similar to: Callerid matching in extensions.conf

i want to tag categories to its menuname. i have a csv containing menu item name and in other csv i have a column containing some strings, i want to pick that strings from categories and look into menu items if any menu item containing that string i want to create a new column next to menu item name flagged as 1 otherwise 0 and the only condition is once a menu item flagged as 1 i don't need

Can someone recommend the best way to view current calls in progress on the Asterisk console? Neither the 'show channels' or 'sip show channels' commands are easy to read. hestia*CLI> show channels Channel Location State Application(Data) SIP/2944093-f9e2 (None) Up Bridged Call(SIP/2944079-e7f2) SIP/2944079-e7f2

I'm having some issues with Background() and Read() commands. See the example below. This is when I wait for Background to finish playing the sound file, before entering '12345#'. All works fine. hestia*CLI> -- Executing Answer("SIP/2944093-3366", "") in new stack -- Executing Wait("SIP/2944093-3366", "1") in new stack --

Is it possible to take the contents of a mailbox and feed them to the account's .active_sieve file for reprocessing? (For example, when editing the sieve file for my list account I introduced a typo, so a hundred or some list messages ended up in the inbox instead of filed properly into the maildir hierarchy. Not a huge deal, as it was simple enough to move them manually, but it got me

Hi all, I am trying to figure out if you can store the definition of a directory storage pool in a custom location. This is how I create the storage pool: xmlDesc = """ <pool type='dir'> <name>guest_images_storage_pool</name> <uuid>8c79f996-cb2a-d24d-9822-ac7547ab2d01</uuid> <capacity unit='bytes'>4306780815</capacity>

Hi, On a particular input file, FLAC (testing with current git) greatly inflates its output if I encode at level 7, which enables --exhaustive-model-search. The source is a 24-bit WAV file of about 60MB; flac -6 encodes this to a 43MB FLAC file, but flac -7 produces a 9.1GB (!) file. The enormous file does seem to be perfectly valid, FWIW -- it (eventually) decodes to a WAV that's

I have a tricky situation. I have a polycom phone with number 3254103. I have configured the phone to forward to a new number, 18059999999. Here's my dialplan: exten => 3254103,1,Dial(SIP/3254103,10,tr) exten => 18059999999,1,Dial(SIP/11101553818059999999@proxy2,40,tr) When Asterisk dials 3254103, here's what comes up on the console: hestia*CLI> -- Executing

> On 29/09/2020 09:09 @lbutlr <kremels at kreme.com> wrote: > > > Is it possible to take the contents of a mailbox and feed them to the account's .active_sieve file for reprocessing? > > (For example, when editing the sieve file for my list account I introduced a typo, so a hundred or some list messages ended up in the inbox instead of filed properly into the

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I made a call from 3254102 to 2944093. I then tried to do a transfer to 3254107. IP addresses have been changed to protect the innocent. It appears this related to bug 3710. It's unclear from the bug if the problem has been fixed or not. If it hasn't, then this seems pretty serious and would I guess affect any NAT-ted phones ability to transfer calls. Here's the REFER that the phone

I have a SIP user, 2944093 that dialled 3254102. I'm trying to transfer the call from 3254102 to 3254104. When I try and transfer the call, I get the following on the Asterisk console. Mar 3 15:14:18 NOTICE[23124]: chan_sip.c:6731 get_refer_info: Supervised transfer requested, but unable to find callid '16749440-c28be02e-64b73be7@172.31.16.67'. Both legs must reside on Asterisk box

Trying to get SIP realtime working here... I'm connected to the database... *CLI> realtime mysql status Connected to vox180internal@db1.ipt.XXX.com, port 3306 with username voxadmin for 6 seconds. I can get information for the extension in question... *CLI> realtime load sipusers name 2944093 Column Name Column Value

Hmmm. Not having much luck with this. I'm trying to call the DUNDILOOKUP function and assign it to a variable in an AGI script. I've tried setting with EXEC CMD and with SET VARIABLE. In both cases, it's treating DUNDILOOKUP literally, rather than calling a funciton. I've tried this: EXEC "Set" "DIALPATH=${DUNDILOOKUP(2944093|180net)}" and also: SET VARIABLE

Got an email this morning with the subject "Welcome to Gizmo Project". I didn't sign up with those yokels. Anyone else got spammed by them?

The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug.

Arrgh. I just made a call with Asterisk to extension 2944093. That extension exists in astdb and I have rtcachefriends=yes in sip.conf. Asterisk did a database query... SELECT * FROM ast_sip_users WHERE name = '2944093' Uhm... Why? Doug

Oh heck. It really looks like realtime has been seriously screwed up. When a call comes in to Asterisk, I can see asterisk executing these queries. SELECT * FROM ast_sip_peers WHERE host = '2XX.YYY.142.205' SELECT * FROM ast_sip_peers WHERE name = '2944093' SELECT * FROM ast_sip_peers WHERE name = '2944093' So, the first thing it does is check and see if there are any

I hope I have this wrong, but when I have a bunch of priority 1 NoOp's created from regexten in sip.conf, and I do an 'extensions reload', I lose all the priority 1 NoOps! This can't be right... this means that in a production environment, if you make a change to your dialplan and do an 'extensions reload', you lose your ability to terminate calls to phones on this system.

I am using the latest SVN version 1.2 of Asterisk When I attempt to test the output of certain variables, for use in file naming etc, certain key ones appear to fail. exten => 5555,1,NoOp(${EPOCH}) Returns -- Executing NoOp("SIP/200-638c", "1141352935") in new stack exten => 5556,1,NoOp(${TIMESTAMP}) Returns -- Executing NoOp("SIP/200-8cc9",