similar to: Asterisk as a skinny/sccp "client"?

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091111/2b828eff/attachment.htm -------------- next part -------------- Hello. I am trying to execute an fax reception script and i am getting the following: [Nov 11 08:40:52] WARNING[12800]: app_system.c:88 system_exec_helper: Unable to execute '/var/lib/asterisk/scripts/mailfax ""

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

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 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 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: 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 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

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

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]

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 all, Can this be done ? Would setting the variable FAXRESOLUTION to a appropriate value affect this change ? > http://www.asteriskguru.com/tutorials/rxfax.html Variables connected with the application LOCALSTATIONID - used by to application to identify itself to the remote end LOCALHEADERINFO - used to generate a header line on each page REMOTESTATIONID - set by the application, the

Below is my extensions.conf for the fax part [incoming_28345474] ; ;******************************************************************** ; BEGIN - Inbound call handlers ;******************************************************************** ; exten => 8862100,1,NoOp(${CALLERID(num)}) exten => 8862100,2,Background(if-u-know-ext-dial) exten =>

I have this in my extension.conf: [incoming_28345474] ; 8862100 is the hotline number of the Welltech 3804 ; exten => 8862100,1,NoOp(${CALLERID(num)}) exten => 8862100,2,Wait(1) exten => 8862100,3,Set(CALLERID(num)=${CALLERID(num)}) include => fax2emailstart [fax2emailstart] exten => 3000,1,SetVar(CALLEDFAX=${EXTEN}) ; me exten => 3000,2,Answer exten =>

Hello All, I am looking to receive faxes via my inbound SIP, but I can get it to answer the fax. Now I did test the SIP inbound to a phone and that does work it is just the fax part I am having issues with any help would be great. This is the error I am getting Mar 11 15:08:49 NOTICE[14322]: rtp.c:430 ast_rtp_read: RTP: Received packet with bad UDP checksum Mar 11 15:08:49 WARNING[14322]:

Hello All, I just started to use asterisk with Digium card (4 fxo ports) and I've met some problems ( I'm just new in asterisk so questions may be stupid ) my environment: Debian testing, asterisk 1.0.9 zaptel-1.0.9 TDM04P 1) when asterisk receiving incoming call on TDM card all networking cards stops to send or receive any data for some time nothing suspicious in the log files,

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)

Well, my faxes passes through asterisk successfully, however I still have some problems about fax reception by rxfax. The softfax answers, and negotiates transmission, however then as some stage of communiation something is wrong. But I have nothing more but this log: Jun 2 00:10:21 DEBUG[16900]: chan_zap.c:4242 zt_read: DTMF digit: * on Zap/10-1 Jun 2 00:10:22 DEBUG[16900]: chan_zap.c:4242

Hi all, My BRIstuffed 0.2.0-RC7k is running fine on my debian box for voice calls over ISDN2. Now I want to implement receiving incoming faxes into my setup so I did a google and some reading on the wiki. I got the spandsp 0.0.2pre10 package compiled and installed, patched asterisk's apps makefile and compiled * again. This all worked out fine. When integrating the RxFax into my dialplan the