similar to: kore dump

Hello list I am having an issue with Chanspy/SIP that I?m hoping someone has come across and resolved in the past. I am sending calls that come in TDM through T1 ZAP channels and go out to a SIP trunk. If I spy on the SIP channel, I can hear the person on the SIP side of the call just fine, but the person on the ZAP channel fades in and out. If I spy on the ZAP channel, and can hear

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I would like to know if anyone here knows the answer to the following question I need to implement the following conferencing feature for my agents. 1. Agent receives call from caller 2. Agent conferences a verification service 3. After finishing the verification, agent needs to drop third party (Verification service) and continue on the line with caller. My problem

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I am seeing a peculiar message on my console screen on my new installation of Asterisk 1.4.5I would appreciate any comments. Really destroying SIP dialog '3f9a224b517f88c961a311324bfe24b7 at 64.211.222.23' Method: OPTIONS Really destroying SIP dialog '099a002b064129d74fc2e4cd4a88c2ef at 64.211.222.23' Method: OPTIONS Really destroying SIP dialog

Ther is addon available for Gnome Shell Von meinem iPhone gesendet > Am 29.05.2017 um 17:14 schrieb Raymundo N.F. <raymundo.nf at gmail.com>: > > Thanks For the Answer Mike and Pete. > But Mike, In Centos 7 this directory (/usr/share/gnome-shell/js/ui/layout.js) > doesn't exist. > Another Solution?. :) > > > 2017-05-27 17:25 GMT-05:00 Pete Biggs <pete

Hi, I have these extensions: exten => 101,1,Dial(SIP/101,15) exten => 102,1,Dial(SIP/102,15) exten => 0,1,Dial(SIP/101&SIP/102,15,r) They work fine and I get the ringing sound if I dial them directly. However, I also have this extension: exten => s,1,Answer() exten => s,2,Background(viagenie) exten => s,3,WaitExten() The ringing sound doesn't work for any extension

Usage: gnome-shell-extension-tool [options] Options: -h, --help show this help message and exit -d DISABLE, --disable-extension=DISABLE Disable a GNOME Shell extension -e ENABLE, --enable-extension=ENABLE Enable a GNOME Shell extension -c, --create-extension Am Montag, den 29.05.2017, 11:26 -0500 schrieb Raymundo N.F.: > Yes

Does anyone know how to gain access directly to the configuration files in AsteriskNow? I have dual NICs and need to change the binding in the config file. I believe they blocked ssh2 access by default. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070504/c4d288d1/attachment.htm

If you use both the public and private interfaces for VoIP in the Asterisk Server, make sure you don't specify one of them for the binding in sip.conf Example bindaddr=0.0.0.0 will allow SIP traffic on any of your interfaces. Ed Nu?ez IT/Telecom Engineer 4037 Metric Drive Winter Park, FL (o) 407-384-4200 x 1656 (f) 407-384-4222 (c) 732-925-0730 -----Original Message----- From:

On Sat, 2017-05-27 at 14:51 -0400, Mike - st257 wrote: > awk one-liner found here: > https://unix.stackexchange.com/a/196726 > > Whether it's persistent through package updates is another matter. > Unfortunately that doesn't work as the requisite file (/usr/share/gnome-shell/js/ui/layout.js) doesn't actually exist in CentOS 7 - it's for an older version of

All, Since I have now (at least partially) got my 7961G phones working with Asterisk, I have temporarily moved on to try to get the expansion modules working. There doesn't seem to be much in the way of documentation here either. Does anyone have this combination working (or any 79X1) here? My goal is ultimately to do the monitoring approach. I have Google'd around, but come up

All, I have run into some CallerID issues. It seems to have happened as a result of just moving my config from 1.2.12 to 1.4.4 (although I am not sure of this). Therefore I am sure its just a misconfiguration somewhere, I just don't know where. I have throughout the office either Cisco 7961G or Polycomm Soundpoint SIP 430 IP phones. The problem is with CallerID showing up in some

Thiss one ? https://extensions.gnome.org/extension/118/no-topleft-hot-corner/ Running here, on gnome 3.22 /3.24 Am Montag, den 29.05.2017, 11:08 -0500 schrieb Raymundo N.F.: > Thank you Andreas, but i want other solution. The Gnome-Shell- > Extentions > don't work for me. I want to know if exist another way to do this > from the > comand line. :) > > On May 29, 2017

hi, I am facing some issues while using MixMonitor. My extensions logic is attached below: exten => s,1,MixMonitor(${CALLERID(number)}-${TIMESTAMP}-${UNIQUEID}.gsm,b) in this extensions TIMESTAMP is not working in Asterisk 1.4. can any help me why TIMESTAMP is not working in Asterisk 1.4. regards, Asif

I am getting this error [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1920 retrans_pkt: Maximum retries exceeded on transmission 8f68421-22821e1e at localhost for seqno 102 (Critical Response) [Aug 6 15:28:26] WARNING[24452]: chan_sip.c:1944 retrans_pkt: Hanging up call 8f68421-22821e1e at localhost - no reply to our critical packet. any Ideas? Jason

I have a instance of Asterisk 1.2.14 that is being run from safe_asterisk. Asterisk is seg faulting and NOT generating a core dump. Why would that be? How can I make it dump core? Is there a setting in the safe_asterisk script that I am missing? Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Has anyone here ever used a Polycom IP 4000 Soundstation SIP Conference Phone with asterisk? If so, how well does it work and how does it sound?

I've asterisk stop (presumably segfaulting) a couple of times, and I was just beginning to look at how to keep it running - what have others done? I was thinking of wrapping a script around asterisk like this: while 1 do asterisk -f done /Per Jessen, Z?rich -- http://www.spamchek.com/ - your spam is our business.

Greetings, all. I would like to record calls that are entered into queues and I'm not quite sure how to do it. Here's how I'm currently set up: - Call comes in and is placed into Queue #1 (which rings all phones for 15 sec). - If call drops out of this queue, it is placed into Queue #2 (which plays MoH until the call is picked up). I've tinkered with MixMonitor and I have my