similar to: Problems with IAX2?

Two asterisk boxes 150 miles apart, both cvs-head as of this morning (and since Sept 27th), connected via iax2 with low-utilized ds3 internet, C7960 calls exten on remote system (also C7960), and call goes to VM. No other calls in either system (eg, no load). Both boxes have iax config'ed as: trunk=yes allow=ilbc jitterbuffer=yes Recorded VM messages are very distorted. Changing only

I have some stuck channels that I think I'm going to have to bounce Asterisk to get rid of, but am curious to know what they are and how they've managed to accumulate. The show up with a channel identifier of '(None)' as in the output below, and do not show up in the soft hangup list, and so can't be cleared by that method. Here is the output from iax2 show channels:

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

I've been having problems with incoming IAX2 calls - some work, but a large fraction are answered with "dead air" or disconnects from my IAX provider. Disabling the jitterbuffer seems to eliminate the problem (so far)! Has anyone else seen this? I'm using 1.2.6, but I'm not sure what my provider is using. A snippet of the a failed incoming call IAX2 debug is attached

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

I have multiple voiceage g729 licenses installed on a RH9 box, and have a remote C7960 configured to use it (low bandwidth). In calls like: Remote C7960 -> g729 -> asterisk -> g711 -> C7960 the audio is oftentimes rather choppy. Changing the remote 7960 to use g711 seems to eliminate/reduce the choppyness. Any ideas on what might be behind this?

Are these things possible? 1) Set the local Asterisk jitterbuffer size, but only for a particular connection. I'd like to force Asterisk to use a particularly large buffer in certain cases. Should I expect this to work? [general] jitterbuffer=no register => username:password@parcelfarce.domain.net ;parcelfarce register => username:password@iaxtel.com ;iaxtel [parcelfarce]

pstn -> sip gw -> * -> C7960 When I dial into * via the pstn, I hear the ivr menu just fine (good quality). I press 3000 (valid extn), and I begin to hear ringing however the ring back is very very choppy. I answer the C7960, and speech is clear in both directions. Place the C7960 extn on hold, and the MOH is very choppy. Checking 'sip show channels' indicates both the sip gw

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,

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 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.

(Apologies to those Toronto Asterisk Users' Group folks who have seen this message... I figured I'd have more success with a wider audience) I'm trying to boot a Cisco 7960 from an ISC DHCPD server (3.0.3 on FreeBSD 4.11), so far unsuccessful, and getting some odd behaviour on the wire. I wonder if anyone has done this before and therefore can validate whether or not the traffic I am

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?

Have a strange problem... When a C7960 calls the Polycom ip600, the ip600's first line button blinks, the ip600 display shows the proper callerid, but the phone does not ring at all. If I call the same ip600 from a bt102, the ip600 rings properly. If I call the same ip600 from another C7960, the ip600 rings properly. All phones and asterisk are on the same lan within a few feet. The

I'm having a problem with intersite calls over IAX2 being abruptly terminated. Nothing odd shows in any of the logs for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. How sensitive is IAX2 to lost frames, lag spikes or large variations in jitter with the GSM codec and: bandwidth=low jitterbuffer=no trunkfreq=100 ; Raised from

Hi all, I'm experincing IAX2 call drops for about 20% of calls. I tried 'notransfer=yes' and 'jitterbuffer=yes' but to fail. My system configuration is like this. PSTN<========>Asterisk(TDM/Fxo 4port*3)<=====LAN(IAX2)=====>Iaxclient library And iax.con is... ----------------------------------------------- [general] port=5036 disallow=all allow=gsm

Using the following defnitions with a C7960: exten => 3001,1,Dial(SIP/3001,15,r) exten => 3001,2,Voicemail2(u3001) exten => 3001,102,Voicemail2(b3001) exten => 3001,103,Hangup If someone is on this phone (real conversation) and another call comes in, the second call goes through the 15 second timeout and is dropped into the 2-priority as "unavailable" (not the 102 busy as

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

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

Ok sorry for not being specific. I am having a problem when people outside call in to my number which terminates at VoicePluse then The send IAX to me and I do not get any tones. People press buttons but it just goes to the next dialplan fall through. It happens 60-70% of the time. extentions.conf [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no