similar to: /var/spool/asterisk/outgoing/ prematurely hangingup

I have a PSTN termination provider "foo" which will accept standard U.S. calls in the form 1<10 digit ph#>. I have an outbound route named "foo", with dial pattern "5|.", with the only entry in trunk sequence being "IAX2/foo". I have an X-lite local extension, on which I can dial 51<10 digit ph#>, and asterisk will call out over foo and the

Hi, I have a custom agi which at times does not exit gracefull and crashes in between. The logging options are set to the maximum but I dont see something conclusive in the asterisk log. I have noticed it crash after issuing the "SAY NUMBER" and "GET DATA" agi commands and the agi is spawned with no apparent reason after that. I tried running the application locally and

Hi, I'm trying to compile two files together with debug information but seems that LLVM is getting the DW_AT_stmt_list wrong when ld is linking the final executable. Originally, I tried on ARM with Clang (+llc+gas+ln) and, the object files, the DW_AT_stmt_list were null, as expected. When linking, they should point to the offset in the line table, but all of them are still null, so pointing

See "DwarfDebug problem with line section" thread on llvmdev. Bottom line, we may need a target specific patch for targets that do not follow dwarf standard (as per my reading) in this particular case. - Devang On Wed, Aug 25, 2010 at 5:49 AM, Renato Golin <rengolin at systemcall.org>wrote: > Hi, > > I'm trying to compile two files together with debug information

Hi, I am setting up a test call center using *. I am using one Zap channel (Wildcard TDM400P REV E/F -- 4 FXO modules) for incoming call and sip phones (SjPhone) for call agents. I have setup queues and agents. While testing I found that if the agent presses * key in soft phone while attending calls Zap channel gets hung up, and another customer can call. But if the caller hangs up (for example

c/c'd to the Asterisk list as this is probably relevant to Asterisk as well. My detailed study of the operation of the 'outgoing' directory reveals that TXFax() does not delete an expired fax batch file (In the 'outgoing' directory) until after the end of the dial plan execution. Is there a way to get it to delete any expired fax batch file before the end of the dial plan

Hi all, i 've written a little Callgen script for generating calls through the outgoing spool directory. The calls goes over 8 ttyI devices to another pbx and come in through other 8 ttyI devices. But when i generate the calls, sometimes * register the calls but never initiate them. Especially when the files come to fast into the outgoing dir. What can be wrong ? Is it possible that the

Hi all, i tried to write a call generator script. It generate i file like sample.call in ast src tree. But when i set the callerid and make a call over Modem[i4l] then the caller id is not set for the outgoing call. Is it impossible to set the callerid at this time for ttI devices? Thanks for help, Thomas. ******************************************* beroNet technologies GmbH Dipl.- Ing.

I've been playing with the outgoing call spooling feature a bit lately and it all works as it should with the exception of one irritation. I'm mostly using SIP to talk to the phones and using G.723.1 I copy the call file into the spool/outgoing directory and the originating phone rings. I pick it up and the remote phone rings. However there is dead silence from the originating

I've just noticed that if you start a call by writing a file to /var/spool/asterisk/outgoing the cdr created on termination logs the call placed to the local extension - not to the destination in the PSTN. Hence there is no record of the PSTN number dialled. I guess most people want to log the outgoing portion not the local call leg? Anyone know of a setting that changes this? Iain

Hello everyone I'm trying to make asterisk get a call out using the .call system. The setup is A@H 2.6 This is the content of the file is : <<< Channel: Zap/g0/052MYPHONE MaxRetries: 2 RetryTime: 60 WaitTime: 30 # # Assuming that your local extensions are kept in the # context called [extensions] # Context: ext-local Extension: 210 Priority: 1 >>> I'm

> you should mv the file (and in the same filesystem, so 'rename' is used) > You might want to chmod or even chown the file first as well. I wrote a little script that does all of this before the .call file is mv'd into the outgoing directory: cp /tmp/test3.call /tmp/test1.call chmod 666 /tmp/test1.call chgrp asterisk /tmp/test1.call chown asterisk /tmp/test1.call mv

I've seen other posts about this problem, but I haven't found a solution. I'm dumping eight call files into the "outgoing" directory at one time. Three of the calls are successful while the other five are lost. Here is the call file: Channel: Zap/g2/3036701917 MaxRetries: 1000 RetryTime: 60 WaitTime: 45 Application: TxFAX Data: <filename>.tiff|caller Note: All

Hi list, I'm using asterisk to send alerts via phone or fax, by using the spool functionnality (writing files in /var/spool/asterisk/outgoing). The system works, but sometimes a file in the outgoing directory is simply ignored by asterisk (nothing on the cli); the spool file is correct. It happenned only twice since the system is installed (about two months), but since it is an alert

Hi, my etc/pam.d/login file is given at the end. i am using winbind and single sign on is working now. on the fly directory creation also works. when i check the /var/log/messages i have this error in there 1.Jun 23 05:40:46 niit158VM pam_winbind[1696]: user 'linwin/zubair'granted acces 2.Jun 23 05:40:46 niit158VM pam_winbind[1696]: user 'linwin/zubair' granted acces

Here's an easy one for y'all. I am running samba V1.9.16p1 on an NT 4.0 vs. Solrais 2.51 peer to peer, and so far so good. !BUT! I am about to go to Solaris 2.6. Is there anything different that must be done for samba on this newer Solaris version? thanks- john "...you can't stop progress, but you can help redefine it." -anonymous "angels hang their socks on

Samba, We've been using samba-1.9.16p9 here for a long time, with a tremendous record of success. I've recently compiled and installed 2.0.0, but am having problems. Only about half my users can login now, and who's singled out is *seemingly* random. This is on a DEC Unix 3.2 (osf/1) machine. Threw the old binaries back on, and everyone's happy again. What can I do to get

Qualify=yes will send a SIP OPTIONS periodically and keep the NAT open, if you use 1 to 1 NAT (versus PAT where it is "many to one NAT") it will work because port 5060 on the private address will still be port 5060 on the public address. With PAT the port could be anything over 1024, but usually much higher, and the originator will send to port 5060, which your NAT router will drop.

I'm running: Asterisk CVS-HEAD-07/06/04-17:49:49 built by root@gf-002-pbx-001 on a i686 running Linux I've tried placing files (both ending in .call and not) in the correct format in /var/spool/asterisk/outgoing.... I get _nothing_. No log messages, nothing on the console, zip. Permissions seem to be correct on both the files and the directory, as well (* is running as root, for right

Hi list, I have been trying to get the new file extension for Ogg FLAC (.oga) to work with Amarok. But it does not look to be possible due to limitations in the xine and gstreamer engines. Maybe the move to .oga was prematurely? I have done some testing with other players?including Banshee, Quod Libet, and others?but have not had any success with playing the .oga files. Only player I have