search for: nik600

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

Hi., do you think that sbr policy in queue strategy will be useful? Bye ---------- Forwarded message ---------- From: nik600 <nik600 at gmail.com> Date: Sat, 7 Mar 2009 15:21:14 +0100 Subject: add a new queue strategy: SBR To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com> Hi to all isn't there any plan to add the Skills Based Routing strategy in queues.conf? I think that it will be...

...st=dynamic ; This device registers with us username=202 ; Username to use when calling this device before registration limitonpeers = yes call-limit = 2 busy-level = 1 The directive busy-level is ignored.... I've also tried busy-limit but without any result... Thanks -- /*************/ nik600 http://www.kumbe.it

...ueue Data: (Outgoing Line) Blocking in: ast_waitfor_nandfds the only difference i see is the "1st File Descriptor" pointing to -1 2016-06-30 23:29 GMT+02:00 Steve Edwards <asterisk.org at sedwards.com>: > Please don't top post. > > On Thu, 30 Jun 2016, nik600 wrote: > > 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 su...

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

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

...do a make opt: root@asterisk:/data/programmi/asterisk_1.2.18/asterisk-1.2.18/channels/h323# make opt make: *** No rule to make target `opt'. Stop. why? where am i wrong? i've also tried the last version of pwlib and openh323, but without fixing the problem thanks -- /*************/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser

...same => n,Playback(beep&cancelled&goodbye) > same => n,Set(MACRO_RESULT=BUSY) ;Reject the call > same => n,Hangup() > same => n,MacroExit() ;Return > > > On Thu, Jun 30, 2016 at 6:08 AM, nik600 <nik600 at gmail.com> wrote: > >> Dear all >> >> i'm using an "old" Asterisk 1.6.2.9-2+squeeze12, and want to know if is >> possible to configure a scenario like this: >> >> 1) receive a call and put it on-hold in a queue (OK) >> 2)...

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've made some test calls, i've notice that a call of the duration of 1:29 minutes is recorded in the cdr database as 1:45 minutes, is it normal? i think that 15 seconds are too many... how can i correct this? thanks

...cause they fill-in lastdata and lastapp with "ResetCDR" So, what can i do? Is it better to do some customization to generate a CDR event on each dialplan step or is better to parse the logfile and extract the information needed? I'm using Asterisk 1.4.23.1 TIA -- /*************/ nik600 http://www.kumbe.it

...ultaneous call can i handle per server? each server will have: 4 core 3.0 Ghz 4 GB of RAM I need an aproximate sizing: 0-100 calls per server ? 100-200 calls per server ? 200-300 calls per server ? 300-400 calls per server? 400-500 calls per server? Thanks to all in advance -- /*************/ nik600 http://www.kumbe.it

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

...iston <johnkiniston at gmail.com>: > Looking at your logs it looks like you may need to modify your sip.conf, > Check with your provider as to what kind of DTMF they support and configure > sip.conf to use that type of signalling. > > > > On Thu, Jun 30, 2016 at 1:18 PM, nik600 <nik600 at gmail.com> wrote: > >> thanks John >> >> yeah, your approach is much siple, i've tried it but i'm not able do >> detect DTMF tones. >> >> it seems that on calls that i receive DTMF tones are handled correctly, >> but on calls gen...

...key is pressed redirect the current call to the channel2Link, connecting the call in queue with the remote number (?) Step 1,2,3 works properly but i'm not able to link the two channels, even using redirect,goto or pickupChan. Any idea or help will be appreciated! Thanks -- /*************/ nik600 http://www.kumbe.it -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160630/45cfa95c/attachment.html>

...") in new stack [Jun 30 21:56:56] WARNING[5617]: channel.c:2558 ast_waitfordigit_full: Unexpected control subclass '-1' -- User entered nothing. Any idea? if i call from number xxx to an extension that goes to testDTMF at cRETEUNICA it works properly. Thanks -- /*************/ nik600 http://www.kumbe.it -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160630/45708fd4/attachment.html>

...cessfully queued' /></response> </ajax-response> But i can't heard nothing on the channel, i've tried to send the tone both on channel and link, but with no results. If i use normal dtmf from keyboards they works properly. What can i check? Thanks -- /*************/ nik600 http://www.kumbe.it

...l have a virtual interface that will be switched from one to the other in case of hardware problem. The question is: can one server with those settings manage up to 200 simultaneous call? The server will receive SIP calls and forward them through a CISCO router. Thanks to all -- /*************/ nik600 http://www.kumbe.it

...er does a redial (after the hangup) the call is forwarded to xxx at 10.10.10.2 that is the wrong address. I've tried to force SIP_HEADER(CONTACT) in the dialplan with a Set but it seems that i can't due to security reason. Is it possible to avoid this problem? Thanks -- /*************/ nik600 http://www.kumbe.it

...me[|options][|URL][|announceoverride][|timeout][|AGI]) So, setting up an appropriate AGI script can i play an audio file (or create it with some tts) to the call? After the AGI script the call is linked with the operator even if there is an Answer into the AGI? Thanks to all -- /*************/ nik600 http://www.kumbe.it