similar to: App queue only + waiting call pickup

Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the

Ok. I just got a TDM10B and it is in with my X100P. So as it says in the provided instructions, I used the command modprobe tor2 I get an error message saying that there is no such device. My zaptel.conf looks like this: fxsks=1 fxoks=2 So I load the X100P first. (modprobe wcfxo) Then I load the TDM10B (modprobe tor2) Then I'm told that the device doesn't exist. Please help

Is this supported? I have a very simple setup where I have 2 X100P cards and a TDM10B. The TDM10B is connected to a phone that has a digital answering machine built into it. If I make an inbound call on either X100P interface it gets transferred to the TDM10B interface. If I let it ring the TDM10B interface answers the call and the greeting message of the answering machine starts. Then shortly

So I have an X100P, and a TDM10B both working (at least I think they are). The drivers have been loaded and ztcfg -vv shows no errors in the configuration of two channels. When I run asterisk -vvvc and pick up my phone (plugged into TDM10B), I don't gear a dialtone. in phone.conf, I have [interfaces] mode=dialtone format=slinear ... Shouldn't that produce a dialtone when I pick up the

Hello. I'm using asterisk 1.4.30. I've found this patch for app_queue.c : https://issues.asterisk.org/view.php?id=11700 Can I easily implement this by issuing : */wget 'https://issues.asterisk.org/file_download.php?file_id=17192&type=bug' -O - | patch -p0/* ?? Does this mean I have a "patched" asterisk ? (I ask this because some applications require a

Is there anything special needed to load up a TDM10B card?? I got the card today.. Took it from the box, put it into a PCI slot.. connected the power to the card and booted the PC.. I have removed the X100P to avoid confusion and I have the following in the config files.. in /etc/zaptel.conf # For the X100P #fxsks=1 # For the TDM10B fxoks=1 in /etc/asterisk/zapata.conf [channels]

Hello, when picking up an incoming call from one ip phone on another ip phone, the call terminates after about 5 to 10 seconds. When reading out the hangup cause variable in the h-extention of the dialplan, the hangup cause seems to be 111. In the dialplan output, you can see that SIP-peer sipacc3 picks up the incoming channel SipAgenT01-00001454, and the call is answered. After 7 seconds,

Ok. I am confused. I now have conflicting answers to my question: Do you need to use a special phone to use SIP? My setup is X100P and TDM10B. I would like to connect to iConnectHere, which uses SIP. Has anybody done this before (using similar equipment to what I have listed above)? And if it is not possible, could somebody please explain why. I don't understand why this wouldn't

I've been having trouble with a TDM10B that I installed a few days ago. At first I could not modprobe it at all (getting the errors below). I switched the X100P that was also installed (with a Motorola chip) with one with a Tiger-something chip, and then everything starts up okay. However, after a few hours, the FXS port dies (no battery or anything), and if I try to rmmod & modprobe

I want use iLBC and have following in mind, please help me is it possible ? ISDN <-----(ALAW)-----> * <-----(ALAW)-----> SNOM SIP??<-----(iLBC)----->?*?<-----(ALAW)----->?SNOM 1. ISDN incoming codec is ALAW SNOM codec should be ALAW (can't be iLBC because a lack of codec). 2. SIP incoming codec should be iLBC (snom is ALAW). 3. SIP outgoing codec should be iLBC /snom

I have an X100p card....and it is hard to hear the person on the other end. Should I mess with these values? I have heard both yes and no to this question in the past. If yes, how much louder should I make them? Thanks, MIchael

Hi: Did anyone use apt-get install zaptel successfully? After apt-get instal zaptel, use "modprobe zaptel", get a "FATAL modul zaptel not found". Thanks. Yan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040722/4608e1e5/attachment.htm

I'm having a problem defining my channels Here is my zapata.conf ----------------------------------------------------------------------------------------------------- ; Zapata telephony interface ; ; Configuration file [channels] ; rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes

I got the devkit tdm400P(tdm10b) + x100p bundle. Where does one buy more ports for the tdm400p? I couldnt find anywhere on digium site or resellers to buy them? -Dan

Hello all, New to the list, Asterisk and this wonderful world of convergence. From reading the docs, list and websites I hopefully correctly understand that I can use a plain cheap isdn4linux supported ISDN (passive) TA to hook up the PC to the ISDN line from the telco (FXO card?). Next in the PC I also need a FXS card like a Digium TDM10B or PhoneJack so I can hook up an analog phone to it. In

Hello, is there a way to disable call waiting in sip? I`m using grandstream 101 and even when the phone is in use I hear ringing in the headset. It is pretty annoying , is there some way to disable this? I cant find anything like it in the grandstream docs. Thanks -- Anton Yurchenko<phila@dg.net.ua> Digital Generation

First of all I would like to thank Mark for getting roundrobin to go roundrobin. Good job. Now we have some options here for leastrecent and fewestcalls strategy. It needs some work on the logic and Mark recommend that I ask the list and get some input before he makes any changes to it. fewestcalls from what I have seen would always ring the agent with the fewestcalls first then go into

Hi, Which is the correct syntax to call using IAX? I have two Asterisk boxes behind a NAT and one of them use the default port 5036 for IAX, the second one use 5038. To call an extension of the first one, the line in extensions.conf is: exten => _9XXX,1,Dial(IAX/user:pass@195.3.32.191/${EXTEN:1}) and for the second one: exten => _8XXX,1,Dial(IAX/user:pass@195.3.32.191:5038/${EXTEN:1})

Anyone seen this? It ocurred on a 'reload app_queue.so' command. Asterisk version is 1.2.9.1. Tried again, but it was not immediately reproducable. Doug. (gdb) bt #0 reload_queues () at app_queue.c:3339 #1 0xb778a7a8 in reload () at app_queue.c:4012 #2 0x0805bb44 in ast_module_reload (name=0x8137cc7 "app_queue.so") at loader.c:257 #3 0x08092b3f in handle_reload (fd=33,

Hi, Would someone assist me with the configuration of my TMD10B and X100P cards? I would ask Digium but my support was good for only one install...( I had to rebuild my * box). I wrote down the configuration but I must have made a mistake somewhere. I have followed the instructions on the Digium site but I must be missing something. Here's the config (snippets) I've been attempting to