similar to: Exit Call Queue by pressing digit

Displaying 20 results from an estimated 30000 matches similar to: "Exit Call Queue by pressing digit"

2016 Sep 10
2
Queue show : failed to extend from 240 to 327
On 10-09-16 00:50, Richard Mudgett wrote: > > > On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > when I type on the Asterisk CLi 'queue show', I first get a list > of my queues and then the following : > > > failed to extend from 240 to 327
2013 Feb 20
1
Meetme and MEETME_EXIT_CONTEXT
Hello, using Asterisk 1.8.12.2 I am having trouble with exiting the conference room by entering a single digit. option X of the Meetme()-application should do this. I have following in extensions.conf : /exten => _1000X,n,Set(MEETME_EXIT_CONTEXT=dynamic-nway-invite)// //exten => _1000X,n,MeetMe(${CONFNO},dMX)// // // //[dynamic-nway-invite]// //exten => 0,1,NoOp(confno =
2011 Nov 04
1
problem when exiting from "record file" function without pressing the escape digit
Hi everybody, I've been working on a project which records the voice of the incoming call. I use record_file function of asterisk as described below: RECORD FILE <filename> <format> <escape digits> <timeout> [offset samples] [BEEP] [s=<silence>] filename: record1 format: wav escape digits: # timeout: -1 offset samples: 0 BEEP: 1 silence: 3000 *Please read the
2009 Dec 31
1
Asterisk recieves "11" when pressing "1" from SIPphone
[Dec 31 10:39:45] WARNING[17884]: pbx.c:2518 __ast_pbx_run: Invalid extension '11', but no rule 'i' in context ...[snip]... When testing IVR and pressing "1" from my Grandstream SIP-phone, the above message is printed on the Asterisk CLI. How come Asterisk receives my "1" as "11" ?? Settings in my SIP-phone are : Send DTFM : via RTP(rfc2833) &
2016 Nov 21
3
Asterisk 13.12.2 : strange queue behaviour
On 21-11-16 15:17, Matthew Jordan wrote: > > On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > when using Asterisk version 13.12.2 I notice that it takes up to > 30 seconds (sometimes even longer) for a call queue to call its > members. > >
2009 May 22
1
VOICEMAIL : I've tried a lot but mailing through Asterisk is just not working...
Don't be afraid about the info that I'm going to post in this mail, but I want you to give as much info as possible. Also I want to show you what I've tried. What do I want When a voicemail-message is left via the Voicemail()-application, I want the .wav-file send to my mail-address as an attachment. My mail-setup I'm not using sendmail as MTA. I have msmtp as MTA and mutt as
2016 Sep 09
2
Queue show : failed to extend from 240 to 327
Hello when I type on the Asterisk CLi 'queue show', I first get a list of my queues and then the following : failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 323 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to extend from 240 to 327 failed to
2010 Jun 28
3
Pickup a ringing Queue member
Hello. I'm using asterisk 1.4.30. I've found this patch for app_queue.c : https://issues.asterisk.org/view.php?id=11700 Can I easily implement this by issuing : */wget 'https://issues.asterisk.org/file_download.php?file_id=17192&type=bug' -O - | patch -p0/* ?? Does this mean I have a "patched" asterisk ? (I ask this because some applications require a
2016 Aug 12
2
Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
Hello setting "nat=no" or omitting "nat=" in peer definition does not help either. Still no audio. Why do you think this is a NAT issue ? IP and port information in SDP-body is correct. Kind regards. On 12-08-16 09:25, ????? ?????? wrote: > > Try delete nat from 770000wrtc settings ice should do the same > > > On Aug 11, 2016 10:00 PM, "Jonas
2016 Aug 17
4
pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol
On 16-08-16 17:45, George Joseph wrote: > > > On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > On 16-08-16 04:38, George Joseph wrote: >> >> >> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens >> <jonas.kellens at telenet.be <mailto:jonas.kellens at
2016 Nov 21
2
Asterisk 13.12.2 : strange queue behaviour
Hello when using Asterisk version 13.12.2 I notice that it takes up to 30 seconds (sometimes even longer) for a call queue to call its members. Example 1 : [Nov 21 08:17:57] pbx.c: Executing [queue at pbx-routing:15] Queue("SIP/incoming-00000246", "myqueue1,,,,300,,,") in new stack [Nov 21 08:17:57] res_musiconhold.c: Started music on hold, class 'default', on
2012 Dec 08
2
Queue joinempty, even after AddQueueMember
Hello, I add a member to a queue with AddQueueMember, but the Queue still indicates "joinempty" : Add member to queue : /-- Executing [queueadd at sub-GetParams:2] AddQueueMember("SIP/sip17-00005c1e", "myqueue11,member3") in new stack -- Executing [queueadd at sub-GetParams:3] NoOp("SIP/sip17-00005c1e", "AQMSTATUS = ADDED") in new stack/ ...
2011 Jun 08
1
Queue log in MySQL DB
Hello list, I have configured extconfig.conf to save queue log into my MySQL-DB. I notice however that there is still logging too in /var/log/asterisk/queue_log. How can I disable queue logging into text files ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2014 Nov 04
1
queue log realtime mysql
Hello, I have 5 Asterisk servers all using mysql realtime to store queue log information. There is 1 out of 5 servers which stores the data in 4 columns : 'data1' --> 'data 5'. All other servers store data in 1 column 'data' with the data seperated by pipe. I see no difference in my configuration of extconfig.conf and logger.conf. Maybe a hidden default value ?
2011 Mar 24
1
Fwd: Asterisk 1.6.2.10 & CDR custom added field
Hello, is there anyone who can point me to correct information ? Following http://pbxinaflash.com/forum/showthread.php?t=9042 and http://www.voip-info.org/wiki/view/Asterisk+cdr+mysql > Extending CDR does not result in a working environment for me. Any feedback appreciated. Kind regards, Jonas. -------- Original Message -------- Subject: [asterisk-users] Asterisk 1.6.2.10 & CDR
2016 Aug 16
2
pjproject 2.5.5 + asterisk-certified-13.8-cert1 : many Error loading module...undefined symbol
On 16-08-16 04:38, George Joseph wrote: > > > On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > using pjproject 2.5.5 > using asterisk-certified-13.8-cert1 > > > IIRC there were API changes in pjproject 2.5 that aren't accounted for > in
2010 Jan 27
2
CDR messed up when using queue
Hello list, I'm using an IVR where the caller chooses between 1. sales 2. support. When choosing 1 the caller is directed to the sales-queue when choosing 2 the caller is directed to the support-queue. Then the caller is directed to a free agent. I notice in the CDR-rapports that the destination is always '1' or '2', namely the DTMF-digit that the caller presses. How can I
2011 Feb 18
2
DTMF and Snom
Hello list, I'm having some troubles with DTMF tones. When pressing numbers on a Snom phone, the DTMF-signal takes too long. I have the following in sip.conf : dtmfmode = rfc2833 which works well for Grandstream, Yealink and Cisco phones. But not for Snom. Snom support tells me I should use SIP info. Is it possible to have something like this : dtmfmode = rfc2833, info ?? Because
2015 Aug 07
3
compose_func_args: argbuf allocated 4 bytes compose_func_args: argbuf uses 3 bytes
Hello I have 2 strange errors when using the Background()-application and DTMF-input that is received. First of all, my first 2 lines are not being executed. The first line being executed is the Set() application, thus line 3. Secondly, the received digits (911) is not the same as the EXTEN (which is set to 91). exten => ivr,n,Set(TIMEOUT(digit)=2) exten =>
2013 Sep 14
0
(no subject)
To Jonas: I have an asterisk box at home and I have this line in my rtp.conf file: rtpstart=10000 rtpend=10100 And My FW is setup to forward all incoming ports of range 10000-10100 to the asterisk PC. I've never had a problem since one year, but I have never received more than two simultaneous calls with SIP clients. Message: 5 Date: Fri, 13 Sep 2013 11:49:59 +0200 From: Jonas Kellens