full log of the machine with the zap interface, the other is using ztdummy. IAX.conf on machine 1: [general] port=5036 ;iaxcompat=yes

hi We take calls inbound via SIP from our Cisco PSTN gateways, and pass it to customers using IAX (they run their own asterisk servers). We've noticed that asterisk is transcoding the call into a different codec, if the customer prefers a codec different to that which our cisco gw prefers. As such, the quality of the call can degrade. We'd rather asterisk just passed through the RTP

I'm trying to make a call from an IAXPhone client - through the * PBX to an 888 number using the IAXTel link. I'm using the basic conf files for extensions and iax. I get successfully connected (at the "Attempting native bridge" line of the output) and am then able to talk both ways for 20 to 30 seconds and then the IAX phone appears off line. If I wait on the PSTN line for

Hi, I have a customer who experiences, once in a while, one-way audio... That is... they can hear the person they called, but the person can not hear them. The customer is connected via IAX2 to our softswitch. On the customer's end I have the following config in iax.conf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0 ; Address to bind to (all

I need to debug a call quality issue with remote users on the other end of a satellite link. The symptoms are: we here on the Internet side can hear them just fine. On their end, things work sorta OK most times, but they often suffer from severe dropouts and digital warbling, both of which I attribute to them missing packets. Often times they can't make out a word we are saying while we can

Hi All, I've a single 1.2.17 Asterisk system. Gradwell here in the UK is used for PSTN calls via IAX2. Our 'net link is a dedicated 2Mb fibre connection (of which we have ever used 50% max bandwidth). We've no E1/T1 links, everything is IP based. My boss complains that many of the calls he holds with others has a bad quality. He also says that its not just him. My iax.conf file

Hi! I just wondered if anyone would mine posting their successful jitter buffer settings here for me if they get a moment ?? I've spent a few hours trying to set the jitter buffer up reasonably logically and can definitely tell it makes a difference and can introduce latency and echo if setup incorrectly but I can't see a good post anywhere describing properly what the three settings

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,

All, I have spent the last couple of days looking through the mail archives and the documentation on the Wiki, but have not been able to find a solution to the problem. The version of code I am running is from CVS as of 6/30/04. What happens is that when I make an IAX call to another IAX client the caller receives a really bad echo. All of the documentation I found around using

Hi all, I have a VOIP phone with the PA1688 chip; my firmware is V1.42.028. This IP phone is registered in an Asterisk PBX and I've a problem when I dialing internal number. If I dial an internal number, like for example 102, the IP phone takes 35 seconds to send the number to Asterisk; here below the debug output 192.168.0.75: first 192.168.0.75: 1 192.168.0.75: 10 192.168.0.75: 102

I am trying to work out IAX <--> IAX communications with my * box. I have a registration with iaxtel and I thought I would start there for my learning. I am able to call the number for Digium's support line (700-428-6000), but the sound is horribly chopping. Some reading revealed the jitterbuffer settings, so I enabled them in iax.conf. I have the following now: ; Inter-Asterisk

Hello all, What my app does is accepts a call in on a Dial-In Number (DID) via IAX, and then prompts the caller for the top secret password (123) and then authenticates the user and prompts them to dial in the number they'd like to call. Once they press pound after dialing in the number it will read it back to them, if they press pound it will attempt to connect via the second IAX provider,

Hi- I'm using IAX between two boxes, where one box is located in US and the second in Europe. I'm trying to achieve the best voice quality and mainly reliability between these boxes and looking for hints and experience of others. Facts: - Asterisk 1.0.7 - RTT varies from 130-170 ms, depends on time and actual Internet throughput Questions: - What is the sugested codec for such setup?

Hi everyone, I am trying to connect 2 asterisk servers via IAX, but it just fails to do so.. I'm using SIP to connect the IP phones on the LAN at the 2 different physical locations where each server resides and I'm able to communicate on my LAN via SIP without any issues. The problem is that I'm unable to make Asterisk servers talk with each other via IAX.. Here is my issue.

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I'm trying to come up with good jitterbuffer related settings for my Asterisk boxes. I ran 4 pings for about 2 days from my main Asterisk server to remote Asterisk servers. During that time there were some large file uploads which caused the max rtt to be quite large. Here are the results: pkts loss min avg max mdev 132013 %0 70.36 78.13 1967.37 36.04 132013 %0 98.95 120.46 2419.24 111.26