similar to: asterisk 1.2.1 crashed

Apparently we've been having calls sporadically drop. We're using an IAX outbound trunk and SIP adapters on the inside. Below is a log excerpt detailing one of the calls which dropped, and it looks largely normal to me except for this: Jan 5 13:31:07 DEBUG[3776] channel.c: Didn't get a frame from channel: IAX2/teliax-2 Jan 5 13:31:07 DEBUG[3776] channel.c: Bridge stops bridging

I am using Teliax to terminate my calls, and I have 3 licenses' for g729 from Digium. "show translations" verifies that the registration took place. When I place a call, having "allow=g729" as the only allow option in iax.conf, I get the following error: WARNING[361]: chan_iax2.c:6017 socket_read: Call rejected by 208.139.204.228: Unable to negotiate codec If I place a

hi, Can I use quota with ext3 on a loaded system without experimenting deadlocks nowadays? I'm using rh7.3 kernel 2.4.18-19.7.x thanks -- Juan Pablo Abuyeres <jpabuyer@tecnoera.com>

Not to point the finger at Teliax, but I'm having some unique problems with their service that are as yet unexplained. Incoming calls are fine. Outgoing calls don't work, though they did at one time. As of today, I'm running the latest code from CVS. -- Called teliax/13143212222 -- Call accepted by 208.139.204.245 <http://208.139.204.245> (format ulaw) -- Format for call is

Hi All, I was able to install Asterisk and make outgoing calls. Recently I purchased two DID's and I am facing a problem configuring them to my Asterisk, I hope with the help I get from this list I will be able to configure successfully. Mu errors are Feb 28 08:31:58 NOTICE[19133]: pbx.c:1331 pbx_extension_helper: Cannot find extension context 'context_mantra2' Feb 28 08:31:58

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

The situation is one of my asterisk servers is behind a NAT firewall and one is not. Both servers have multiple IAX peers. The NAT firewall has port 4569 mapped through to the asterisk server behind. But, the natted server is almost permanently unreachable from this non-natted server, even though, the non-natted server is almost permanently _reachable_ from the natted server. Details are below

To answer my own message, I figured out a solution (untested) about 10 minutes after posting and leaving the office. Doh! Anyway, the solution (now tested) was to make the Asterisk server behind the NAT register with its peers. Despite reserving port 4569 in the firewall, that was not enough in this particular NAT firewall - it was only being reserved for one connection. Kind regards, Sebastian

I am sending this again since I haven't get it back for twelve hours: When I originate call to IAXComm, more or less one of tree calls fails for no aparent reason. Originating calls to SIP clients works as expected. Anybody has similar problems? Is it asterisk or client problem? Asterisk log: Jul 15 00:00:04 DEBUG[1179663]: manager.c:1018 process_message: Manager received command

I have two units at customer locations in the Caribbean registering to a server in the US. Both units are connected to the Cable TV company's internet feed. If I run mtr to the units I see clean internet and low latency, but when I watch the CLI, I see constant problems. The audio quality is terrible, but I can't see why. Sep 3 16:42:46 NOTICE[20423]: chan_iax2.c:7014 socket_read:

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,

I'm new to the Asterisk software but have successfully set it up to make and receive calls using FXO cards, voicemail transfer etc. I can successfully call the Digium test IAX using the examples provided. I have signed up for an IAX tel account and got a number. The extensions have been set up as per the examples from IAX tel. However when I try to place a call this is what I get: --

I've * in a small office with 10 internal sip extensions on aastra's. Outgoing is pstn over dahdi, voip over teliax and iax to another office. This morning no calls could be made: iax to branch offfices, voip iax over teliax, pstn, or even internal extensions. The aastra's showed "Not in Service". A "core restart now" got everything working again. Before I

Hi.... I'm having a extrange problem.... I cant register with Iaxtel or call to digium... But i cant make or recive IAX calls... ( I made some one with irc users ) Any idea why? At my logs i have this from iaxtel: NOTICE[196621]: File chan_iax2.c, Line 2832 (register_verify): No registration for peer 'xmarts' (from 192.168.0.11) NOTICE[196621]: File chan_iax2.c, Line 4389

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

Hi, if I dial meetme from extension 200 directly it works ok - I get moh as only user (first trace). If I dial to other local extension and trasfer from there I get second trace... Apparent difference between those two is warning : Jan 27 11:06:33 WARNING[6133]: res_musiconhold.c:466 moh_alloc: No class: random What this could mean ? Direct Call log-----------------------------------------:

Hello, I updated CVS and nobody can call me any more with my IAX number 17007591228. I can only call other number but nobody can call me. This is what I get on debug when I call myself: -- Executing Dial("SIP/1011-7424", "IAX/bartosz:password@iaxtel.com/17007591228@iaxtel") in new stack -- Calling using options

Good evening. We have some heavy-loaded servers on ext2, and we want to migrate to ext3fs. But we need full and stable quota support. I have headrd that there are some problems in quota usage under ext3fs. Is it true? Should we decline the ext3 usage as impossible in our servers? We need high-level stability in our server (hosting). Thanks before. -- Best regards,

Hi, I'm having what looks like a deadlock using kernel 2.4.18 + ext3 + quota on a pretty loaded system. meanwhile, i moved back to ext2 again. I'm using mount-2.11g and Quota utilities version 3.05 (I also tried with 3.06). I also tried with kernel 2.4.16 just in case, but the problem exists anyway. Anything I can do? JP

hi guys, I have an asterisk server and a mediatrix 1204 gateway. I make calls through the mediatrix unit (only outgoing calls). The problem is, every call I make through the mediatrix unit is logged in the cdr as 'ANSWERED', even if the call was 'NO ANSWER' in practice. Any ideas how to make cdr records accurate? Thanks! -------------- next part -------------- An HTML