similar to: chan_iax2.c:1219 __send_lagrq mesages

We get this error message [Mar 23 10:10:09] WARNING[4325]: chan_iax2.c:1056 __send_lagrq: I was supposed to send a LAGRQ with callno 14034, but no such call exists (and I cannot remove lagid, either). -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'IAX2/brandx-14819' == Spawn extension (usawide, 3, 2) exited non-zero on 'DAHDI/1-1' -- Hungup

I know in advance maybe I'm overlooking something stupid, however I'm really lost and cannot find the solution... Situation: - asterisk-1.2.13 on a linux box with no iptables active. - one iax2 peer defined - one wildiax phone running on my laptop the soft phone is configured to connect & register on asterisk, however, I cannot get it working. What am I missing? Please help!!

I'm having some trouble with placing some iax calls over an openvpn: Setup A is a 1.8GHz Celeron, T100P attached to a Zhone Zplex. Setup B is a 266MHz P2, T100P attached to a Zhone Zplex. Setup C is a 700MHz P3, T100P attached to an Adtran TA 750. Setup D is a 233MHz Pentium, with an X100P. Setups A and B are on the same physical network. IAX calls routed between them work fine. Setup D is

Hi everyone, IAX2 softphone is not working with the “Unregistered” part in the asterisk (chan_iax2.c and data_pgsql.c) But with the Xlite softphone the unregistered worked properly and ast_data properly updated the IP address and port number in the database. I have seen some codes in the chan_iax2.c file: “ast_verbose(VERBOSE_PREFIX_3 "Unregistered '%s' (%s)\n", p->name,

Hello, i have a simple but annoying problem. I have the following entry in /etc/asterisk/externsions.conf file: ---<Cut Here>--- exten => 10100,1,Wait(4) exten => 10100,2,Playback(transfer,noanswer) exten => 10100,3,Dial(${PHONE30},30,t) exten => 10100,4,Background(extension) exten => 10100,5,Background(is-curntly-unavail) exten => 10100,6,Voicemail(9999) exten =>

Can any one help? Thanks. we have two * servers (Version 1.2.1) and one 1.09 server. Calls between these two 1.2.1 servers have odd behavior. But call from 1.09 to 1.21 server working fine in either situations. See below pls: Local server iax.conf [tosyd] username=mel type=peer secret=xxxx host=198.168.2.66 remote server iax.conf [mel] type=user secret=xxxx host=198.168.2.67

Hi, My termination with sixtel stopped working, is it something I did or anybody else is having the same problem. I am attaching log: *CLI> -- Executing GotoIf("SIP/300-fbe0", "1?4") in new stack -- Goto (macro-dialout-default,s,4) -- Executing GotoIf("SIP/300-fbe0", "1?6") in new stack -- Goto (macro-dialout-default,s,6) -- Executing

============= SJphone Log ============ Outgoing SIP session Respondent: (sip:8612@192.168.2.2) Remote client: Started: May 26 16:33 Accepted: no Ended: May 26 16:34 End reason: Call rejected: 503 Service Unavailable =============== Asterisk Debug ================ Executing Dial("SIP/2201-a83e", "IAX2/<FWDNUMBER>:@iax2.fwdnet.net/612|60|r") in new stack --

Hi, I'm testing an upgrade from 1.4.18 to 1.4.37 in a VM prior to putting it into production. Ive done this by installing 1.4.18 onto the VM, putting my config files in place and then installing 1.4.37 over the top (which is what I'd have to do on production). I've found a few issues in the config files, but nothing I couldn't handle until... I hit inbound IAX issues. My

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

Greetings, I have a somewhat unique (I think) configuration that I am testing involving MeetMe conferencing and have encountered a problem that I'm not quite sure how to solve. Here is a brief description of my setup for the background. I wanted to offer the ability for users to mute and unmute themselves while in a conference. If they enter a conference as monitor only, they are

Ok so it seg faults when I try to dial out through IAX(voiptalk.org), ofcourse it doesn't if I remove allow=speex :P ---- (gdb) run -c Starting program: /usr/sbin/asterisk -c [Thread debugging using libthread_db enabled] [New Thread 16384 (LWP 28283)] [New Thread 32769 (LWP 28285)] [New Thread 16386 (LWP 28286)] [Thread 16386 (LWP 28286) exited] [New Thread 32771 (LWP 28287)] Asterisk

Asterisk Project Security Advisory - ASA-2007-015 +------------------------------------------------------------------------+ | Product | Asterisk | |--------------------+---------------------------------------------------| | Summary | Remote Crash Vulnerability in IAX2 channel driver |

Asterisk Project Security Advisory - ASA-2007-015 +------------------------------------------------------------------------+ | Product | Asterisk | |--------------------+---------------------------------------------------| | Summary | Remote Crash Vulnerability in IAX2 channel driver |

For some reason, starting just today, 1 out 3 of my asterisk servers is having issues calling 1 other server. The only issue I see is that when it registers with the problem server it is using port 1027, not 4569. ie: Registered to 'Server 1', who sees us as 'Server 2':1027 Server 1 then proceeds to timeout trying to register with Server 2. The way I have each server registering

I'm new to this list, and I apologize if this is an already answered question, but my Google-fu was not strong enough to find the answer if it was. I'm having a problem with DTMF on incoming IAX calls. For the first few seconds of the call (between maybe 1 and 15, it varies from call to call) everything works fine. After that I continue get DTMF_E messages from the remote IAX server

Hi, I thought I'd repost this to the -users list for some background on the jitter buffer and its workings and remaining issue.s I'll also pu a little executive summary here at the top: Where a channel is native bridged to another iax2 channel: 1) Lag is not measured and will usually show 0ms. Any other number is an old measurement from the start of the call 2) The jitter

On Mon, 01 Nov 2004 11:38:45 +0100, <lartc-request@mailman.ds9a.nl> wrote: > ******** McAfee GroupShield Exchange ********** > ******** Alert generated at: Monday, November 01, 2004 12:54:27 FLE > Standard Time > ********************************************************************** > > > The item GSEBODY_213471_5524_1099306467.txt has been

Hello, does anybody here know the exact meaning of "numopen" in the open/close file log messages? Does it mean the number of open sessions on the file: - at all? - per user? - per client? Sometimes numopen is greater than zero ofter the last user has closed as file. Is this an error? Here an extract of <hostname>.log --------------------------------- [2001/10/31 08:57:24, 2]