similar to: asterisk call unique id in dialplan

Hi to all can i play a sound or a dtmf tone on a specific channel using AMI? Thanks to all -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser

i am experiencing problem with asterisk 1.2.18 I've downloaded and installed pwlib and openh323 with the following commands: cd /path/to/pwlib ./configure make clean opt cd /path/to/openh323 ./configure make clean opt then 'ive set the corresponding PATH PWLIBDIR=/data/programmi/asterisk_1.2.18/pwlib_v1_10_0/ export PWLIBDIR OPENH323DIR=/data/programmi/asterisk_1.2.18/openh323_v1_18_0/

Hi is it possible to spy (not record, spy) partially on a channel? for exaple, i'd like to listen only the input or the output voice. is it possible? thanks -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser

Hi i have a PRI connected to a TE205P. Actually, can i send and receive FAX through Asterisk using stable solutions? Or shall i connect an ATA to Asterisk and then a modem with Hylafax? -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser

Hi what version of openh323 and pwlib are suggested for asterisk 1.4.21.1.? Thanks to all -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser

Hi i generate a call from the dialplan in this mode: exten => 1002,1,Answer() exten => 1002,2,Dial(SIP/user at host) the call is generated, but after some seconds it is interrupted, here the asterisk log: *CLI> -- Executing Answer("SIP/host1-0819d0d0", "") in new stack -- Executing Dial("SIP/host1-0819d0d0", "SIP/caller at host") in new

Hi i just want to let you know that is available a new release of ccmanager. I've added the possibility to import queue_log information in a mysql database and to generate reports using this information. The software is in a beta state and provides this functionality: - users management - call generation (making a GET or POST request on a certain URL) - queue management (LOGIN / LOGOUT /

Hi the new asterisk 1.4 supports to store queue log information directly into a database? (like CDR) ? thanks

hi i'd like to write a simply application in php with phpAgi that: - connect to Asterisk - call an external number using a Zap channel - play a message here is some code: <?php $asm = new AGI_AsteriskManager(); if ($asm->connect()) { $asm->Originate("Zap/g1/1","number","default","1"); /* play message... */ } else {

sorry for top-posting, the two topics started with 2 different reason subject, but then we finished on the same problem. btw,the 2 show channel are reported above: the channel with DTMF working kcenter*CLI> core show channel SIP/pbx2-000004b9 -- General -- Name: SIP/pbx2-000004b9 Type: SIP UniqueID: 1467323106.1275 Caller ID: xxxx Caller ID Name: xxxx

Hi i've got an openvox a800p01 with 1 FXO and 4 FSX i've done the following: - downloaded zaptel-1.4.7.1 > >> - downloaded the file opvxa1200.c > >> - copied in zaptel-1.4.7.1/ > >> - edited makefile adding opvxa1200 in the modules and the voice > >> opvxa1200.o : zaptel.h wctdm.h > >> - edited zaptel.sysconfig adding MODULES="$MODULES

Hi can you help me to build a asterisk manager command event to join a conference? i've seen that there is the event Event: MeetmeJoin Channel: <channel> Uniqueid: <uniqueid> Meetme: <meetme> Usernum: <usernum> Can you explain me how it works? Can i use it to join an existing conference? Thanks

Hi i have a Te205P connected to a PRI E1, can i force the outgoing callerid to change for each context? for example: [outgoing_context_one] ;force callerid to 12345 exten => _XXXXXXXXXXX,1,Dial(Zap/${EXTEN}) [outgoing_context_two] ;force callerid to 22222 exten => _XXXXXXXXXXX,1,Dial(Zap/${EXTEN}) Can i do that? thanks to all -- /*************/ nik600

Hi i've read this post http://lists.digium.com/pipermail/asterisk-dev/2007-May/027666.html I just want to know if there are some upgrades... on 1.4 or 1.2. I'd like to store two records in the CDR instead of one, when a call is transferd. Is it possibile now? Thanks to all -- /*************/ nik600 https://sourceforge.net/projects/ccmanager

I've connected some analogic phone to some fxs modules on an analogic card. I want to disable by default the call waiting sound. I know that dialing *70 before to call the call waiting is disabled until the next call, but isn't there a setting or a dialplan command to set up this automatically? Thanks -- /*************/ nik600 https://sourceforge.net/projects/ccmanager

this is the point, and the strange thing: DTMF is set to rfc2833, but is working both on incoming and outgoing calls, it is not working only on calls generated with the Originate AMI command, or with the queue member that point to Local dialplan, as you suggested 2016-06-30 22:53 GMT+02:00 John Kiniston <johnkiniston at gmail.com>: > Looking at your logs it looks like you may need to

Dear all due to some custom requirements we are planning to use the manager events for creating some custom reports. I've enabled cdr_manager, then in manager.conf i've enabled timestampevents = yes and in queue.conf eventmemberstatus = yes. I know that these settings can generate a lot of manager events but i'm planning to have a very simple application on the Asterisk server that

CCMANAGER 0.5 released!! NOTE: this is a previous alpha release, maybe there is some customization to do on the settings files, i can't write a clear and complete howto at the moment I don't have released upgrades in the last months but the project is still alive i'm too busy at the moment, i'm following other projects to have some resources (both money and time) and then i can

When i made internal call into my LAN using x-lite sip phone client I retrive in askterisk CLI : ----------- ERROR ---------- Verbosity is at least 6 -- Remote UNIX connection -- Executing Dial("SIP/201-979d", "SIP/201|60|t") in new stack -- Called 201 May 29 18:09:28 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call

Dear R users, I am encoutering the following problem: I have a dataset with a 'unique_id' and different 'visit_date' (formatted as.Date, "%d/%m/%Y") per unique_id. I would like to create a new variable with the most recent date of visit per unique_id as shown below. unique_id visit_date last_visit_date 1 01/06/2010 01/06/2011 1 01/01/2011 01/06/2011 1