similar to: Re: Asterisk-Users digest, Vol 1 #1380 - 15 msgs

Greeting, I am trying to setup PLAR signalling in asterisk. I have modified the FXSLS TX bits in dahdi-base.c on line 2580, and I can make calls. .sig_type = DAHDI_SIG_FXSLS, .bits[DAHDI_TXSIG_ONHOOK] = DAHDI_BITS_ABCD, /*changed by for PLAR*/ .bits[DAHDI_TXSIG_OFFHOOK] = (0), /*changed by for PLAR*/ .bits[DAHDI_TXSIG_START] = DAHDI_BITS_ABCD, /*changed by for PLAR*/ When I got to change

Does anyone know if asterisk supports PLAR (Private Line Auto Ringdown). The Oreilly (Asterisk: Future of Telephony) book mentions it in passing saying that all you need to enable it is to set immediate=yes in zapata.conf. Has anyone implemented this in brokerage trading environments? Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Im new, but trying real hard! I just need general direction, not details yet..i'll try to figure those...just looking to avoid brick walls...bottlenecks...inefficiencies etc upfront. Hardware: motorola vt2442 - trixbox Apps: Dot Net application that operates the Manager API and the FASTAGI interfaces. I have the 2442 set as a PLAR so as soon as the ext is off-hook, it dials into the *61

Hello, Could somebody tell me if I can connect CISCO 2600 router with support of H.323 to Asterisk ? If it is possible could somebody tell me how to do it. I would like to document it and put on some website so everyone can see it. Regards, -- bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have cisco phones running SCCP, and a cisco 2600 with FXO I'm using for PSTN access. I can dial out, but inbound calls are not ringing a phone. Please see my config In the 2600 I'm PLAR'ing the line and I have a SIP Dial-Peer for 4001 voice-port 1/1/0 output attenuation 0 echo-cancel coverage 32 no comfort-noise timing hookflash-out 50 connection plar opx 4001

I would think I could do this but for some reason I am stymied. I have a PRI from RCN connected to a cisco 3640 (in my day "cisco" is all lower case :-)). My config looks something like this on the cisco... --------------------------------------------------------- voice-card 3 dsp services dspfarm ! ip cef ! isdn switch-type primary-5ess ! controller T1 3/0 framing esf linecode

Hello, everyone. Right now, we are trying launch our own PBX system based on Asterisk(Fedora) with Cisco 2611. Cisco has 2 port FXO card installed on it. For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx fine. (I'll call it F). Using softphone, I can dial in extension 1001 on asterisk, which should talk to cisco. After initial connection to Asterisk, I have try to

OK, here's a configuration challenge: I want to have certain line appearances able to be "interrupted" by various other line apperances elsewhere in the office. This is harder to describe than it is to demonstrate, so I'll do that: Let's assume I have Cisco 7960's on all desks. 1) Call comes from inbound line X destined on extension 1234 2) Phones A, B, C all

say I needed to send a broadcast message that I wanted every user to hear when the pick up thier phone? can I "Play,message" on a line just before they get dialtone? or maybe after they dial before ring? how about a "ringdown" to a voicemail box and on end return them to thier line for the dialout? can * do ringdowns? when a user picks up an extension it automagically

Hey all, I'm pretty new to Asterisk and VoIP in general, so I'm hoping I can get a bit of help here. First I'll explain my setup, and then my problem. Right now I have a Cisco 3640 with a VIC2FXO card in it which has 2 FXO ports. I have an analog phone line plugged into the first port (voice-port 1/0/0). I've got it setup so that calls coming into that analog line are

Basically, I am looking to set up an extension which will be used as a "help-line". I want it to function kind of like the bat phone from the old Batman series, where Commissioner Gordon would pick up the extension in his office and it would ring the phone back at Wayne's mansion. Is there a way to duplicate this functionality with Asterisk? I just need asterisk to auto-dial an

Hi, I know how to use LASSO for model selection based on the Cp criterion. I heard that we can also use cross validation as a criterion too. I used cv.lars to give me the lowest predicted error & fraction. But I'm short of a step to arrive at the number of variables to be included in the final model. How do we do that? Is it the predict.lars function? i tried >

Me too, with 3X Snom 360's in CAP positions. I just gave up in the end and cron'd a SIP REBOOT from my * box at strategic times. If your phone supports the REBOOT directive I'd just reboot your phone every day, 4 AM or what have you. -----Original Message----- From: Bartosz Jozwiak [mailto:bartek@cq-link.sr] Sent: Monday, April 24, 2006 12:35 PM To: Asterisk Users Mailing List -

Hi all, I have looked through the wiki for any information on how to make an extension autodial another extension when it goes off hook. Anyone done this or know how it's done. regards Dave

When are 'we' going to have this solution Steve? :) You keep talking about it, and we keep asking when it's going to come about. I know myself, SS7 will be a make or break for our continued use of Asterisk. Even if we had some price indications would be good, and/or a timeframe? Don't want to seem pushy, but it's been on the cards for quite some time now. Ben -----Original

You have no dial-peer telling the router what to do with the outbound call. http://www.tape.net/~gerry/asterisk/cisco26x0.html At 12:50 PM 9/26/2003, you wrote: >Like Gerry wrote for callerid you need VIC-2FXO-M1 card. > >Right now I am stuck on making outgoing call. > >Could soembody help me with the configuration. > >On cisco I have soemthing like that: > >dial-peer

I'm having great success using Cisco 7960s with Asterisk but one Cisco 7960 specific function that's puzzling me is programming the 'line buttons' (6 buttons on the upper right side of the phone) for speed dial. I'm not using any Cisco applications like Call Manager and the phones are SIP. *All* of the documentation I've found over the last several days continues to refer

Hello, I have a question. I set up an extension to 1234 exten => 1234,1,Dial(SIP/1234@sip.greentone.com:5060) And when I dial that extension I got in SIP message "403 FORBIDDEN" Can somebody tell me why I cannot call that extension? When I am not using Asterisk I can call that extension without any problems. My SIP proxy is VOCAL. I am new here so I do not know a lot yet. Thank

CentOS Errata and Security Advisory 2011:1380 Critical Upstream details at : https://rhn.redhat.com/errata/RHSA-2011-1380.html The following updated files have been uploaded and are currently syncing to the mirrors: ( md5sum Filename ) i386: 2a9257de63e8d6912120d260a78c7b54 java-1.6.0-openjdk-1.6.0.0-1.23.1.9.10.el5_7.i386.rpm 1413a1ec73bee83f83185f9c83a3fe94

CentOS Errata and Security Advisory 2011:1380 Critical Upstream details at : https://rhn.redhat.com/errata/RHSA-2011-1380.html The following updated files have been uploaded and are currently syncing to the mirrors: ( md5sum Filename ) x86_64: dcd51b2589fe34fe74e8526907ae431a java-1.6.0-openjdk-1.6.0.0-1.23.1.9.10.el5_7.x86_64.rpm 75ab7dff52bb89a11a51a780c13a1913