search for: app_chanspi

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]

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

asterisk 1.4.32 have zapata.conf soft link to chan_dahdi.conf to use flash operator panel < 2.0 (from extensions.conf) exten=> 304,1,ChanSpy(Zap/4|q) exten=> 304,2,hangup There is no entry ChanSpy(Zap/41) in extensions.conf 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",

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

Hi all, I 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

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.

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

hi, exten =>6000,1,dial(SIP/6000,15,tr) exten =>6002,1,dial(SIP/6002,15,tr) exten =>6004,1,dial(SIP/6004,15,tr) exten =>6006,1,dial(SIP/6006,15,tr) exten =>6008,1,chanspy(SIP/6006 | wbq) when i dial 6008 ,it is connected ,but i can't able to hear the voice of the any one. when coversation between the 6002 to 6006. in my Console mode i got the following comment *CLI>

Hi all, according to http://sip.fontventa.com/content/view/15/44/ i have compiled the mpeg4ip libries without problem. After copying the app_mp4.c file 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} $<

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:

Hi list, I'm having some strange problems since I updated to CVS HEAD three hours ago... First: I was using Iax Comm in some PCs, it suddenly stopped working, what I get is som pieces of audio once in a while, I mean instead of listening to the ring tone and then the voice on the other side I just hear a bit of the ring tone, maybe another bit, a bit of someone answering... like

Hi, I am facing a problem where for legal obligations (LI) I have to copy/mirror/forward the RTP streams for some selected call to an external address/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

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

The Asterisk Development Team has announced the release of Asterisk 1.6.0.26. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.0.26 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by

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 attachment was scrubbed... URL:

Hi, exten => 6000,1,Dial(SIP/6000,15,tr) exten => 6002,1,Dial(SIP/6002,15,tr) exten => 6004,1,Dial(SIP/6004,15,tr) exten => 6006,1,Dial(SIP/6006,15,tr) exten => 6008,1,Chanspy(SIP/6006 |wbq) when i dial 6008 ,it is connected ,but i can't able to hear the voice of the any one. when coversation between the 6002 to 6006. in my Console mode i got the following comment *CLI>

* * hi, I have some qustions ... first one is ..chanspy in asterisk 1.4 version really work with whisper mode... and second is i got following error,when i using chanspy in 1.4. version.. when i dial 6008 ,it is connected ,but i can't able to hear the voice of the any one. when coversation between the 6002 to 6006. in my Console mode i got the following comment *CLI> -- Executing

I would like to be able to stop the chanspy application and go to the next step in the dialplan but I do not see a way to do that. I have looked at the code and I do not see a way to stop the chanspy application. Even if there are no channels that match the chanprefix pattern the chanspy application is not exited. Hitting the * key stops spying on a channel but then starts spying on the same

The Asterisk Development Team has announced the release of Asterisk 1.6.0.26. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.0.26 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by