search for: app_chanspy

Hello List! My Asterisk stopped making SIP-calls today, I could call from external, and saw Call coming in over PRI, but calling the SIP/Device wont work. I saw 5 open channels - all chan_spy. Only a restart helped. In the messages-file i found from yesterday: [Mar 4 17:28:01] NOTICE[25769] app_chanspy.c: Attaching SIP/209-0000170fto SIP/210-0000170e [Mar 4 17:29:38] NOTICE[25790] app_chanspy.c: Attaching SIP/209-00001715 to SIP/210-00001714 [Mar 4 17:31:06] NOTICE[25812] app_chanspy.c: Attaching SIP/209-0000171c to SIP/210-0000171a [Mar 4 17:31:19] ERROR[25818] acl.c: Cannot create socket [Ma...

Hello I have an old Asterisk where I need to listen to Agent calls. So I created this code: exten => _555,1,ChanSpy(Agent) exten => _555,n,Hangup() But I always get: 2009-10-22 16:00:38 WARNING[5695]: pbx.c:1720 pbx_extension_helper: No application 'ChanSpy' for extension (default, 555, 1) It seems that Asterisk doesn't have ChanSpy enabled... is this possible? Which

...On dialing 304 and Zap/41 is in use this happens: [Jul 1 18:24:47] VERBOSE[14447] logger.c: -- Executing [304 at flash:1] ChanSpy("Zap/31-1", "Zap/4|q") in new stack [Jul 1 18:24:47] VERBOSE[14447] logger.c: == Spying on channel Zap/41-1 [Jul 1 18:24:47] NOTICE[14447] app_chanspy.c: Attaching Zap/31-1 to Zap/41-1 If while spying on Zap/41 that channel is hung up: [Jul 1 19:06:48] VERBOSE[15242] logger.c: == Done Spying on channel Zap/41-1 [Jul 1 19:06:48] VERBOSE[15242] logger.c: == Spying on channel Zap/4-1 [Jul 1 19:06:48] NOTICE[15242] app_chanspy.c: Attaching Za...

Hi folks, We have a very rare problem with Asterisk 1.2 where Chanspy reports the following: Oct 31 19:53:29 NOTICE[10490] app_chanspy.c: Attaching SIP/105-48807010 to SIP/2015-11b21148 Oct 31 19:53:42 NOTICE[10490] app_chanspy.c: Attaching SIP/105-48807010 to SIP/2066-11d42b80 Oct 31 19:53:46 NOTICE[10490] app_chanspy.c: Attaching SIP/105-48807010 to SIP/2022-11c10170 Oct 31 19:53:58 NOTICE[10982] rtp.c: Unknown RTP codec 1...

...read on serveral pages that chan_spy is not part of asterisk anymore as on http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy on the bottom of this page. I have a testing server with debian-testing and debian packages for asterisk installed. In the modules directory /usr/lib/asterisk/modules is a app_chanspy.so already there. The currently installed version is 1.4.13. So, it is usefull to use app_chanspy.so for an asterisk for a callcenter? Which application does now the same for recording/spiing channels from a third person? Many thanks in advance. Jan -- Jan-Hendrik Palic TNG - NETWORK MANAGEME...

Hi All, I have installed ChanSpy but its not compiling gives me the following error....if anyone can help that will be really appreciated... [root@Asterisk asterisk-1.0.9]# contrib/scripts/astxs -install -autoload apps/app_chanspy.c make: *** No rule to make target `apps_env'. Stop. -c apps/app_chanspy.c -o apps/app_chanspy.o Thanks Umair. _________________________________________________________________ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/...

I have read about that on asterisk 1.6, there will be a parameter "o" (Only listen to audio coming from this channel), I have tried, but I still get inbound and outbound audio from the spied channel. Has anyone used this feature? Is it working? Is there any work-around? I will like to only spy the outbound audio from a channel, I dont want to hear the incomming audio of that channel. I

Everyone we have a few new things to give back to the asterisk community. http://bugs.digium.com/bug_view_page.php?bug_id=0002379 http://bugs.digium.com/bug_view_page.php?bug_id=0002380 http://bugs.digium.com/bug_view_page.php?bug_id=0002381 These include app_chanspy, the ability to spy on ANY bridged call taking place inside asterisk. NOT just ZAP as with ZapScan/Barge. Native format_* files being used for moh. Reload enabled res_musiconhold. format_mp3.c that produces SLNR output to asterisk, format_slinear.c for raw headerless audio, format_base65_wav_...

...|tr") in new stack -- Called 6006 -- SIP/6006-08594188 is ringing -- SIP/6006-08594188 answered SIP/6002-08590248 -- Executing [6008@from-sip:1] *ChanSpy*("SIP/6004-08589040", "SIP/6006|wq") in new stack == Spying on channel SIP/6006-08594188 [Nov 11 15:19:37] NOTICE[8974]: app_chanspy.c:202 start_spying: Attaching SIP/6004-08589040 to SIP/6006-08594188 Segmentation fault (core dumped) linux:~ # please reply reg . this issue. Regards, Thiru -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachment...

...into de Asterisk apps directory and changing the Makefile like. [...] app_sql_odbc.so: app_sql_odbc.o $(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} -lodbc app_mp4.so : app_mp4.o $(CC) $(SOLINK) -o $@ ${CYGSOLINK} $< ${CYGSOLIB} -lmp4 -lmp4v2 ifeq (SunOS,$(shell uname)) app_chanspy.so: app_chanspy.o $(CC) $(SOLINK) -o $@ $< -lrt endif [...] i get following error. Mar 21 19:08:22 WARNING[26686]: pbx.c:1720 pbx_extension_helper: No application 'mp4save' for extension... it seems that after the recompilation of asterisk no app_mp4.o/app_mp4.so is created in...

Hello All: I saw there are app_fax and app_chanspy modules in 1.6.2.6, but there is NO sample configure file for them. Is anybody know how to use them, or where is the documentation for them? Thanks -- Refer to: http://www.microsuncn.com Best Regards Alan Zheng -------------- next part -------------- An HTML attachment was scrubbed... URL: htt...

...seconds) xxxx-in.wav file.... and last but not least when I try to ChanSpy a SIP channel (at last that's what I moved to the last CVS Head for) it makes * crash.... see what goes on (I was attached to an asterisk running using "asterisk -vvvvvvvvvvvr") Mar 27 23:12:41 WARNING[5631]: app_chanspy.c:280 start_spying: Attaching SIP/matt2-05b0 to SIP/matt-967b Godzilla*CLI> Disconnected from Asterisk server Executing last minute cleanups Godzilla:/etc/asterisk# I'm using Asterisk CVS-HEAD-03/27/05-17:22:56 currently running on Godzilla (pid = 5700) thanks a lot. M.

...s/port and I have not found a way to do it with built-in functionality. Do I miss something? The basic requirements are: * Raw RTP (no transcoding, header and payload as is) * Direction (did it arrive at asterisk or was it sent) * End indication once the call has been cleared I tried to use app_chanspy and ran into three different problems: - It is always transcoding - No support for input only on the audiohooks used by app_chanspy (send input, output to two different dest ports) - End of call only through hangup A post mentioned the framehooks but from what I see: - It is the audio pay...

I am still trying to figure out a reasonable way to exit the chanspy application in a dialplan. For the most part I understand how things are working and there is one change I would like to propose. The way the 1.4.23.1 code seems to work is that if there are no channels that match the chanprefix argument the chanspy code stays in a loop waiting for a new channel to come into being that matches

...ed, patched by jedi98. Tested by mattbrown, Chainsaw, mikeeccleston) * If the peer record is from realtime, it could be set to 0, due to MySQL not representing NULL well in integer columns. (Closes issue #16683. Reported by wdoekes) * Resolve a crash caused by a race condition in app_chanspy.c (Closes issue #16678. Reported, patched by tim_ringenbach. Tested by dvossel) * Fix deadlock in app_queue with use_weight during reload. (Closes issue #16677. Reported, patched by tim_ringenbach) * Stop playing the message number multiple times in app_voicemail. Also remove some...

Hi ALL, I have downloaded app_chanspy.c and chanspy_sounds.tgz. But I haven't found any instructions on how to compile and where to untar these files... I tried to put the .c file on <asterisk-src>/apps and remake asterisk, but it seems it was not enough... Thank you! Dov -------------- next part -------------- An HTML atta...

...15|tr") in new stack -- Called 6006 -- SIP/6006-08594188 is ringing -- SIP/6006-08594188 answered SIP/6002-08590248 -- Executing [6008@from-sip:1] ChanSpy("SIP/6004-08589040", "SIP/6006|wq") in new stack == Spying on channel SIP/6006-08594188 [Nov 11 15:19:37] NOTICE[8974]: app_chanspy.c:202 start_spying: Attaching SIP/6004-08589040 to SIP/6006-08594188 Segmentation fault (core dumped) linux:~ # please reply reg . this issue. Regards, Thiru -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachment...

...15|tr") in new stack -- Called 6006 -- SIP/6006-08594188 is ringing -- SIP/6006-08594188 answered SIP/6002-08590248 -- Executing [6008@from-sip:1] ChanSpy("SIP/6004-08589040", "SIP/6006|wq") in new stack == Spying on channel SIP/6006-08594188 [Nov 11 15:19:37] NOTICE[8974]: app_chanspy.c:202 start_spying: Attaching SIP/6004-08589040 to SIP/6006-08594188 Segmentation fault (core dumped) linux:~ # please reply reg . this issue. Regards, Thiru -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachment...

...DoChanMon,n,ChanSpy(${CfMC_WhoHear},q) exten => DoChanMon,n,Hangup() and I would like to be able to make the agent available again without them having to hang up and log back in again. If I add a line of code like this: if (running == 99999) running = -1; in app_chanspy.c around line 348: if (!ast_strlen_zero(inp)) { running = atoi(inp); if (running == 99999) running = -1; break; Then hitting 99999# on the keypad does what I want. I do not par...

...ed, patched by jedi98. Tested by mattbrown, Chainsaw, mikeeccleston) * If the peer record is from realtime, it could be set to 0, due to MySQL not representing NULL well in integer columns. (Closes issue #16683. Reported by wdoekes) * Resolve a crash caused by a race condition in app_chanspy.c (Closes issue #16678. Reported, patched by tim_ringenbach. Tested by dvossel) * Fix deadlock in app_queue with use_weight during reload. (Closes issue #16677. Reported, patched by tim_ringenbach) * Stop playing the message number multiple times in app_voicemail. Also remove some...