similar to: Asterisk does not persist callgroup and pickgroup configuration.

Greetings. I got two asterisk servers, one is connected to another via sip trunk. The incoming calls are routed to the time period an if matches is transfered to the designed extension. If don't, is redirected to a second time period. Then, if the call matches the second time period it need to be transfered to the trunk that directs to the second server. How do I do to configure it this way?

Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? Any further information needed just ask. -- Att.* *** -------------- next

Sauda??es. Como que se faz um conector RJ45 em uma ponta e RJ11 e outra. Pretendo conectar a linha de um ATA em uma placa Khomp KFXO IP. A ponta que tem o conector RJ45 est? crimpada com a sequencia 568B e vai ser conectada na placa Khomp, mas a ponta RJ11 eu n?o sei como deve ficar. Li alguns manuais na internet mas n?o entendi ao certo como tem que ser feito. -- Att.* *** Luis H. Forchesatto

Greetings. I've run into a trouble when trying to add a new Win7 machine on a domain. The domain is controlled by a server running Samba + LDAP (samba compiled with ldap support), on a Debian 5 OS at the local network. I've added the machine name to the LDAP three through phpldapadmin using the option "Samba3 Machine" on the related submenu and via terminal on samba. Then I

Hi All: I used FreePBX to configure Asterisk, and tables are create in MySQL by using FreePBX install script. I created two x-lite softphone accounts by using FreePBX, they are stored in table sip as friend. I followed wiki doc to edit the extconfig.conf file. I can not get those two softphone to talk since I got the error message from Xlite as: Call failed: 503 service Unavailable I noticed

I stood up a samba 4 (4.0.9) Active Directory domain controller on a Red Hat Enterprise Linux 6.3 server, configured in accordance with the Samba AD DC HOWTO <https://wiki.samba.org/index.php/Samba_AD_DC_HOWTO> , and tailored to the domain name I want. I'm trying to join a Windows 7 Enterprise Edition client to the domain. Windows responds with "Your computer could not be joined

If you define a sip peer, wheather or not you put an entry in the field OUTBOUND CID, if you dial an external extension (let's say an extension on another asterisk server, connected via IAX2 connection) the callerid received by the foreign asterisk is device <YOURNUMBER>: i.e device <567> If you take a look at etc/asterisk/sip_additional.conf, you can see under the SIP extension

Hello all, For some reason I am not showing registration in SIP. Can anyone give me an idea what can cause this? asterisk1*CLI> sip show registry Host Username Refresh State -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050311/bd3a7577/attachment.htm

Hi Francois, I kwnow, but I have "DSP:on" also if i not have an hardware echocan module :/ and I always have "Echo Cancellation: 0 taps, currently OFF". This is my zapata.conf [channels] language = it usecallerid = yes callwaiting = yes usecallingpres = yes callwaitingcallerid = yes threewaycalling = yes cancallforward = yes callreturn = yes switchtype = euroisdn

Greetings. I've plugged 3 analog lines on an ethernet cable in an Khomp card to receive it's incoming calls. Without any configuration, when I call those numbers the asterisk server automatically answer the call and play the default music. The problem is: I need to discern the lines and redirect each one to his respective extension. Since they doesn't got any Caller ID Service the

I have a little confusion about the general settings (other than the register values) in the SIP General area. I understand that for examle in a SIP context like [FWD] or [BROADVOICE] the entries in those areas are ths settings that take effect in any communication woth FWD and/or BROADVOICE. However, I'm confused as to the purpose of the "general" settings -- to what or which

Hi, I have problem with X100P clone card.I can not force it to work under Asterisk 1.4.27.1 and DAHDI Version: 2.2.0.2. I looked over and over on configuration and could not see any mistakes. Here are relevant configuration files. /etc/dahdi/system.conf > fxsks=1 > echocanceller=mg2,1 > loadzone = hu > defaultzone = hu > dahdi_tool > OK Wildcard X100P

Hi there! I am having problems registering my 7970 hardphone with Asterisk 1.4(with FreePBX interface). I had an earlier post about trying to get it to work first with a 7970 emulator (Cisco IP Communicator) on the Asterisk Forum : http://forums.digium.com/viewtopic.php?t=19160 Instead I decided to try the real phone instead, and was able to advance further. The firmware was able to install

Hi All, I'm trying to connect to a SIP carrier who never connected with Asterisk. I managed to connect with a sipura phone or a grandstream, no problem. When I configure asterisk, I'm able to send out calls to the carrier no problems, however, receiving calls doesn't work, and I keep getting the following messages: <-- SIP read from 69.xx.xx.xx:5060: INVITE

I am trying to connect two Xten eyeBeam Video Phone No problems in voice connecting. I tryed to modify my sip.conf [general] language=it videosupport=yes ; enable Asterisk video support port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=h263 allow=gsm allow=ulaw allow=alaw ; H.263 is our video codec ;

I was previous using Asterisk 1.2.9.1 and decided to get some real servers outside of my house. It was time for Asterisk 1.4.4. I figured since all the conf files were in /etc/asterisk form the old box, i'd just copy tha directory over to the new server. My SIP DID AGI stuff worked, except running 'asterisk -r' doesn't. It tells me ' Unable to connect to remote asterisk (does

I have downloaded and installed Asterisk@home and I have installed X-Lite on my Windows machine and I am able to connect it to the Asterisk server. I went ahead an created an account on Broadvoice today and followed the directions on http://voip-info.org/wiki-Asterisk+settings+Broadvoice and http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup but when ever I try and make a call from

Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my

Longtime lurker, first time poster. :) A client of mine is in need of having Asterisk record every call that comes in from a specific incoming route. I've added the following lines to the sip_additional.conf file, but no recordings are showing up in the /var/spool/asterisk/monitor/ folder. record_out=always record_in=always Another page I came across on Google (

Egad, not again with Broadvoice! Anyhow, I recently installed AAH and configured my TDM11B and got that and some SIP phones working. I still have some issues to work out, etc, but my current problem is Broadvoice. I have checked out all of the online resources, including the recent list exchange about the recent changes made by Broadvoice. However, the one thing I have found to be consitent in