similar to: Aastra 480i & Asterisk 1.8.3.2: No musiconhold

On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: > I want to change call files, which has caller id in them, to call > originate from dial plan. > But I don't see such parameter here > https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate > > How can I pass callerid to following: > > exten => 6003,n,Originate(SIP/6003 at

On Thursday, December 25, 2014 03:53:44 PM Dmitry Melekhov wrote: > 25.12.2014 15:46, Anthony Messina ?????: > On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: > I want to change call files, which has caller id in them, to call > originate from dial plan. > But I don't see such parameter here >

Is there a variable for the filename that is created by the FollowMe application when "a" is specified as an option to record the caller's name? I'd like to clean up the recorded name files after the call is complete. Thanks -Anthony -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next

Working with asterisk 1.4; using the AMI Originate command, it is possible to do something like: Variable: CDR(accountcode)123456 Or must the variable names be "var[n]" where n is a number? I'd like to set the accountcode for a Local channel that originates a call. Thanks. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE

with newest Asterisk version.=20 When holidays will end more and more people will start to complain about = this. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -----Original Message----- From: asterisk-users-bounces at lists.digium.com = [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Anthony = Messina Sent: Sunday, December 30, 2007

After updating to 1.4.15, I have the following issue: When I try to use the "M" macro option in the Dial() option, I get the following in the console: -- Executing Dial("Zap/1-1", "Zap/g2/w5051234|60|M(set-userfield^local)KT") -- Called g2/w5051234 -- Zap/3-1 answered Zap/1-1 [Dec 6 12:10:58] ERROR[19496]: app_dial.c:1541 dial_exec_full: Unable to start

Unfortunately, after http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc1c1fb12cc0661f3810ef47ad33206b2e398 I am unable to build DAHDI-Linux in a mock chroot for packaging purposes. I believe this is related to the Makefile calling install_firmware with only 2 args, where install_firmware is a shell script with DESTDIR set to $3, which is empty. In this case, the DESTDIR

I was cleaning and working on laptops most of the day. Check my logs, I did plenty of work. -----Original Message----- From: "asterisk-users-request at lists.digium.com" <asterisk-users-request at lists.digium.com> To: "asterisk-users at lists.digium.com" <asterisk-users at lists.digium.com> Sent: 11/29/2008 1:13 PM Subject: asterisk-users Digest, Vol 52, Issue 81

i currently have a samba pdc, samba bdc and samba member server all running samba-3.0.23c-1.fc5. up until the 3.0.22 releases, i never had any problems with users authenticating to member servers. problem now is, a user from windows xp professional (which is part of the domain) can auth to the pdc and bdc, but not to the domain member server. the same thing happens from windows xp home (even

I am working on getting freenum.org ISN/ITAD numbers integrated into my exiting dialplan however I am having trouble getting the extension matches to work "as expected." I would like to be able to do something like: exten => _X.*.,1,Macro(isn-outbound...) Where I would expect that any extension that starts with at least one number, but includes a literal "*" followed by

Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: -- Starting simple switch on 'Zap/1-1' -- Executing [2111 at internal:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing

I am a fairly novice Asterisk 1.4 user who used to use CallWeaver, based on asterisk 1.2. I used Realtime MySQL with CallWeaver and am currently using the very same MYSQL tables (and columns) with Asterisk 1.4.11 and things are working well. The questions I have are, since new configuration variables have been added into Asterisk 1.4, can I simply add columns in my MySQL sippeers table for

Recently, I've been having issues with the URIs returned from e.164.org and toll free calls. It seems that the URIs that are returned from ENUMQUERY and ENUMRESULT are no longer the proper numbering schemes that the poviders use. I've been using the following [enum] template in my outbound route for quite some time with great success until recently. [enum](!) exten =>

I have been playing with a 480i with the new firmware 1.2.0.162 I hope to get some form of paging intercom function to work. In the wiki someone post that ALERT_INFO type of paging might be in this version of firmware but I have been unable to find anything on this yet. I have tried sending the ALERT_INFO to the phone a number of ways with no results. I then hooked up my bt100 and tried to dial

Having scoured the web, I still am no better off .. I have 2 Aastra480i's , and 120+ cisco 7940's :) . I am trying to decide which model to use going forward when we purchase more kit. They both seem much on a par regarding features. Q1: Is there anyway of making the cisco auto-answer _without_ having to manually edit the configuration on each phone ? I've been able to get the

Does anyone have success in upgrading Aastra/Sayson 480i CT firmware? All I get, no matter what I've tried is "Unable to upgrade firmware". tftpd is working because the dialplan freshens, and I have aastra.cfg whatevermacaddressfile.cfg and firmware.st in /tftpboot Am I missing something stupid? Is there another way to upgrade it? Chris Coulthurst chris@shuksan.com

I have read the user reports on the Aastra 480i SIP phones, but the last report is from October, and the advice was to stay away from these units. I was looking at the cisco 79XX series or the SPA-941's, but they don't have backlit displays. I also like POE support so I don't have to run multiple wires to the phones and makes for a neater look. I dislike the looks of the SNOM

But where do can you get this later firmware from? I'm still on 1.0.0.78 on my 480i. Regards Lee -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Peter Passchier Sent: 05 August 2005 00:04 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] AASTRA 480i Firmware 1.2.0.162 SIP ALERT_INFOproblems

Anyone know what the POE draw is for the Aastra 480i phones? We have switches that will do 15 watts on 12 ports but only do 7.7 watts on all 24 ports. A Cisco 3560 switch will do 15.6 watts on all 24 ports. Just trying to find out if we need that much power. Can't seem to find any info on the Aastra site. Comments?

I'm currently testing these phones and so far I'm very pleased with them. I'm using the SIP image (Version: 1.0.0.23 Release: 0032-01-A4). There are a few issues with the speaker not working properly when you dial out or pick a call but this is a known issue and will be hashed out in the next release. I've been working with the developers at Sayson and they have been very