similar to: T1 PRI with groups

Hello, Is there any way to force more ringing time on incoming lines? I have heard of the route of answering and then playing a ring tone for a set amount of time, but is there any way to actually have the line ring from the outside for a set amount of time within the dialplan? I thought this would work, but no luck exten => 9999,1,Ringing(5) exten => 9999,2,Answer *CLI> show

Hi, the option c for the announce of the user count does not work me in * 1.0.9. exten => 9999,1,Wait(1) exten => 9999,2,MeetMe(|Mdcs) And how to handel the marked mode with option A? I can't find any sample config for this. Regards

Hi all, I use the G option in my dials for redirect both parties in the conference. There is a way for auto-include in a conference other parties that first two without using AGI? I try with: [from-internal] exten => 9999,1,Dial(IAX2/DIP02/9999||G(fromiax^9999^1) [fromiax] exten => 9999,1,MeetMe(9999,qdxAa) exten => 9999,2,MeetMe(9999,qdx) exten =>

Dear all I am going to setup Asterisk Call center solution and i have setup my queue and agent i have 2 SNOM ip phone but when i call to queue my agent phone is rining without musicnhold or when both phone is busy then i call to queue its directy hangup without musicnhole means my call not goes in to queue what is the problem my queue.conf [root at pbx asterisk]# cat

Hi I have a situation where I want to be able to count how many times a caller goes round a loop of "Please hold...", "please continue to hold". I have found an example on voip-info but I can't get it to work. Not sure if I've got some syntax wrong somewhere? All that happens at the moment, is I hit is the playback of "som-debug" at 9999. Any ideas would

Here's a possible bug, or more likely, I'm just missing something. We have a pots card in one of our asterisk boxes. Its a simple asterisk setup with one FXO/FXS card and basic static extensions file, etc. When we dial out over the pots line, 4 out of 5 times, it will work. However, every 4 or 5 times, we get an error back from the provider that says "The number you have dialed.....

I have a Cisco 2600 which understands the "gsmfr" codec, which appears to be what Asterisk calls "gsm" -- at least it ends up using it. I also have a PSTN gateway which is speaking ulaw. When the 2600 calls through Asterisk to the PSTN, it negotiates the g711ulaw codec, but when the PSTN calls through Asterisk to the 2600, it seems that Asterisk is doing translation, and it

I've got a system where I'm integrating a Nortel Option 11c with a Trixbox 2.0.0 system using a Sangoma A101 T1 card. (Running on a Dell PowerEdge 350) We've got things mostly up and running and all seems well... except... If I call from a SIP extension (X-lite soft phone) dialing 9xxxx where xxxx is an extension on the Opt 11, the call goes through to the Opt 11 but I have terrible

Hello everyone, We have recently turned up a new T1 from McLeod (Midwestern CLEC). It is configured like so: /etc/zaptel.conf: loadzone=us defaultzone=us span=1,0,0,esf,b8zs ;(also tried 1,1,0,esf,b8zs) bchan=13-23 nethdlc=1-12 dchan=24 /etc/zapata.conf: switchtype=national context=pri-in signalling=pri_cpe group=1 channel => 13-23 I can get hdlc0 (and pvc0) up just fine after the

I've posted this problem a couple of times before with little or no response. Basically I have a T100P in my * box. Incoming calls are working great. However outgoing calls are not working at all. I've copied a previous post into this message which should have all the necessary info. Any ideas or suggestions would be greatly appreciated. Thanks. Mark

I am trying to put a call on a E1 ISDN : The configuration are simple: zapata.conf : [channels] context=inbound switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes ;echocancel=no echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 ;immediate=yes immediate=no callerid => asreceived amaflags

Hi All, I have installed a Digium TE110P card on an Asterisk 1.2.1 system. Its connected directly to the PSTN. But I am unable to make outbound calls on the zap channels. The light on the card is green. Asterisk CLI shows all 24 channels when I give the command 'zap show channels'. I also noticed that Asterisk CLI shows an incoming call every few seconds on the 24th channel. This must be

HI, I'm trying to config a span from port 2 of a DIgium T410p to a Brooktrout TR114-P8V-T1 card. I have a T1-PRI from the TELCO in port 1 (thru first port) working just fine with Asterisk 1.0.3 - been working fine for some time now. No problem with dialplan PSTN calls. Now I'd like to "route" specific DID numbers in thru the TELCO T1-PRI and out thru the second port (span) on

Does anyone know how to monitor * to see if they are receiving timing slips on a span connected to a T100P card? I am seeing b-channel restarts quite often and also getting "No D-channels available" warnings from time to time. Yesterday I had all the b-channels crash during a MeetMe Conference. Not good! This PRI is connected to an Avaya Definity PBX that is onsite and located in the

What I''m doing: I have Shorewall on a SuSE 9.0 machine, which is the firewall/router on the network. External interface is eth0 172.16.1.1, internal interface is eth1 10.40.1.1. (I used the Two-interface Linux System Quickstart Guide). All works well with that configuration. I also use PPPD for dial-in clients, and have two modems for incoming calls. Recently I added VPN interface

I'm trying to get "blind transfer" from an incoming DAHDI line to an external number to work on an * 1.6 install using a T1 from XO. The documentation is very "distributed" and incomplete, so while it's not working, it's definitely more likely my error somehow. Couple questions if anybody is out there who even knows what TBCT is. 1) Is this even

root@graff:~# xm usb-list-assignable-devices 1-6 : ID 14dd:0002 Peppercon AG Multidevice 2-2 : ID 051d:0002 American Power Conversion Smart-UPS 3000 RM XL FW:691.16.I USB FW:7.3 root@graff:~# xm usb-hc-create bosch-w2k3-pv 1 1 root@graff:~# xm usb-list bosch-w2k3-pv Idx BE state usb-ver BE-path 0 0 1 USB1.1 /local/domain/0/backend/vusb/55/0

Folks, I'm trying to get Asterisk to load my voicemail configuration from MySQL. I've followed the instructions at: http://www.voip-info.org/wiki-Asterisk+voicemail+database I restarted Asterisk, but no luck: the voicemail.conf does not get updated. I started with a sample voicemail.conf that I found on the Wiki. Or was it from Voicepulse? I can't remember. For initial testing, I

Hey folks, I searched the list archives and found this : echo "AUTOFSCK_TIMEOUT=5" > /etc/sysconfig/autofsck echo "AUTOFSCK_DEF_CHECK=yes" >> /etc/sysconfig/autofsck http://lists.centos.org/pipermail/centos/2006-November/029837.html http://lists.centos.org/pipermail/centos/2009-September/thread.html#81934 Will this do all disks? I want to do a reboot of a couple

I'm having trouble setting the outbound caller ID on calls I make from my PRI trunk group. The PRIs are served out of a 5ESS. Telco has set the PRIs up for user provided caller id information, so I believe I just don't have it set up right in my dialplan or something. I can't seem to find an example of setting the outbound caller ID specifically for a 5ESS. Does anyone have an