similar to: Transfer / Music-On-Hold

I have a Cisco 7960 running 7.2 of their SCCP image; I am running Asterisk 1.0.5 and using the latest Sourceforge version of SCCP2. When I make a call (or receive one) the "Transfer" softkey does not show up - as a matter of fact only 2 softkeys show up (redial & something else), but those even are not active. On a 7960 running SIP the Transfer and other buttons do show up and are

I know this is on the wiki, I just want to confirm so I don't blow up my cisco phones. I've got several cisco 7940's all running using cisco power cubes. However, my boss wants me to switch just a few over to poe, but doesn't want to fork out the dough for a nice cisco poe switch, or anybody else's poe switch for that matter. So my question is, what is the '99.999%

Hello, We are working on a new Asterisk installation and have run into some problems related to playing MusicOnHold for a caller when they have been placed on hold by an agent, that took the call from a queue. A. When pressing the HOLD button on SNOM 190 and Grandstream BudgeTone SIP phones, MusicOnHold works fine when making inbound or outbound direct calls by extension. Music starts to play

Asterisk 1.4.6 at FreeBSD6.2-RELEASE Client hears pure silence when waiting for call answer. Music on hold stops when transferer pics a number and client doesn't even hear ringing. Is this normal behaviour? How to change this? Log says everything, MOH should stop after call pickup, not before Dial. -- Executing [113 at firma:1] Dial("SIP/zytek-08737000",

I've dug through the documentation, and I must have just glanced over this, but how do I set this system up so it executes the call when 4 digits have been entered. Currently it seems to be operating on a timeout system. Cisco 7940's Right now if I press the digits 6000 the phone waits like 8 seconds before going forward. (If I press dial it is immediate). Anybody? -C

Is there a way to ring selective line on multi-line phone. For example if I'm on the phone talking internally on line 1 and the calls comes-in the line 2 will automatically ring. The phone P104 allow extension to be assign each line. Is there a way to call certain line (example line 3) on multi-line phone instead of line 1 when the phone is not busy? For example the Sip phone P104 has

Here is what I am trying to do. I have a SIP soft phone running on a PC that is streaming a local radio station. I assigned "mono out" in XP Equalizer as the mic so now I have the softphone streaming audio. I then create a conference room and dial that room from the softphone. Now anyone who joins the conference room hears the streaming audio. How can I configure Asterisk so that

Hi list, is there a way to achieve in asterisk (version 1.4.x) the behavior described below? * a caller place a call to an extension, and I want the caller hears the extension ringing for some seconds, and then hears the music on hold (or a courtesy message) _in the same call;_ * the called extension must continue to ring until answered. With the m(...) option in the Dial

Date: Mon, 23 Jun 2008 08:00:08 -0400 From: "David Backeberg" <dbackeberg at gmail.com> Subject: Re: [asterisk-users] Replace music-on-hold on MeetMe with ringing sound To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <3de056a30806230500k7e66185l7bfe473ed398ebf6 at

I have some Cisco 7940's running SIP image 6.3 and a newphone account. Reguarding my dialplan I'm having a small issue. I'd like to dial 9,2,xxx-xxx-xxxx for a LD Nufone calls - however I also need to dial local phone numbers ie 9,2xx-xxxx Currently my dialplan looks like so <TEMPLATE MATCH="9,1.........." Timeout="0" User="Phone"/>

Hi, I have trouble getting MOH to work after an upgrade from asterisk 1.4 to 1.6.1.4. The call goes on hold, MOH is started, and then stops right away. Here are the files both of type .raw: Tsunami*CLI> moh show files Class: default File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-2 File: /etc/asterisk/musiconhold/Fr?d?ric Chopin - Polonaises Op. 40-1 These files

When I receive a call, I want to automatically park it from the dialplan so that I can retrieve it later. However, I don't want callers to be aware that they are being parked, so I want to play a ringing tone to the caller. Park() is supposed to be able to do this: Park([timeout][,return_context[,return_exten[,return_priority[,options[,parking_lot_name]]]]]) options r: Send ringing

Hi, I'm am currently in the process of trying to integrate an * box with an NEC Electra Elite IPK. Currently, we have 7 POTS lines coming into our building. These lines are plugged into our NEC using the appropriate analog line interface card from NEC. The NEC effectively has NO configuration done to it, other than to make all the internal phones ring when a call comes in. We also

I recently posted a message about an interesting dilemma which I could not figure out. After doing a whole lot more digging I think I have figured out part of what is going on. The parking app is implemented as an app, ie can be called from extensions.conf) but if you look really closely, it is also implemented in each channels driver (handler, whatever you want to call it) and those

Has anyone encountered this SIP problem? I dial a SIP phone from a ZAP phone and after talking for a few minutes the SIP user transfers the ZAP user to another ZAP user. The transfer works correctly, but when the native bridge occurs -- Attempting native bridge of Zap/22-1 and Zap/16-1 the music on hold never stops. The two lines are bridged correctly and you can hear one another but you

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <font face="Tahoma">Hi Everybody,<br> <br> I&acute;m getting as strangeous issue on Asterisk 1.4.31 (Using Elastix) ... In some calls, i get an atxfer or musiconhold in the middle of

Would anyone know why Voicemail in * doesn't get the DTML keypresses from my Cisco 7940 running SIP (POS3-07-3-00) ? Is it something to do with "dtmf_avt_payload: 101" setting in SIPDefault.cnf in the tftp server? Thanks for any help! Derek -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823

Has anyone else noticed this? I use 'set verbose 25' (insane, but I want to see *everything* right now) and would like to do the same for 'set debug', but as you see, set debug has a bad impact on my CLI output. asterisk*CLI> -- Executing Answer("SIP/824-b4ff", "") in new stack -- Executing Wait("SIP/824-b4ff", "0.5") in new

Hi everyone, I'd say this question has come up and been answered before but I haven't been able to find it. I have a Cisco 7940 that I've upgraded to SIP firmware (currently P0S-3-06-3-00 - for some reason there was a failure when trying to upgrade to V7 so I left it at V6). The problem I'm having is that when I connect to voicemail the DTMF key presses dont seem to work

I have a requirement to play different hold messages depending upon the extension that originated the call. I noticed a musicclass setting in sip.conf, but it appears this is global. I tried setting this on all of my individual extensions, but it didn't have any affect. Is there a way to achieve this, either through sip.conf or in the dial plan? Thanks, James