similar to: [Asterisk-Dev] CVS-HEAD change: queue/agent persistence

I had the same problem with snom 190 phones. Using the transfer with # instead of "Transfer Button on the phone" worked for me. In my configuration "REFER" was not send, so the transfer with the button on the phone did not work. Guido Hecken -----Urspr?ngliche Nachricht----- Von: steve szmidt [mailto:steve@szmidt.org] Gesendet: Sonntag, 26. Dezember 2004 17:14 An:

Hello, I am trying to setup an Asterisk GUI with the help of astman(please visit http://astman.sourceforge.net/am-user-guide.html). I have installed astman and currently assessing my GUI using; http://ipaddress-of-asteriskbox/cgi-perl/am-main.pl I am trying to get the menu options in my GUI to work but to no avail. Currently my parameters are set to; Asterisk Install Directory:

hello i need to know how to enable the feature in the agents.conf to make the users got to press # to answer the call when is in the queue and the agent is logged in. at this time the call enters the queue and the agents who is logged in only beeps once and then the call enters automatically. can anybody help me?? TIA Edgar

> -----Urspr?ngliche Nachricht----- > Von: Mojo with Horan & Company, LLC [mailto:mojo at horanappraisals.com] > Gesendet: Dienstag, 25. M?rz 2008 23:23 > An: Asterisk Users Mailing List - Non-Commercial Discussion > Betreff: Re: [asterisk-users] Asterisk parking hold and > transferdigittimeout > > It seems that the dialplan comes into play. If your parking >

> We have our Asterisk server running smoothly with a SIP BRI gateway > for inbound calls. However if the Internet connection goes down and a > DNS server becomes unreachable Asterisk basically does not function. > By this I mean it does not answer call coming in from the gateway > (which is on the local LAN) and you can't even reload it - just hangs > there. If I change the

Hi, anyone out there with the same problems and a possible solution to the following? The functions callparking and hold use the same transferdigittimeout in features.conf. While I think 3 to 5 seconds are enough to let the user "find" their keys on the phone, the double ammount of time ( 2 x 5 secs) you have to wait before a call is parked and the parkposition is announced, is

Hi, all, I have buy 5 Digium's G.729A codec(it just support G.729A license) When I calll with 2 SIP UA that support G.729A and G.729B, its rtp frame have some problem when softswitch with Asterisk. The voice frame have been drop, so sometime I can't hear voice. If I want to fix the problem when softswitch G.729A and G.729B codec. What source code I must to modify ? Or some people have

Hello, I would like to know if FreeBSD has drivers for TE405P cards submitted to the ports. I have acquired a card and would really like to use it with FreeBSD. The wiki said the drivers havent been tested yet. Have there been any updates since then? regards Kavit _______________________________________________ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com

Hey everyone, I posted bug #3746 for people to report on the latest code in the stable branch. Once I get enough reports that it is working fine, we will release 1.0.7. Thanks, Russell Bryant _______________________________________________ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit:

Thanks Mark will try that out! -----Original Message----- From: MF Hulber [mailto:asterisk-admin@hulber.com] Sent: 22 March 2005 05:25 To: Reuben Grech Subject: [info] [Asterisk-Users] :: BIOS Motherboard Settings :: I have the same motherboard. I put the card in the 2nd slot from the bottom. In this slot, if you look at the manual, it will possibly be in conflict with some USB channels. I

Does anybody have any experience using a Patton SmartNode as a SIP/Telco gateway with Asterisk? They seem really inexpensive and appear to support all of the necessary features, but I don't have any experience with their products, so I don't know if they are any good. We are currently using a Cisco 2600 w/ PRI card and it works fine, but I was looking for someone else as a possible

I'm having a problem getting my queue to function as it should. After 20 seconds or so, it should prompt the user with a message "thanks for holding..... press # to leave a message or stay on the line to continue holding". I set up the "context" in the queues.conf file, so if a user presses a digit, they should be able to leave. But I get a SIP BUSY message. Here are my

Hello, I have SER and Asterisk running on the same box. I want SER to listen on port 5060 (it is) and Asterisk to listen on port 5062. I have configured my phones to register with x.x.x.x:5060 (SER) and Asterisk will purely act as a voicemail server at the moment. However I cannot get Asterisk to listen on a different port. It is my understanding that I just need to set the port in sip.conf

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Greetings Everyone! Version 1.0.6 of Asterisk, zaptel, libpri, and Asterisk-addons has been released. There is also a new tarball for Asterisk-sounds. They are available for download on the digium FTP site: ftp://ftp.asterisk.org/pub/asterisk/ ftp://ftp.asterisk.org/pub/zaptel/ ftp://ftp.asterisk.org/pub/libpri/ ChangeLogs are available with the

before inventing the wheel once more, I would like to ask the list Problem: the dial command does it's job by exten => _X.,1,Dial(SIP/${EXTEN}@192.168.161.135,60,tTr) now, if the given number (extension) is one of these +49(2244)870663 or +49(02244)870663 the dial command fails, since it cannot handle "(" ")" and "+" My idea was a regex, I use normaly

Here is a part of a working sip.conf for Asterisk with SNOM190 Phones. Try using host=dynamic and have a closer look at the configuration in the snom 190. Also, try using dtmfmode=rfc2833 . [general] realm = hallinux2.gwsnettech.local port = 5060 bindaddr = 0.0.0.0 context = default disallow=all allow=alaw allow=ulaw allow=gsm register => 081503:xxxxxx@sipgate.de/081503 language=de tos=0x04

Hello, How can we monitor agents voice channels for training or quality control purpose. While agent is talking to a customer we need to be able to monitor voice channel (the actual voice conversation). If possible we would like to do that without putting agents in conference rooms. Is there any possible way to do that? Has someone done this? In addition when we tried to put the agent in

I just setup an Asterisk system on a small Shuttle box; I am only using SIP channels and have no FXO/FXS cards. The system works fine in that I can call my inbound number (Broadvoice) and have the system answer and I can make outgoing calls. The problem is that every time I want to change something in the sip.conf file, I have to do a 'restart now' instead of a 'reload' or

Hello everyone, I am trying to install windows xp as a guest OS. It runs through the installer just fine, but every time I reboot, it keeps booting back to the installer. Here is my config: ------------<code>--------------------------------------- kernel="/usr/lib/xen/boot/hvmloader" builder="hvm" memory=512 device_model=''/usr/lib/xen/bin/qemu-dm''

We have a large install of 360's running rev 4.1 with zero problems. I did another, smaller install couple weeks ago with 40 360's running rev 5.3. In both cases, the install was identical, same Asterisk version, same dialplan, everything the same except the differences were: 1. Different firmware rev 2. Different physical LAN Guess what? On the smaller install, lockups and reboots.