similar to: BlindXfer Sensitivity

Hi Guys We have a client running on a polycom vvx400 IP phone on our asterisk 1.8.18 system The issue we have is the switchboard lady uses ## to transfer calls but sometimes it just does not work and just plays the DTMF tone to the calling party. Is there any way to adjust the sensitivity of the blindxfer feature? The polycom Transfer button is useless as there is a big delay

On Mon, Feb 16, 2015 at 10:13 AM, Andrew Colin <andrew at vsave.co.za> wrote: > The strange thing is its only sometimes my dial string is as follows > > exten => s,1, Dial (SIP/200,, tT) > > For that particular route but obviously s,3 as have Ringing () first etc. > After she pushes ## 6 times it will go thru sometimes. > > How is the DTMF being transmitted from

I'm wondering whether there's a problem with the blindxfer and atxfer commands. I was using Asterisk STABLE and pressing the # key to transfer calls worked fine, except of course when you called up FedEx and they asked "Enter the number of packages, followed by the Pound key". I found on the wiki (http://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf) that

On 12/21/2014 04:42 AM, Patrick Beaumont wrote: > 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 12/21/2014 11:09 AM, sean darcy wrote: > On 12/21/2014 04:42 AM, Patrick Beaumont wrote: >> 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

On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote: > I've got a confbridge set up which works if dialed locally: > > -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack > -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1", "1") in new stack > -- Executing [266 at internal:3]

On 12/20/2014 03:22 PM, sean darcy wrote: > On 12/19/2014 09:42 AM, Rusty Newton wrote: >> On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote: >>> I've got a confbridge set up which works if dialed locally: >>> >>> -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack >>> --

I've got a confbridge set up which works if dialed locally: -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1", "1") in new stack -- Executing [266 at internal:3] ConfBridge("DAHDI/1-1", "1") in new stack -- <DAHDI/1-1> Playing

On 12/19/2014 09:42 AM, Rusty Newton wrote: > On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> I've got a confbridge set up which works if dialed locally: >> >> -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack >> -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1",

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:

Problem with this is client needs to listen to the call recordings and my interface will only display .wav or .mp3 so they will moan if they have to wait until the next day for today's recordings Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: binary <dreamer.binary at gmail.com> </div><div>Date:01/07/2014 6:09 PM

On Mon, Dec 22, 2014 at 4:00 PM, sean darcy <seandarcy2 at gmail.com> wrote: <snip> > How do I enable DTMF logging? > > logger set level DEBUG > No such command 'logger set level DEBUG' (type 'core show help logger set > level' for other possible commands) > > didn't work, even though: > > help logger > logger mute

Hey guys Is it possible to record with mixmonitor straight into mp3. I am trying to reduce disk space and want my calls to be recorded in mp3 Instead of wav. Sent from Samsung Mobile <div>-------- Original message --------</div><div>From: Sameer Rathod <sameer at hostnsoft.com> </div><div>Date:30/06/2014 9:23 PM (GMT+02:00) </div><div>To:

Hello all. Dialing with tT options and function Read (to prompt number) has a trouble for me. Can I temporarily features off during Read? features.conf: [featuremap] blindxfer => ## ; Blind transfer (default is #) atxfer => ** ; Attended transfer I try: exten => s,n,Set(LOCAL(tmp_atxfer)=${FEATUREMAP(atxfer)}) exten =>

Hi Is there a way to disable blind and attended transfer during a call. I am trying this configuration but unfortunately with no luck: - in features.conf [applicationmap] disabletransfer => 9*9,self,GoSub(disabletransfer,s,1) - in extensions.conf [incoming] exten => 99,1,Set(__DYNAMIC_FEATURES=disabletransfer) exten => 99,n,Dial(Sip/alice,120,tT) exten => 99,n,Hangup()

Hi folks, I was using the following featuremap: blindxfer => *1 disconnect => *9 atxfer => *2 parkcall => *7 automixmon => *0 and everything worked. Then the need arouse to use some features like automixmon during a conference, but MeetMet has the * key bound to the (admin) menu. Thus, in order to enable features like automon and transfers even during a conference, I

Hello asterisk-users, I setup my asterisk to support several features like automon,blindxfer,atxfer,parkcall etc. by using features.conf and the global variable DYNAMIC_FEATURES=automon#blindxfer#atxfer#parkcall#disconnect in extension.conf. Every Dial() command in my diaplan has the appropriate parameters out of {tTkWwW}. For calls from my SIP phones everything works fine. Pressing #1 will

i have TB instaled and i cant get call pickup when another phone rings i tried ** , *8 , *8# , **+ext but nothing seems to be ok.on extention menu i put call pickup=1 and call group=1 but nothing look at my features.conf; ; Sample Parking configuration ; [general] ; do not manually enter parkinglot config information, use the parkinglot module ; ; the parking_additional.inc file is

Hi, I used to have "##" configured in asterisk 1.2 for blindxfer. Now, when I press ## I hear it on the other end instead of initiating a transfer. What has change and how can I go back to the old behavior? I kept the same feature.conf file with these lines: [featuremap] blindxfer => ## ; Blind transfer Also, for the first time ever, I have a problem connectiong

I have a group of local users. They want to participate on a conference call to a PSTN line in USA. To connect they need to enter a code there as well. I want to use a local conference room, where LAN users and local users can call in. The conference room should be connected to the conference abroad. features.conf has: [featuremap] blindxfer => #1 ; Blind transfer ;disconnect