similar to: ChanSpy in Asterisk 1.2.24

Hello I'm trying to use an AGI that returns the queues status (numbers of available agents, etc ), but I'm having some problems with it (it's still very buggy). Is there any AGI repository with source code samples? Had anyone used an AGI to check queues and agents status? Thanks regards Joao Pereira -- StarTel - A Rede Livre Joao Gomes Pereira www.startel.pt +351 304500650 sip:

Hello I'm trying to register an Asterisk working behind Nat. Here is the trunk: register=username:password at sip.startel.pt [startel] type=peer host=sip.startel.pt username=username fromuser=username secret=password qualify=yes disallow=all allow=ulaw allow=alaw allow=gsm insecure=very port=5060 nat=yes canreinvite=yes The problem is: Asterisk is registering with its

Hello Im using Asterisk as a SIP client of Kamailio with RTPproxy. Asterisk is behind NAT. X-Lite and SNOM phones behind NAT work fine. But when I try to connect with an Asterisk behind NAT, the Asterisk client doesn't receive sound. I already tried in 2 different NATs, with no firewalls. This is my Asterisk config: [kamailio] type=peer host=xxx.xxx.xxx.xxx disallow=all allow=ulaw

Hello to all I'm using asterisk 1.4 and dahdi. I had everything working fine, and I could place calls through my R2 channel. But now the channel is always "RED" and Im getting this error message: TE4XXP: Version Synchronization Error! Here is my chan_dahdi.conf------------------------------ [channels] language=en context=incomingr2 signalling=mfcr2 mfcr2_variant=ar

Hello: I have the linux version 2.0 of x-lite downloaded. Does anyone know exactly what settings needed to reach the asterisk server on my home network? Internet ->DSL transparent bridge ->router ->asterisk ->softphone x-lite attempts to login and register, but times out. There must be some setting I'm

Hello to all I'm trying to record the calls going to my queues, but asterisk creates 2 files, one with the inbound and another with the outbound sound. I know Sox should mix the 2 files automatically in the end, but this isn't happening. I have sox installed in my server. How can I force Sox to mix the files? Here is my config: queues.conf----------------------------- [general]

I wonder if Asterisk's skinny/sccp channel driver could be used as a "client" to register with a Cisco PBX. That is, along with a SIP client, say, have Asterisk and said SIP client stand in for a Cisco phone, or an IP Communicator. Anyone done this? Cheers, b. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

Hey gang, I'm trying to find a) If you can put channel variables into a Call file and b) what the appropriate syntax is. Any ideas? Thanks, PB -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090721/cb8c2656/attachment.htm

hi, i have a new install, 1.6 2, 2 extension, but the call doesnt get thorugh: here is my sip debug outout: thx for ur help!! <asterisk-users at lists.digium.com> --- (13 headers 16 lines) --- Sending to AA.BBB.CCC.DD : 28127 (NAT) Using INVITE request as basis request - Y2QxNTg4NjE3MTZjNGMzZGM5NzE3YWY4NjAyOTYzMjk. Found user '701' for '701' Found RTP audio format 107 Found

Hi all, We have our system hosted publicly and 4 phones are connected remotely at employee's home, and when they try to do a assisted transfer to one of the employee at the main office, the call is lost. For ex: person A calls person B, person B calls person C for assisted transfer, and as soon as person B hits transfer button again to transfer person A to C, the call is lost. But in the

Hello I just updated the kernel, dahdi-linux and dahdi-tools Im also using now asterisk 1.4.26.1 And im still with a red light (not RED/YELLOW anymore): [root at catumbela ~]# /etc/rc.d/init.d/dahdi status ### Span 1: TE4/0/1 "T4XXP (PCI) Card 0 Span 1" (MASTER) HDB3/ RED 1 PRI CAS RED 2 PRI CAS RED 3 PRI CAS RED 4 PRI

Hello I have a simple configuration to allow the admins to listen the agents calls: exten => _654,1,ChanSpy(Agent) exten => _654,2,Hangup() The problem is... even when the agents hung up... it seems the channels remain active: asterisk*CLI> show channels SIP/211-b3042018 654 at default:1 Up ChanSpy(Agent) SIP/211-b3fbf768 654 at default:1 Up ChanSpy(Agent)

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",

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

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 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 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

hello list, i use chanspy with the code below [app-chanspy] exten => _007.,1,Macro(user-callerid,) exten => _007.,n,Answer exten => _007.,n,Authenticate(1111) exten => _007.,n,ChanSpy(SIP/${EXTEN:3},dqs) exten => _007.,n,Hangup i have a question related to chanspy i have created extension from 100 to 300 and i will give the permission with group of extension i want to use