[asterisk-users] EAGI script with missing audio on /dev/fd/3

hen i call, the script is executed and the call goes in queue, i can hear the MOH, the file /tmp/pippo is created but it is empty. Any idea or suggestion? PS: if i use the application monitor or MixMonitor the call is recorded correctly. I'm using Asterisk 1.6.2.9-2+squeeze12 Thanks -- /

Re: [asterisk-users] how to join 2 channels using AGI/AMI

On Fri, 1 Jul 2016, nik600 wrote: > > i've tried rfc2833,inband and info having the same behaviour in all >> situation. >> >> 2016-06-30 23:53 GMT+02:00 nik600 <nik...@gmail.com>: >> sorry for top-posting, the two topics started with 2 different >> reaso

Re: [asterisk-users] how to join 2 channels using AGI/AMI

ito,,1) exten =>01,n,SayDigits(${digito}) Any idea? 2016-07-01 0:13 GMT+02:00 nik600 <nik...@gmail.com>: > i've tried rfc2833,inband and info having the same behaviour in all > situation. > > 2016-06-30 23:53 GMT+02:00 nik600 <nik...@gmail.com>: > >> sor

Re: [asterisk-users] how to join 2 channels using AGI/AMI

i've tried rfc2833,inband and info having the same behaviour in all situation. 2016-06-30 23:53 GMT+02:00 nik600 <nik...@gmail.com>: > sorry for top-posting, the two topics started with 2 different reason > subject, but then we finished on the same problem. > > btw,

Re: [asterisk-users] how to join 2 channels using AGI/AMI

e only difference i see is the "1st File Descriptor" pointing to -1 2016-06-30 23:29 GMT+02:00 Steve Edwards <asterisk@sedwards.com>: > Please don't top post. > > On Thu, 30 Jun 2016, nik600 wrote: > > this is the point, and the strange thing:DTMF is s

Re: [asterisk-users] how to join 2 channels using AGI/AMI

Kiniston <johnkinis...@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 P

Re: [asterisk-users] problem with DTMF detection on calls created with Originate AMI command

i'm using Asterisk 1.6.2.9-2+squeeze12 2016-06-30 22:14 GMT+02:00 Richard Mudgett <rmudg...@digium.com>: > > > On Thu, Jun 30, 2016 at 3:00 PM, nik600 <nik...@gmail.com> wrote: > >> Dear all >> >> i'm creating an outgoing call to number xxx with this c

Re: [asterisk-users] how to join 2 channels using AGI/AMI

t; n,Playback(pls-wait-connect-call) > same => n,MacroExit();Return > > exten => REJECT,1,NoOP() > same => n,Playback(beep) > same => n,Set(MACRO_RESULT=BUSY);Reject the call > same => n,

[asterisk-users] problem with DTMF detection on calls created with Originate AMI command

bx2-04ad", "digito,,1") 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@cRETEUNICA it works p

Re: [asterisk-users] how to join 2 channels using AGI/AMI

oh, yes! Many thanks 2016-06-30 15:28 GMT+02:00 Guido Falsi <m...@madpilot.net>: > On 06/30/16 15:08, nik600 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 l

[asterisk-users] how to join 2 channels using AGI/AMI

(?) 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 -- _ -- Bandwidth and

Re: [asterisk-users] how to know status of asterisk from php

webinar every Thurs:               http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 http://www.kumbe.it

[asterisk-users] understand which asterisk thread is consuming CPU

93m 5596 S1 3.1 0:00.68 asterisk 15956 root 20 0 662m 93m 5596 S1 3.1 0:00.68 asterisk -- /*/ nik600 http://www.kumbe.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] 1.6 how to use groupcount and counteronpeer in queues to avoid ringinuse

-- /*/ nik600 http://www.kumbe.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello

[asterisk-users] [compat] section in asterisk.conf : compatibility with pipe delimiter

ignores the | delimiter, if i try with the comma it works. Reading the the upgrade file it seems that the pbx_realtime should affect also the extension.conf settings... where am i wrong? Thanks to all in advance -- /*/ nik600 http://www.kumbe.it

Re: [asterisk-users] problem with ringinuse=no, queue members receive randomly two calls

i've also tied this tests: - changed hardware - upgrade to 1.4.31 - kernel recompiled with 1000 Hz option - changed SO (Slackware 13) - run the system on hardware (no ESXi) But i've not resolved the problem. Do you have any idea? On Thu, May 6, 2010 at 11:54 AM, nik600 nik...@gmail.com wrote

Re: [asterisk-users] problem with ringinuse=no, queue members receive randomly two calls

i get may debug messages like this: DEBUG[30684] channel.c: Internal timing is disabled (option_internal_timing=0 chan-timingfd=-1) Is because dahdi is not installed? Can this be a possible cause of this behaviour? On Tue, May 4, 2010 at 9:54 PM, nik600 nik...@gmail.com wrote: Dear all

[asterisk-users] problem with ringinuse=no, queue members receive randomly two calls

setinterfacevar=yes eventwhencalled=yes eventmemberstatus=yes ringinuse=no member = SIP/PL1039 -- /*/ nik600 http://www.kumbe.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk

Re: [asterisk-users] problems originating an outgoing IAX2 call

I've resolved...it was a limitation of the provider for calls without a CallerID On Sun, Apr 18, 2010 at 7:43 PM, nik600 nik...@gmail.com wrote: Dear all i'm trying to originate an outgoing call with the command originate, from Asterisk's CLI i'm typing: CLI originate  IAX2/my-iax-provider

[asterisk-users] problems originating an outgoing IAX2 call

-- /*/ nik600 http://www.kumbe.it -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello

Re: [asterisk-users] Live Audio Streaming- From Aux interface-Online resource

Many thanks Jonathan! On Wed, Mar 31, 2010 at 10:29 AM, cov...@ccs.covici.com wrote: What is the significance of /dev/fd/3 where does it come from? I'ts the file descriptor 3 for the EAGI process, wich contains the audio. -- /*/ nik600 http://www.kumbe.it

Re: [asterisk-users] Live Audio Streaming- From Aux interface-Online resource

but the recording file is continuosly growing and ffmpeg ends the conversion before of the call completion. If you can give me a practical example i'll appreciate it a lot. Bye -- /*/ nik600 http://www.kumbe.it

[asterisk-users] distribuited ACD on many asterisk nodes

users information and thansfer call. Do you know if there is something similar somewhere ? Maybe Asterisk has already some magic sauce to do that ? ;-) Thanks to all -- /*/ nik600 http://www.kumbe.it -- _ -- Bandwidth

[asterisk-users] usage of manager events to create custom reports

this kind of approach - if someone else has done something similar and wants to share his experience - how much is affordable the events generation excpecially in system with a high load Thanks to all for any contribute. Hi -- /*/ nik600 http://www.kumbe.it

Re: [asterisk-users] how to announce the agent answering in a queue to the caller

I've tested and confirm that the AGI script can do that. i had to enable setinterfacevar=yes in the queue conf and then can read the MEMBERINTERFACE channel variable. Just because it can be useful for someone else. On Fri, Oct 23, 2009 at 9:44 PM, nik600 nik...@gmail.com wrote: Hi to all

[asterisk-users] how to announce the agent answering in a queue to the caller

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

Re: [asterisk-users] wrond DTMF detection on Zap channel

-complete-2.2.0.2 libpri-1.4.10.1 Any idea? On Tue, Oct 13, 2009 at 11:51 PM, nik600 nik...@gmail.com wrote: for disabling the hardware DTMF you intend to recompile zaptel with vpmdtmfsupport=0? Thanks On Fri, Oct 9, 2009 at 6:54 PM, C F shma...@gmail.com wrote: are you using chan_local

Re: [asterisk-users] wrond DTMF detection on Zap channel

for disabling the hardware DTMF you intend to recompile zaptel with vpmdtmfsupport=0? Thanks On Fri, Oct 9, 2009 at 6:54 PM, C F shma...@gmail.com wrote: are you using chan_local? try disabling the hardware DTMF. Sent using my wired Blueberry. On 10/9/09, nik600 nik...@gmail.com wrote

[asterisk-users] live audio streaming using monitor, mixmonitor or chanspy

Hi to all, is it possible to setup a live audio streaming in Asterisk using for source monitor, mixmonitor or chanspy? Thanks -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] wrond DTMF detection on Zap channel

On Fri, Oct 9, 2009 at 6:54 PM, C F shma...@gmail.com wrote: are you using chan_local? try disabling the hardware DTMF. Sent using my wired Blueberry. On 10/9/09, nik600 nik...@gmail.com wrote: Dear all i have a TE205P connected to an Asterisk 1.2.18. Yes i know, the version is old but since

[asterisk-users] wrond DTMF detection on Zap channel

as i get the wrong dtmf tone directly from Asterisk. It's not a phone problem as the same phone may retry and then it works. Is it possible to relate it with the load of the server? Can you suggest me something? Thansk -- /*/ nik600 http://www.kumbe.it

Re: [asterisk-users] put some IVR into a queue after the call queuing

any interest in it? I'm evauating to add this feature but before to do that i'd like to know if there is some other approach that can avoid some developement. Regards On Wed, Sep 30, 2009 at 12:48 PM, nik600 nik...@gmail.com wrote: Dear all is it possible to handle a queue using a programmed

[asterisk-users] put some IVR into a queue after the call queuing

) And then manually match information between unique ID and queue_log to consider info on queue A,B,C,D, as a single queue. Or is there some magic sauce to specify an IVR script that is executed when a call is in a queue? Thanks -- /*/ nik600 http://www.kumbe.it

[asterisk-users] the correct way to setup a transfer with REFER in SIP

? My aim is to make a REFER to b...@test and free completely Asterisk. Thanks to all in advance, bye. -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

Re: [asterisk-users] problem with transfer application (REFER)

... Bye On Fri, Jun 12, 2009 at 2:45 PM, Giorgio Incantalupogincantal...@fgasoftware.com wrote: Hi nik600, I had some trouble transferring calls with that version of Asterisk even if I used the normal transfer via features.conf. Upgrading to 1.4.24 helped a bit (even if not completely). My advice

[asterisk-users] problem with transfer application (REFER)

/ configuration to use a complete and stable implementation of the REFER functionality? Thanks to all in advance -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing

Re: [asterisk-users] Fwd: add a new queue strategy: SBR

, 2009 at 3:57 PM, Florian Hackenberger f.hackenber...@chello.at wrote: On Sunday 08 March 2009 17:11:33 nik600 wrote: Hi to all isn't there any plan to add the Skills Based Routing strategy in queues.conf? I think that it will be enough to add an int skill to the struct member and then order

[asterisk-users] what can we do with lost voice packet on a congestioned VPN?

difference. Is there something else that i can do? Thanks to all -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

Re: [asterisk-users] log to cdr each dialpan action, not only one record for each call

that? -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] log to cdr each dialpan action, not only one record for each call

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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk

Re: [asterisk-users] log to cdr each dialpan action, not only one record for each call

On Thu, Mar 12, 2009 at 8:13 PM, Matt Riddell li...@venturevoip.com wrote: On 13/03/2009 8:02 a.m., nik600 wrote: Hi to all. What can i do if a customer needs to log in the CDR all the dialpan actions related to a call? I mean, not only the lastapp e the lastdata but all the dialpan actions

Re: [asterisk-users] log to cdr each dialpan action, not only one record for each call

application call... well, this will be surely the best. I'll read the documentation and let you know, thanks. -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users

Re: [asterisk-users] log to cdr each dialpan action, not only one record for each call

, can you tell me where you have to place the code to log when an app is called? Thanks -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] Fwd: add a new queue strategy: SBR

On Tue, Mar 10, 2009 at 4:01 AM, James Sneeringer jsnee...@gmail.com wrote: On Mon, Mar 9, 2009 at 5:44 PM, nik600 nik...@gmail.com wrote: Thanks, i've tested and it works (1.4.23.1). Just 2 questions: 1) this approach seems to be an hack and not the implementation of a feature is it really

Re: [asterisk-users] Fwd: add a new queue strategy: SBR

that? maybe i should open a new post but i think that this kind of approach isn't much better than the callback functionality, what do you think about that? -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Fwd: add a new queue strategy: SBR

. Thanks for your time. Bye -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman

[asterisk-users] Fwd: add a new queue strategy: SBR

Hi., do you think that sbr policy in queue strategy will be useful? Bye -- Forwarded message -- From: nik600 nik...@gmail.com Date: Sat, 7 Mar 2009 15:21:14 +0100 Subject: add a new queue strategy: SBR To: Asterisk Developers Mailing List asterisk-...@lists.digium.com Hi to all

Re: [asterisk-users] Fwd: add a new queue strategy: SBR

but priority are se to the call, not to the agent! or am i wrong? On Sun, Mar 8, 2009 at 5:32 PM, David fire ddf...@gmail.com wrote: the queue already have prioritys. David 2009/3/8 nik600 nik...@gmail.com Hi., do you think that sbr policy in queue strategy will be useful? Bye

Re: [asterisk-users] put the hostname of asterisk in the callerid uri to avoid NAT problems

it. The rtp traffic is redirect correctly but the SIP INVITE contains the ip of the lan and not of the nat. I'll try with SipAddHeader and then let you know... thanks -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation

Re: [asterisk-users] put the hostname of asterisk in the callerid uri to avoid NAT problems

On Sat, Feb 7, 2009 at 8:31 AM, nik600 nik...@gmail.com wrote: hi is it possible to set up in the dialplan (on in sip.conf, or something else) the hostname of the outgoing uri call? This is my scenario: - CCM integrated with Asterisk via h323 - SIP user registerd to Asterisk - Asterisk

[asterisk-users] put the hostname of asterisk in the callerid uri to avoid NAT problems

) the call is forwarded to x...@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

[asterisk-users] server sizing for ~ 200 simultaneous call

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 ___ -- Bandwidth

Re: [asterisk-users] problem with PlayDTMF: no error but no tone

otherwise. It means that rfc2833 was offered, but doesn't work! Well, info and inband works. Bye On Thu, Jan 22, 2009 at 11:18 AM, nik600 nik...@gmail.com wrote: Is there the possibility to increase the debug of an AJAM command? If DTMF works on channel, and my command is queued successfully

Re: [asterisk-users] problem with PlayDTMF: no error but no tone

Is there the possibility to increase the debug of an AJAM command? If DTMF works on channel, and my command is queued successfully, what can be the problem? Thanks On Thu, Jan 15, 2009 at 4:34 PM, nik600 nik...@gmail.com wrote: Hi to all i'm using PlayDTMF with AJAM, after the authentication

[asterisk-users] problem with PlayDTMF: no error but no tone

' //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

[asterisk-users] problem with dahdi and meetme

need DAHDI services, you must correctly configure DAHDI. Where am i wrong? Thanks -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE

Re: [asterisk-users] problem with dahdi and meetme

PS: asterisk is compiled with dahdi support On Mon, Jan 12, 2009 at 1:39 PM, nik600 nik...@gmail.com wrote: Hi to all. I'm trying to use meetme on asterisk 1.4.22.1. On a debian i've compiled (as i need h323 support) openh323_v1_18_0 pwlib_v1_10_0 dahdi-linux-2.1.0.3 dahdi-tools

[asterisk-users] asterisk 1.4 with h323 for debian

hi to all. Do you know if there is an asterisk 1.4 package with h323 support for debian? I've found this http://packages.debian.org/etch/asterisk-h323 but has asterisk 1.2.13. Thanks to all. -- /*/ nik600 http://www.kumbe.it

Re: [asterisk-users] busy-level / busy-limit Asterisk 1.4.22

sorry if i ask it again, but where can i find the patch for enable busy-level/limit in 1.4 ? thanks On Tue, Nov 18, 2008 at 12:09 PM, nik600 nik...@gmail.com wrote: Thanks, is it possibile to retrieve a patch from Asterisk trunk? how? On Tue, Nov 18, 2008 at 11:54 AM, Steve Howes st

Re: [asterisk-users] how to set the busy signal usign softphones

Ok, i've resolved, the problem was related to the sip type settings. It must be peer instead of fried. Bye On Fri, Jan 2, 2009 at 5:41 PM, nik600 nik...@gmail.com wrote: Thanks for your reply. Now, i use devstate too, but it doesn't work (or, maybe i suppose that it should work differently

Re: [asterisk-users] how to set the busy signal usign softphones

the user wants waiting calls or not and decide accordingly. __Yehavi: 2008/12/20 nik600 nik...@gmail.com On Sat, Dec 20, 2008 at 2:25 PM, Benoit maver...@maverick.eu.org wrote: Have you tried to set the call-limit to 10 or 2 for example, i know it's what's needed

[asterisk-users] how to set the busy signal usign softphones

method to limit the call available for a user to 1 but granting him the possibility to transfer a call? I know that there is the busy-level settings, but i'ts available only in 1.6. Thanks to all in advance. -- /*/ nik600 http://www.kumbe.it

Re: [asterisk-users] how to set the busy signal usign softphones

(correctly...up to 2 calls), even when he is busy. -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http

[asterisk-users] stream a file on a channel using AMI

Hi using AMI, is it possile to stream a file on a specific channel? Thanks to all in advance. -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list

Re: [asterisk-users] sip trunking and call transfer

do it from a SIP protocol perspective. I'm not sure to what extent Asterisk supports this capability. -- Raj Jain ok, thanks for your reply! I'll search about Asterisk SIP referer implementation. -- /*/ nik600 http://www.kumbe.it

Re: [asterisk-users] sip trunking and call transfer

Maybe my question is not clear or is too stupid? (sorry) Maybe this is already done in SIP trunking? Or (worste case) is impossible to do that? Thanks On Fri, Nov 21, 2008 at 8:53 AM, nik600 [EMAIL PROTECTED] wrote: Hi to all. i-ve got a question: what happen when a call between 2 trunks

[asterisk-users] sip trunking and call transfer

- Caller3 or b) Caller 1 - Trunk A/C - Caller3 So: is it possible to avoid the scenario a) ? Thanks to all -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing

[asterisk-users] busy-level / busy-limit Asterisk 1.4.22

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 ___ -- Bandwidth and Colocation

Re: [asterisk-users] busy-level / busy-limit Asterisk 1.4.22

Thanks, is it possibile to retrieve a patch from Asterisk trunk? how? On Tue, Nov 18, 2008 at 11:54 AM, Steve Howes [EMAIL PROTECTED] wrote: On 18 Nov 2008, at 10:30, nik600 wrote: the busy-level / busy-limit setting in sip.conf is available for Asterisk 1.4.22 ? http://www.voip-info.org

[asterisk-users] view the current calls and their codec

Hi to all. Is possible with the Asterisk 1.4 cli view the current calls and their codec? Thanks to all -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing

Re: [asterisk-users] view the current calls and their codec

. (218)727-4332 x105 - nik600 [EMAIL PROTECTED] wrote: Hi to all. Is possible with the Asterisk 1.4 cli view the current calls and their codec? Thanks to all -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation

Re: [asterisk-users] view the current calls and their codec

- nik600 [EMAIL PROTECTED] wrote: And if i have an h323 configuration? Thanks On Tue, Nov 11, 2008 at 4:17 PM, Tim Nelson [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] ~]# asterisk -rx 'sip show channels' assuming you want SIP... substitute sip for iax2 if you prefer... Tim

[asterisk-users] skype and Asterisk opensource integration

Hi to all except of some commercial hardware / software gateways, is there any opensource or free project to setup a Skype Account on Asterisk? Thanks to all -- /*/ nik600 http://www.kumbe.it ___ -- Bandwidth and Colocation Provided

[asterisk-users] problem with asterisk 1.4.21.1 and h323

-- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September

[asterisk-users] OpenH323 and ptlib version for asterisk 1.4.21.1

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

Re: [asterisk-users] OpenH323 and ptlib version for asterisk 1.4.21.1

/libpthread.so.0 #12 0xb731905e in clone () from /lib/libc.so.6 Can someone help me please? Thanks in advance to all On 7/17/08, Patrick [EMAIL PROTECTED] wrote: On Thu, 2008-07-17 at 19:34 +0200, nik600 wrote: Hi what version of openh323 and pwlib are suggested for asterisk 1.4.21.1.? Thanks to all

[asterisk-users] disable DTMF on a particular channel

Hi to all is it possibile (via AMI or dialplan) to disable the DTMF tone on a particular channel? Thanks in advance -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser

Re: [asterisk-users] play sound on a specific channel

i've seen that there is the PlayDTMF command. Bye On Tue, Jun 24, 2008 at 8:37 AM, nik600 [EMAIL PROTECTED] wrote: any idea? On Sat, Jun 14, 2008 at 9:50 AM, nik600 [EMAIL PROTECTED] wrote: Hi to all can i play a sound or a dtmf tone on a specific channel using AMI? Thanks to all

Re: [asterisk-users] play sound on a specific channel

any idea? On Sat, Jun 14, 2008 at 9:50 AM, nik600 [EMAIL PROTECTED] wrote: 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

[asterisk-users] retrieve the status of a sip user using AMI

Hi to all. How can i retrieve the status of a user using the subscription? For example, if i use: exten = 200,hint,SIP/200 exten = 200,1,Dial(SIP/200) After that, how can i retrieve the status of the SIP/200 user using AMI ? Thanks to all in advance -- /*/ nik600 https

[asterisk-users] play sound on a specific channel

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

[asterisk-users] use of AJAM wth high load

-- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing

[asterisk-users] Patch for app_asr.c: DTMF instead of goto

will hear (and Asterisk can detects, via AGI or dialplan) 200,300,400 DTMF tones. You can find more information here. http://www.kumbe.it/pagine/dettaglio/34/206.html Bye -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https

Re: [asterisk-users] asterisk virtualization on VMWARESX infrastructure

on a single machine, i intend an enterprise SX infrastructure with multiple nodes and auto failover policy. If Asterisk doens't suffer a virtualization, a service virtualized on a solid infrastructure is more scalable and hardware independent -- /*/ nik600 https://sourceforge.net/projects

[asterisk-users] upgrade of asterisk .... to what?

to determine what is beta and what is stable? Thanks -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] upgrade of asterisk .... to what?

On Thu, May 22, 2008 at 12:31 PM, Gordon Henderson [EMAIL PROTECTED] wrote: On Thu, 22 May 2008, nik600 wrote: Hi to all i'm managing a call center with 20 operators using Asterisk. I'm still using Asterisk 1.2.x as i love his stability. Now, i'm planning to migrate to 1.4.x, but i don't

[asterisk-users] how to retrieve sip tag from dialplan

Hi to all is it possible to retrieve the sip tag (server side) of a sip call from the dialplan? Thanks. -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser

Re: [asterisk-users] Need comments on CRM development / Asterisk Customization

and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker

[asterisk-users] disable call waiting by default

? Thanks -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk

Re: [asterisk-users] OpenVox A800P01 and ZT_CHANCONFIG failed

On Dec 24, 2007 8:07 PM, Darrick Hartman [EMAIL PROTECTED] wrote: Tzafrir Cohen wrote: On Mon, Dec 24, 2007 at 05:11:44PM +0100, nik600 wrote: maybe i've guess the problem! on the same server, i've got a B800P. I've tried to manually remove all isdn module and zaptel modules

Re: [asterisk-users] OpenVox A800P01 and ZT_CHANCONFIG failed

On 12/23/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Dec 23, 2007 at 07:05:37PM +0100, nik600 wrote: 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

Re: [asterisk-users] OpenVox A800P01 and ZT_CHANCONFIG failed

maybe i've guess the problem! on the same server, i've got a B800P. I've tried to manually remove all isdn module and zaptel modules. After that, i've done modprobe zaptel modprobe opvxa1200 and now the card has been correctly registered! On Dec 24, 2007 2:32 PM, nik600 [EMAIL PROTECTED

[asterisk-users] OpenVox A800P01 and ZT_CHANCONFIG failed

-- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list

Re: [asterisk-users] OpenVox A800P01 and ZT_CHANCONFIG failed

interface Subsystem: Unknown device 9100:0001 Flags: bus master, medium devsel, latency 32, IRQ 10 I/O ports at a800 [size=256] Memory at f800 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 -- /*/ nik600

[asterisk-users] OpenVox B800P and asterisk 1.4/ mISDN-1_1_7

/mISDN-1_1_7/drivers/isdn/hardware/mISDN/stack.c:235 st(0100) addr(41000100) layer -1 out of range * DMESG Can you help me to guess the problem? Thanks -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https

[asterisk-users] new Asterisk installation with openvox 1.2 or 1.4?

problems with misdn drivers. Thanks -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api

[asterisk-users] store 2 separate records in cdr when a call is transferd

-- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list

Re: [asterisk-users] store 2 separate records in cdr when a call is transferd

for blind transfer! Many thanks! On Nov 20, 2007 2:24 PM, Atis Lezdins [EMAIL PROTECTED] wrote: nik600 wrote: 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

[asterisk-users] Call center manager for Asterisk (Release 0.5)

instructions for the creation of queue_stats table - added the files view.sql bye -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/reportmaker https://sourceforge.net/projects/nikstresser

[asterisk-users] multiple iax users on the same host

for each iax account? Thanks -- /*/ nik600 https://sourceforge.net/projects/ccmanager https://sourceforge.net/projects/nikstresser ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list

  1   2   3   >