similar to: Example: Writing a click-to-call application using pbx_spool

Displaying 20 results from an estimated 2000 matches similar to: "Example: Writing a click-to-call application using pbx_spool"

2009 Feb 03
2
some kind of timeout problem in pbx_spool.c
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
2006 Jun 06
0
pbx_spool - outgoing qcall failure upon call progress
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
2005 Oct 15
2
What would cause a high memory usage in pbx_spool.c ?
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
2003 Oct 08
2
pbx_spool and contexts
When I drop my file into the outgoing folder, the call is completed but the 'Context' entry is not respected. Instead, it drops into the default context. It does drop "properly" into the default context and function as would be expected. I looked through the source but didn't see any reason it would be completely ignoring the context. Call file: (where
2004 Oct 13
0
Song Updates
My bad, try curl -s --user etc etc etc... I suppose that'd help. :] On 13 Oct 2004 at 13:02, Dave St John wrote: > > Here is the error i get now. > > [root@mc1 gork]# curl --useruser:pass > 'http://64.62.252.140:9120/admin/metadata?mount=/live.ogg&mode=updinfo&song=test_name+te > st_Title' > <b>source will not accept URL updates</b> >
2003 Jun 27
2
Working: TFTPd for NAT'd Cisco 7960 and ATA-186
For anyone who is interested, I have a working tftpd (modified wvtftpd) capable of serving configuration, dialplans, and ringtones to Cisco 7960/7940 and ATA-186 devices that are located behind NAT firewalls. As TFTP is not a very firewall/NAT friendly protocol, I had to break some rules to get it to work with these cisco devices. It might cause problems for other TFTP clients, but it works with
2004 Oct 14
2
Song Updates
Let this be a note for everyonein the future then. When using mp3s i don't think there is a --enable-mp3-updates (I think it worked by default for me), yet if using ogg vorbis you must compile icecast2 with --enable-vorbis-updates in order to update the song/title id3 metadeta (Instructions on how to do so with `curl' are below). If I was using ogg I may have been of more use, yet I
2004 Oct 02
0
Song Updates
You did not type the exact string I told you to first of all. You forgot to use the 's after before the h in http and at the end. curl --user admin:hackme 'http://192.168.0.1:8000/admin/metadata?mount=/live&mode=updinfo&song=Artist_name+Title' notice the useage of the 's in my string and not in yours. Your string is launching nonsense into the background via the two
2013 Nov 28
4
puppet in java
We have bunch or property files(key/value pairs) used in different modules in our java web applicaiton. our applicaiton is also distributed, part of that runs on a head office and some of the parts run at the branch. All the branches run a local server for day to day activities. We are looking to automate these files when moving to different environments like deve, test, prod. So that we can
2004 Nov 09
2
Auto dial Out
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
2014 May 15
1
Call file problem, DelayedRetry/retrying spite MaxRetries: 0
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
2011 Jun 15
1
call file challenge...
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
2009 Mar 16
1
Bristuff bug or feature ? (Was: Are .call files working with extensions.ael ? bristuff problem)
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
2009 Sep 02
1
Skype for Asterisk callfile question
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
2013 Jun 24
2
Asterisk-11 loop behaviour
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
2003 Jul 23
1
Newbie Help
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
2009 Mar 11
4
Are .call files working with extensions.ael ?
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:
2005 Aug 23
1
latest CVS on Mandrake 9.2 Mini ITX
All, wondering if you can help, I had a perfectly working Mandrake 9.2 box running on a via Mini ITX 5000/classic. Asterisk (zaptel and libpri) was built from CVS head around 22nd July 2005. I decided now was a good time to ghost it up....although humorous for you all suffice to say, I ended up with a partitionless box! Following the install script I had created I built an identical server to
2020 Apr 23
0
/outgoing/ .call files and RetryTime problem
asterisk-16.8.0 Hi I've set up a callback script to retry a number if it's busy, but as I watch the console output asterisk seems to rush 3 or 4 calls at once before waiting the RetryTime of 20 seconds that I've set. The script: -----8<------ CALLERID=$1 EXTENSION=$2 TEMP=`mktemp /tmp/call-XXXXXX`.call cat <<EOF > $TEMP Channel: IAX2/account at
2006 May 24
1
Placing call files in /var/spool/asterisk/outgoing/ does not work
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