similar to: problems with'#' transfer after hold...

Hi.. Has anybody been experiencing any problems with transfers using # after holding? Transfers using the # and music on hold work fine by themselves. However, when we place somebody on hold we can no longer use the # to transfer. This is a problem since we use the # button to park calls. So, say a call comes in, the operator is on a call already, places call on hold and answers the

I cannot seem to get any transfers to work at all. The console show I have #1 amd #2 set up for Blind and Attended Transfer, but when I hit these buttons on my handset nothing happens (other than I hear the dtmf tones on the other end of the line). roo*CLI> show features Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8

On Thu, March 5, 2015 05:30, Ruben R?gels wrote: > > > Am 05.03.2015 um 01:09 schrieb James B. Byrne: >> I am trying to determine how the transfer button on the Snom-870 >> works >> with Asterisk. Is the ## special code employed as when it is >> entered >> through the handset or is the blind transfer through the phone >> function accomplished in a

I got a Cisco 7960 phone recently, and have downloaded and set up Asterisk version 0.5.0. Very nice! I've set up the software on a test box for now and have configured the system to route calls that start with 7 to FWD. Once I'm happy with my various tests, I will set this all up on a dedicated box, will get a couple of POTS interface cards and will set up some proper routes etc.

Hi, I've recently installed Asterisk and have got the majority of it configured (what an excellent piece of software it is, too), but I'm having a couple of problems. The first one is with music on hold! I've downloaded and installed mpg123 as specified: ># whereis mpg123 >mpg123: /usr/local/bin/mpg123 It's the correct version: >#

Is there any way to program a soft key in ADSI to put a caller on hold. Then able to retreive that caller. Example - Softkey Hold Softkey Retreive Call Softkey End Call -gcc

Howdy, Was there ever a fix for this? I have Trix 2.6 running asterisk 1.4 and have to set an extension with call-limit=1. However that user can no longer do attended transfers from Linkys 962 ip phone. Is there anyway around this? Cheers, Taff..

Comments? Feature request: Add the ability for the "T" and "t" suffixes in a Dial command to call an extension directly (if specified) instead of going only to the hardcoded "transfer" command. Feature request: Flash events, when presented inside of an existing call, will call a pre-specified extension just like the "T" and "t" request

Hello everybody, I have a very strange issue with a Linksys SPA-3000 (1 fxo + 1 fxs) used as PSTN gateway to asterisk in a small office. Everything works just fine, except that sometimes, and it seems that only for long incoming calls, the IVR menu appears on the middle of the call(like a three way call, call goes on with prompts playing over the parties). Dialing an extension at the prompt at

Have you enabled DTMF logging and seen the DTMF codes being recognised by Asterisk? I had a bunch of soft phones that I had to change to using ?sip info? for the DTMF signalling as the RFC signalling was not always being recognised. This would cause transfers to appear as if the user had not dialled any digits. On 20/12/2014 20:52, "sean darcy" <seandarcy2 at gmail.com> wrote:

I am having problems getting any form of call transfer working. I have reconfigured blind transfers to be #1 and assisted transfers to be *2 but these are not working. Looking at the wiki (http://www.voip-info.org/wiki-Asterisk+cmd+Transfer) it it does not mention IAX so I assume I have to use the native IAX transfer supported by Diax? I have tried using Diax but am getting a problem that after

On Thu, March 5, 2015 09:56, Ruben R?gels wrote: > > Hi again, > > I'm glad to hear that I provided a somehow useful answer. > > Unfortunatelly, I don't know these details. > If you wasn't lucky consulting the snom docs, maybe the snom support > can be helpful with information about the exact implementation > details. > > You also could use "sip

--On Monday, January 19, 2004 11:01 AM -0500 Andrew Kohlsmith <akohlsmith-asterisk@benshaw.com> wrote: <SNIP'd from the "ADSI phone vs. IP phone" thread> > I'm looking at ADSI phones simply because I don't have to re-tool my > entire building; I can use the existing phone network and (I think) get > all the functionality I need with the (far) cheaper

Hello all, First of all, let me apologize about the length of this message, but I suppose it was necessary to include the details. I've spent quite some time already trying to get the call transfer function to work on my Asterisk installation. Let me first describe the general situation of the setup I am using, so you might be able to pinpoint the cause of the problem. I'm currently

I've got a situation where I need to use a transfer to the parking lot as hold, but am not going to use BLF indicators on the phone to pick up the parked calls so I need to hear the 3-digit extension after the transfer. I'm using Snom 300 phones and have tried setting a programmable button to Key Event F_TRANSFER 700, which successfully does the transfer but cuts off audio so you

Hi, I would like to use the new atxfer (native assisted transfer, see mantis item #3241) , but I've partially been able to make it work. I can receive a call and then having the caller hear MOH while talking with another extension (the one I want to transfer to), but then I can't make the caller and the trasferred talk hanging up or pressing any key combination I'm aware of. My

I am currently running stable, CVS-v1-0-05/25/05-12:07:15, with Polycom SIP phones, running 1.4.1. Too many of our transfers using the Transfer end up with zombie channels after a REFER. As such, I implemented # transfers, and all is well. Sort of. I have a reproducible issue. Take a call from a queue. Press #, and it'll transfer just fine. Now, take a call from the queue. Put them on

Recently started using * and really am having fun. One problem I encountered... I am using an SPA-3000 3.1.10d When I have transfer enabled - 'T' in the dial string I cannot reliably send DTMF keys to a bank, voicemail, or other service requiring tones. If I disable (remove transfer option) from the dial string all is fine. I would like to be able to use features but the ability to have

I am trying to connect to a vocal server from an asterisk server. A call is received via iax2 to my asterisk server. I then initiate a SIP connection to the vocal server. everything works great except dtmf doesnt work. A cisco 5300 can connect to this vocal server and do dtmf without a problem. I have my dtmf set to rfc2833 in the general section of the sip.conf . I can confirm that the

Greetings. I am attempting to configure a system based on Asterisk 1.2.3 to be used as a backup should our aging voice mail/auto attendant system fail, which seems increasingly likely given its advanced years. The first part of this task is getting the auto attendant feature to work correctly, which I would have figured to be relatively easy. I have successfully built a menu structure, but cannot