similar to: Harris picking up before extension

Hi there! I'm setting up a connection between Asterisk ver. 1.2.13 and a Harris 20-20 PBX. More less everything went fine, but the problem I have now is that when dialing to the Harris PBX it seems to pick up my call as soon as it reaches it. For example if from the Asterisk outgoing folder I dial an extension, say 100, and play it a prompt as soon as it is picked up, the promt is beign

I'd like to have Asterisk log useful messages during operation. Is there any way in extensions.conf that I can manually log messages to a file, say via syslog()? The console output is ugly, with all the extra "Executing NoOp("SIP/pstn.voip.com-08a28bd0"," crud at the front of each line. I'm not sure how to save console output anyway. Thanks, Doug.

Hi! I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've followed the info in http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration but still not able to make Asterisk communicate with Cisco. I keep on receiving --- SIP/2.0 400 Bad Request - 'Malformed/Missing URL' --- and --- SIP/2.0 404 Not Found --- messages

Hello Astrites, I was just about to send out a long email about not being able to detect hang-up with my CAC II FXO module on my PSTN POTS lines. I had tried many configurations. I had attached a toner to the line to see if I was getting this "disconnect supervision" signal after hang-up. (toner has a line powered light in the off position) It seemed that I was getting the signal,

Callerid issue 1) if you run ztmonitor on the fxo line & call in do you hear the fsk tone if yes then we beleive the CAC is passing fsk 2) in chan_zap->ss_thread around line 4154 (current cvs) if you get to the callerid_feed at least once then if you get to chan_zap->ss_thread->callerid_get around line 4163 (current cvs) does this parse fail

Hi, Is there a way to create trunk groups while asterisk is running. For exemple let's say that zapata.conf defines g0 as channels 1-23 I would like (while asterisk is running) define g1 as 1-10 and g1 as 10-23 Any hints appreciated. Andre Courchesne

Hello! I'm tryuing to bring up an R2 connection but eventhough I've followed the guidelines in: http://zarzamora.com.mx/asterisk/17 something seems to be missing. When an incomming call is generated I get: Nov 24 06:01:17 WARNING[-197416016]: chan_unicall.c:612 unicall_report: MFC/R2 UniCall/24 <- 0001 [1/ 1/Idle /Idle ] Nov 24 06:01:17 WARNING[-197416016]:

Hi guys, I''m setting up asterisk to run with another pbx server. This pbx server support a feature that allows 2 extensions connect to the same FXS. No I put asterisk in the middle. Asterisk receives the call and dial to a SIP/peer. How the pbx installed support 2 extensions to one fxs... How can I figure out in asterisk which extension was dialed before the call came to asterisk?

I should've mentioned that before. I've tried doing that and it has no effect. I've tried both upper and lower-case 'r's. I've also tried a workaround that I thought would work, but it doesn't: Answering the call and then using the playtones(ringing) command before connecting to my cellphone. -----Original Message----- Date: Sat, 1 Apr 2006 19:59:46 +0100 From:

how to adjust voice volume for sipura 2000 and cisco ata186?

Hi, all I have installed TDM400 with 1 FXS and 1 FXP ports. Now I am goig through documentation on how to configure it. It mentions 3 protocols: Loopstart, Groundstart and Koolstart. Which one do I use? Can someone send me sample zaptel.conf file for Australia? This will save me some time and will be used as a working example. Thanks, Rudolf

I need to know how to setup the data side of the T1 on my Linux Box. I have found information about configuring a PRI and HDLC but nothing about the Frame-Relay type setup for data. The following is information from our T1 provider. Network T1: Framing = ESF Line code = B8ZS Build out = 0-133ft(DSX)/0dB(CSU) Clock = network Pulse-density-enforce = off alarm-option = on alarm-delay = 15

Hi, I have this setup: (PSTN E1 PRI) -- Asterisk -- (crosscable) -- Alcatel PBX --- analog phones and a few of VoIP phones directly connected to Asterisk. Calling a number (only one until now!) - an automatic responder (IVR) - from VoIP phones works, from analog phones doesn't work: NOANSWER after a few seconds. I'm using no 'r' in dial options (this caused a problem with an IVR

Ok guys, after doing many tutorial arond the web i realize that im not going to learn rails well until a made my own app, so i defined what i want and structured in a way that hope you can understand, maybe we can help each other and define a good starting base(models and relationships) that can be helpful for all of us. Its a little(i think cms), it will consist of basically 4 sections:

Hi everyone, I've just got my X100P card installed and working but there seems to be an issue with hang-up supervision. If I stuff a call out over the X100P card onto the PSTN that's fine. When I hang up the SIP phone the PSTN call ends. If I receive a call from the PSTN, it's answered and everything is ok until the remote party hangs up. Asterisk thinks the call is still active and

I bought an Aastra PT480 from digium, but I wanted to see if I could get some more help with this before Monday. Any help would be appreciated. I have the phone connected to the TDM400P card, and I also have the T100P and the X100P in the same box. My problem is, it appears as if the phone and asterisk can't understand each other. The port the phone is connected to always remains

Hi everyone (Mark, Jim). I am new to the list but thanks to both Mark and Jim, I have being using "asterisk" since summer 2001. I am just updating my version that was a year old. Yes, I know, I got busy with other things like paying bills so I don't have to sleep with the dog anymore. Anyway, one of the frustrations I have been dealing with (keep in mind that my version of

On 14/06/16 17:32, Juan Ignacio wrote: > Rowland, a question. > > > "is to copy idmap.ldap from the first DC to all others and then keep > them in sync, the other is to use RFC2307 attributes." > > I can do the same with my member server? Maybe it works, or not for > beign a member server. > > Maybe i can change my Member Server to a Domain Controller and

Hello, I have a 4 port FXO digium card with 3 PSTNs attached to it and AsteriskAtHome setup. Everything is working fine except outbound calls. When I dial a outside number, it works fine, but when another employee trys to dial out while I am on a line, it will not go. I have a outgoing route setup in the AMP interface. Dial Pattern: 1NXXNXXXXXX NXXNXXXXXX NXXXXXX Trunk

I am having a problem using the Adtran 750 FXO quad card with a Groundstart trunk line. I am able to receive calls on the trunk line, however dialing out is not working. The Adtran does not seem to be doing the signaling. Has anyone used the 750 FXO card in Groundstart mode? Any special configuration issues that I should be aware of? Syed Akbar Alico Systems Inc www.alicosystems.com Tel: