similar to: Asterisk 1.07 crash under Debian Sarge

I am hoping some of you can help me troubleshoot this problem I am having with my home asterisk machine. I have incoming POTS service using a SPA-3000 (extension 119). Calls on that line go to an attendant recording that offers a menu choice: press 1 for Nancy, press 2 for the rest of us. In reality, pressing anything other than 1 sends the call to the rest of us by dialing both extensions 101

Has anyone seen an issue with SIP phone (polycom 500) dropping calls at irregular intervals with no errors in the asterisk log files? I am having this issue as described and it is a complete pain in my rear to trouble shoot because when I call my cell phone I can get a call to last over 30 minutes yet when I call another office that uses a standard pbx I can't get past 10 minutes. For some

Hello everybody. I have an customer asterisk 1.0.5 running well since 3 monthes, 2 TDM cards 4 FXO, 4 FXS. Since one week, unable to pass call between Zap and Sip getting the "exception on 15, channel 1" The * box is connected to an eads PBX and it seems that failure started when they make some changes on the PBX. Have someone an idea and what is causisng this failure? Here are the

Hello, I have problem to make SIP calls, i have asterisk + PC InterP4 + Digium TDM400P here is the content of the sip.conf: [SIP_PROVIDER] type=peer fromuser=testcomclient username=testcomclient secret=testr host=IP_SIP_PROVIDER ;dtmfmode=rfc2833 context=interne canreinvite=no ;allerid=Beer disallow=all allow=ulaw allow=gsm allow=g723.1 ; Asterisk only

Hi, I am having trouble with dropped calls in Asterisk. I've done a bunch of searching but all I could find was setting busydetect and callprogress to yes in zapata.conf to help combat the problem, but I did this to no avail. The following is the output from /var/log/asterisk/full at the time the call was dropped on me. May 13 08:37:13 DEBUG[5379]: Stopping retransmission on

Hey, I've done some searching for this and never really found a concrete answer. Is there a specific reason or solution why just in the middle of a call Asterisk will drop it and I'll get dial tone again? Anyways, this is the output from /var/log/asterisk/full at the time of disconnection: May 13 08:37:13 DEBUG[5379]: Stopping retransmission on

Hi, I am trying to list the shares on a windows ME machine and I am getting the following [root]# smbclient -L 192.168.0.253 added interface ip=192.168.0.1 bcast=192.168.0.255 nmask=255.255.255.0 session request to 192.168.0.253 failed (Called name not present) session request to 192 failed (Called name not present) session request to *SMBSERVER failed (Called name not present) I have also tried

I would be very glad of pointers to information on how the concepts of packages, namespaces and environments are interrelated in R. I am trying to get a handle on this both so I can delve further into understanding other people's code and so I can organize my own in a more coherent manner. From my reading about environments it seems they function as what I would intuitively call

Hi Guys, I have a really odd one here. We are dropping calls occasionally... there are no error messages being spat out, but I can see this suspicious behavior in the debug logs; Jun 30 14:58:48 DEBUG[19856] pbx.c: Function result is 'Other' Jun 30 14:58:48 DEBUG[19856] pbx.c: Function result is '(null)' Jun 30 14:58:48 DEBUG[19856] pbx.c: Function result is 's' Jun 30

I'm using Asterisk 1.0.4 with AMP and Broad Voice. I have that with only 5 XTen Lite phones. I'm able to call / etc with internal phones just fine. I can call outside Vonage Numbers, and other BroadVoice Numbers. I have vonage where I live (626) and can call that fine. However, other 626 numbers I get similar errors as below. However, everytime, I try to call cell phones, and or

Hello folks, I'm having trouble configuring Asterisk to play an "invalid extension" message to anyone dialing an undefined extension. First I tried using the 'i' pseudo-extension, but it didn't work at all; searching the wiki I found that page: http://www.voip-info.org/tiki-index.php?page=Asterisk%20i%20extension where it basically says that the 'i'

Hi all, I've been running Asterisk with a TDM400P for about 6months, no problems. 2 in/outgoing analog lines, one analog phone. Recently I was messing with the XTEN client, got to finagling with things, and not knowing what was wrong I upgraded from 1.0.7 to 1.0.9 (both asterisk + zaptel). I was testing various things, and found everything worked except outgoing calls. So I checked

All, The Windows 2000 client is on my side of the firewall (the error message is: "remote computer is not available"), the Windows XP client is on the other side of the firewall, (the error message is: "The specified network name is no longer available"). I am able to ping both the IP address and server name of the CIFS/Samba server. I cannot perform the mapping, via Map

I am using GRASS 5.7 and R 2.0 for OS X. I have a simple xy location I am doing some theoretical work with and would like to import maps into R using the interface in order to do analyses. Yet when I attempt to do this, I get the following error: >G<-gmeta() Error in gmeta() : region for current mapset is invalid line 11: <top: 100> run "g.region" I traced the

I've been monitoring this problem for almost a month now. I realized that it is more extensive than what I described previously. I can very easily replicate this problem on every Zap channel. Following is the senario: 1. Call Zap/5 via say SIP/15 -> Zap/5-1 created and starts to ring 2. Call Zap/5 via say SIP/21 -> Zap/5-2 created and starts to ring 3. Hangup SIP/15 ->

Running a Dell Pentium III with Solaris 8.0 for Intel installed. Have X11 and Flex installed Downloaded Wine-20010731.tar.gz and am attempting to install on the machine. Please HELP!!! thanks warren I am getting the following errors: after running make depend && make LD_LIBRARY_PATH="../../unicode:$LD_LIBRARY_PATH" ../../tools/winebuild/winebuild -fPIC -L../../dlls -o

I have searched all over the web and have not really found anything related to this error.... The only thing I found is related to a system stops responding on DTMF, which does not happen here... THe following is the output from the CLI: *CLI> 2004-08-09 17:36:29 DEBUG[229390]: chan_sip.c:2332 sip_alloc: Allocating new SIP call for 640E2D47-E98B11D8-8FDBE54A-5FB5A0CB@65.67.76.30

Hello, I have a TDM400P with 2x2 configuration of FXOs and FXSs. I set a test extension of '444' to dial out a specific zap trunk and call a local #. First time I call out to '444' everything works fine. If I hang up the call, and within 10 seconds dial the same number again, I get a fast busy. Seems it isn't letting go of the trunk or something, and I don't have a

Hi! We have a SIP server with a TE410P card with asterisk version Asterisk CVS-D2005.02.12.14.37.11-04/13/05-16:14:03. Sometime the calls get disconnected with now reason and the users get a busy signal. The log file show this for one of the calls that got disconnected: Aug 31 22:51:53 VERBOSE[3911]: -- Accepting call from '46362302' to '36917474' on channel 0/5, span 1 Aug

Well, I'm writing yet another queue_log analyser program in PHP, and I have noticed the following entry in my queue_log file from today: 1104796626|1104796618.532|queue|NONE|ENTERQUEUE||no 1104796664|1104796618.532|queue|NONE|EXITWITHTIMEOUT|1 So, pretty sure that I didn't make someone wait 30 minutes in my queue. extensions.conf snippet: [remote-oldnum] exten => s,1,Answer exten =>