similar to: Placing call files in

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

Hello, With an extensions.ael enabled system, I keep getting whatever I change into my "astup.call" file : [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:267 apply_outgoing: At least one of app or extension (or keyword message/pdu) must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/astup.call [Mar 12 00:13:56] WARNING[2538]: pbx_spool.c:457 scan_service:

Hi, Is the following behaviour a bug or a feature ? Using bristuff-0.4.0-RC3d.tar.gz, the call file thereafter produces : [Mar 16 15:39:36] WARNING[25547]: pbx_spool.c:267 apply_outgoing: At least one of app or extension (or keyword message/pdu) must be specified, along with tech and dest in file /var/spool/asterisk/outgoing/astup.call [Mar 16 15:39:36] WARNING[25547]: pbx_spool.c:457

Hi list, To make outgoing calls by skype i would like to have our crm app create callfiles like we do for normal calls. If i read the instructions it says this : ---quote--- The syntax for making an outgoing call using Skype for Asterisk is as follows: Dial(Skype/[<originator>@]<destination>) ---unquote--- So i create a callfile that looks like this: --- Channel: SIP/228

I am using outgoing call files. I typically see the "ooh something changed / timeout" on a regular bases every second to be exact. Then it stops until some other call event happens. So I "mv" my call file to the outgoing spool directory, I am listening to that message, another call file is "mv"'ed into the directory and something happens to the timeout that its

Steve, I am trying to get tx_fax to work. I am using a TDM2401E card. I have a 3 page fax and I only receive the first page on every attempt. I think I have enabled debug output below. Can you tell me what the problem might be? I am using snapshot from oct 26. asterisk 1.2.13 and libtiff 3.6.1-12 from redat/centos 4.4. THanks, Jerry --------- Oct 28 13:13:40 DEBUG[22763]: app_txfax.c:69

Has anyone tried to (recently) install asterisk in a location not relative to /, as a non-root user? Ie editting the PREFIX directive in Makefile. Why? Several quite obvious reasons: a). Allows an asterisk user to be created, and operators to log into the box as asterisk user, without having root access. b). Much easier backups, because everything is beneath the same directory structure. c).

Has anyone tried to (recently) install asterisk in a location not relative to /, as a non-root user? Ie editting the PREFIX directive in Makefile. Why? Several quite obvious reasons: a). Allows an asterisk user to be created, and operators to log into the box as asterisk user, without having root access. b). Much easier backups, because everything is beneath the same directory structure.

I'm using Slackware 11. I unistalled the package that provides libtiff 3.8..... and installed the most current 3.7.... for lib tiff. I downloaded asterisk 1.4 beta3 and the 1.4 beta2 addons and untared them. created a simlink: ln -s asterisk-1.4.0-beta3 asterisk I've compiled spandsp from as follows cd /usr/src wget

Hello, I m using asterisk version 1.6.2.0 beta. I m trying to test load on it, for which i m using WINSIP installed at two computers and facing two problems. Problem 1: I got 100 users registered to asterisk from each winsip and then initiates 100 calls from one winsip other winsip. But the problem is approx of 60 calls get mature and asterisk give error for the remaining like shown below.

HI I am trying to use the outcall going by the wiki.( http://www.voip-info.org/wiki-Asterisk+auto-dial+out) But I keep getting the errors below. Here is a sample of a callout file. What am I doing wrong? ////Begin Outgoing.call//// Channel: sip/2075 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: managers Extension: 2184 Priority: 1 ////End outgoing.call//// Nov 9 20:32:02

From: Greg Knaddison <greg.knaddison at gmail.com> > Look - brevity is the soul of wit. It also keeps other folks on lists > from calling you a troll. The problem is that then the same people then start complaining about my "oversimplification." In fact, that's part of the problem here. I really _tried_ to make a recommendation without breaking down into the

I am using Realtime extensions as well, in case that would matter. Following problem arises from time to time, a call will successfully terminate: [May 14 14:31:41] VERBOSE[3274] pbx_realtime.c: -- Executing [t at project_init:1] Hangup("SIP/peer-2-00002f7e", "") [May 14 14:31:41] VERBOSE[3274] pbx.c: == Spawn extension (project_init, t, 1) exited non-zero on

Greetings!! We're getting some strange results using call files.. no matter the technology, DAHDI, SIP, etc., we get a "Call failed to go through, reason (3) Remote end Ringing" message when attempting to originate a call from a call file. Numbers changed to protect the innocent.... using call file.... //------------CALL FILE------------// Channel: DAHDI/g1/918005551212

Does anybody have a work around for this problem. I create a call file in /var/spool/asterisk/outgoing and Asterisk picks it up and starts placing the call. However if the called channel provides any sort of progress indication (such as a SIP or IAX channel indicating ringing that causes the console to say "SIP/xxxx is ringing") the code in pbx_spool.c indicates a call failure and

Hi list, It's been a while since I've done asterisk stuff, and I'm wondering if there any news in the field. What do you people use today for http management of debian based Asterisk setup? Preferably something with the proven ".deb" extension. Any recommendations are welcome. Thank you, Maxim. -- Cheers, Maxim Vexler "Free as in Freedom" - Do u GNU ?

Hi, After only 4 days I have 107472352 bytes in 46007 allocations in file 'pbx_spool.c' asterisk*CLI> show memory summary 180 bytes in 2 allocations in file 'netsock.c' 12 bytes in 1 allocations in file 'devicestate.c' 2268 bytes in 1 allocations in file 'jitterbuf.c' 8160 bytes in 1 allocations in file

Hello I am experiencing a freeze of the Asterisk proces when issuing a 'sip reload'. I have this issue every time on asterisk versions : 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3. I do not have this on versions certified-13.8-cert2, certified-13.8-cert1 and asterisk 1.8.32.3. The only solution is a cold restart of Asterisk. I can execute any command on CLI except 'sip

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 Snom870 Handsets We are in the process of moving to an Asterisk based PBX. At the moment most things work as we wish. However, I have just notices that when I force a reload using 'amportal a reload' I see this loop start in 'asterisk -rvvvvvvvvvv': > Channel Local/s at tc-maint-000002a4;1

Hi - after hearing others rave about * I thought I'd have a go - extract from a 'make' on a stock debian system as follows... (I tried to post the whole make up to this point but it was too big for the list) make[1]: Leaving directory `/usr/src/asterisk/channels' make[1]: Entering directory `/usr/src/asterisk/pbx' gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes