similar to: Call Drops while doing assisted transfer from remote location

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 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: 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

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

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, 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 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

Hi all, I'm having some problems getting TDMoE setup for the 1st time. I have a TE405P installed in the main server with an ethernet cross-connection to the secondary machine. (Yes, I know about IAX2 but I want to use TDMoE to simulate using T1s.) I'm using -HEAD from yesterday. On the main machine /etc/zaptel.conf: loadzone = us defaultzone=us

Guys, I'm trying to configure a TE406P with MFC/R2. here goes my zaptel.conf: span=1,0,0,ccs,hdb3,crc4 cas=1-15:1101 dchan=16 cas=17-31:1101 span=2,0,0,ccs,hdb3,crc4 cas=32-46:1101 dchan=47 cas=48-62:1101 The first strange behavior is that the: zap show status shows this: Description Alarms IRQ bpviol CRC4 T4XXP (PCI) Card 0 Span 1

Hi anyone have a idea for debug my digium TE405P card ? My zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = fr defaultzone = fr My Zapata.conf: [channels] language=fr context=from-E1 switchtype = euroisdn pridialplan = unknown signalling = pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes

Dear, I always had one E1 card with one span, so I've never had any problem in set it up through /etc/dahdi/sustem.conf and /etc/asterisk/chan_dahdi.conf because I put span=1. But now I have a PBX with two E1 cards with 4 span (8 span in total). How do I have to define both card in system.conf and chan_dahdi.conf, and how do I have to refer each span to the corresponding card ??? Thanks a

I have a box with 12 T1s (4 Te410P cards). The PSTN provider is reporting slips and ask me to update the clock source. I have my system.conf set as the following but when I run dahdi_scan only the ports on Card 1 are showing up with syncsrc=1 system.conf : span=1,1,0,esf,b8zs bchan=2-24 mtp2=1 span=2,2,0,esf,b8zs bchan=26-48 mtp2=25 span=3,3,0,esf,b8zs bchan=49-72

We are using AAH with Asterisk 1.2.7.1 with a TE405P as listed below. We are getting frequent restarts on the spans which lead to dropped calls. I have pasted some hopefully pertinent information below -- anyone have any clues that might help? Thanks Next line is repeated throughout messages, going through every channel in every connected span. asterisk/full.1:Apr 24 01:15:25 VERBOSE[4196]

Hi, I have a digium card (igium, Inc. Wildcard TE405P quad-span T1/E1/J1 card 5.0V (rev 02)) 4 ports I want to make a loop test between digium card E1 to test the configuration of dahdi What I want to do scenario is I connect port 1 and port4 in the digium card with E1 cable SIPcall-->E1 Digium port 1--->(Loop)E1 port 2---->sip extension local. kindly can any can help me to

Hi All, I am facing problem with E1 line. I have installed Asterisk (1.4.20.1) on a system with Digium TE420 card (Zaptel- 1.4.10) But every now and then I face problem of down E1's. The log show lot of entries like "pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 2" This happens on a regular basis and the E1 becomes up after some time. My

Hi, Running 1.4.26.1 here. I have installed TE420B card in my server, and followed the appropriate steps (as far as I know to configure it). This TE420B is connected to a CLEC (T1s), so I am using pri_cpe as singalling type. When I dial out, I get this message: Dec 4 11:37:31] WARNING[27983]: app_dial.c:1275 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 0