similar to: [Asterisk-Dev] Cisco ATA 186 chan_mgcp Transfer problem

Hi all, i am using CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a with Cisco ATA-186 3.1.1 atamgcp We are used to make an special ;) blind transfer like (Flash)Number(Hangup before anyone answers or ring). Then * crashes (see below) if the man in the middle is an cisco-ata-186-mgcp If one waits until the last one rings, then hangup, everything is fine. If one waits until the last one

Hi, in following of a recent discussion I got to work on MGCP with the Cisco ATA186 again, and got it to work very nicely. However, there is a little thing with transfers I would like to get comments on: Call comes in from PSTN and goes to an ATA186 (MGCP) Call is answered and then, using flash, transferred to another extension If the extension is available, there can be an announcement and

Hi all: I've got some core dumps with chan_skinny. The client is ATA186 with v2.16.1.ms ata18x (Build 030814a). The * version is CVS-10/05/03-16:03:26. When I make a call, the phone connected with ATA rings only 1 time and * dies. Maybe I have some errores in ATA config. If someone has proven configs for ATA, please send me the details. Thanks in advance, Gus The logs: *CLI> Version

Hello All, I am trying to get some ATA 188 units to register with my Asterisk box over SIP. I continue to get the same "401 Unauthorized" Error when they try to register. If I turn Sip registration off, I can use the phones without any problems with a static IP assigned in my sip.conf file, but I can't get the second phone port working. I've set up two separate logins both

Hey all! I'm having problems trying to set up an ATA 186 with my Asterisk box. When I get the phone to place the call, I type the extension and I only get busy signal after 5 seconds. So I can't call my Asterisk box from my ATA and either call from my Asterisk to my ATA. Does anybody know what can be happing? Log is attached.. tks regards Oz -------------- next part -------------- >

Hello, I need to detect both fax and answer machine, and it should be valuable that the detection will be run by the Diva card itself. So : - I read Diva Documentation, and I found that the Diva could send some specific DTMF, if I had "[..] enabled [this functionnality] by the application for a designated controller through a manufacturer request command 9 [...]", but I didn't

hi using chan_capi, I get _lots_ of hanging channels after a while. This was first beleived to be SIP related, but I doubt it. below, 'roy' is on MGCP, and 'fax' is just a bridged dial if someone dials in, it's re-routed to another external number roy asterisk1*CLI> show channels Channel (Context Extension Pri ) State Appl. Data

hi all, i have a probably very stupid question/problem. for testing purpose i am trying to get asterisk running with two isdn cards. I'd only like to here the demo sound when i call the number - but nothing works. The output of show channels is not showing any channel - should there be 4 channels ? - capi info shows my two cards perfectly. The ISDN Controller's are attached to an PTMP

Hello. I have two snom200 ip phones and 1 mp108fxs (audiocodes 8 fxs) and i dont get a ring in the caller phone when I dial from a snom200 to the other snom200 or the mp108fxs, I made a debug with ethereal, and I can see a "Ringing" packet being return from the called snom200 or mp108fxs to the asterisk box, but it is not being re-transmitted to the caller snom200. Altough

Hi all, -------- I have a Asterisk at a public Network (official IP address). In the local network I have isntalled a Snom 200 IP phone and in my home network (behind NAT) a Snom 100 device. I can dial the Snom200 device from my home location without any problems but the Snom200 can not dial me. It always gets a "we do not rely". I tried to forward the SIP Port (5060) UDP via UPnP

Dear All, I have a production system with and Eicon Diva Server XBRI, and all works fine, but Today I have seen two crash with the next messages in console: -- Executing Dial("CAPI[contr2/971846147]/10", "SIP/790||r") in new stack == Everyone is busy at this time -- Executing Hangup("CAPI[contr2/971846147]/10", "") in new stack == Spawn extension

I am using the snom 200 with Phone type snom200-SIP Version snom200-SIP 2.04g Bootloader URL http://www.snom.com/download/snom200-boot1.9.bin Firmware URL http://www.snom.com/download/share/snom200-2.04o-SIP.bin I am using asterisk stable tree. I had to disable "Challenge Response on Phone" on my snom; I could not get it to work with

hi all I'm trying to use the manager interface to do some nagios (http://nagios.org/) integration, and I find some parts of it not really optimal. What I'd like to change, is to make \r\n\r\n an actual terminator, something it isn't today, AFACS. Below is the Status output - it shows Response, Message, \r\n, Status post, \r\n, Status post etc etc. Without a parsable terminator, I

Check the Redirection on the web interface of your Snom 200. If it says "When Busy" that's your problem. It should say "Never". Also make sure on Sip->Lines your line appearance says "All" or else you will have the same problem. Hope this helps :) Greg Gregory P. Scasny Golden Technologies Inc. http://www.golden-tech.com 219-462-7200 -----Original

Try this one. Took me a while too. http://www.snom.com/download/share/snom200-2.05c-SIP.bin > -----Original Message----- > From: M3 Freak [mailto:m3freak@rogers.com] > Sent: Monday, May 17, 2004 11:42 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Snom200 Firmware: I only see 2.04g > > Hello all, > > I've noticed several messages about the

Alright, this may seem like something relatively easy to do but I must be missing something or had a neuron misfire. I am trying to get The Status lights on my Snom200 hardphones to display the status of each one of my PSTN lines in my Asterisk server. Current Config: 3 X100P cards Asterisk CVS-02/25/04-18:06:52 5 Snom200 phones I am currently using the following macro to dial out

The snom 200 and 220 have five programmable buttons. Each button has a led that can be used to indecate if an extension is idle, in use, or ringing. A button pannel for the 220 is also comming out soon that will have 20'ish programmable buttons on board. This is a killer app for any company that has receptionists handle calls, and pretty usefull for everyone else. As a matter of fact,

hi when two calls are active on controller 2, chan_capi won't use controller 1. this is with AVM C2 roy -- Executing Goto("SIP/torgeir-b476", "capiring|BYEXTENSION|1") in new stack -- Goto (capiring,90044875,1) -- Executing Dial("SIP/torgeir-b476", "CAPI/22545066:bBYEXTENSION|120|Ttr") in new stack == data = 22545066:b90044875 ==

Hi; I've been asked to take a (remote) look at an Asterisk@home system running asterisk 1.0.9 on Centos 3.5. It's running chan_capi-0.3.5 It has an AVM c2 ISDN card which is plugged into what I believe to be a couple of BT ISDN2e "System Access" (i.e. point to point) connections. We've placed a support call to BT to find out how these lines are actually provisioned, but

Hello, I'd like to be able to see if a channel is use and handle the call differently if it is. The best I can find is the command ChanIsAvail(). The problem is, I have an snom200 phone which does call waiting, so even if it is engaged in a call, a second channel is still available on it. I would like to be able to differentiate between these two cases: no calls engages, or calls