[asterisk-users] Playback on h exten

') == Spawn extension (play, s, 2) exited non-zero on 'SIP/300-0045' This is my dialplan: [from-test] exten = _X.,1,Dial(SIP/${EXTEN},60,rjtTg) exten = h,1,Goto(play,s,1) [play] exten = s,1,Noop(play) exten = s,2,Saydigits(123579) Anyone can help me? Thanks Enrico. -- -- Pasqualotto Enrico

Re: [asterisk-users] Playback on h exten

Yes, correct now it works for Dial. I think is the same with c option on Queue, do you think there's a way to do it on h exten? My goal is to inject my dialplan on hangup macro. Enrico. - Messaggio originale - If you choose to go with the Dial command and use the g option, you

Re: [asterisk-users] Logging Asterisk console

On Tue, 2009-04-07 at 15:21 +0200, Marco Sambo wrote: Hi Enrico, I do that by modifying logger.conf [logfiles] logpro = notice,warning,error,debug,verbose and modifying asterisk.conf [directories] astetcdir = /etc/asterisk astmoddir = /usr/lib/asterisk/modules astvarlibdir =

[asterisk-users] Logging Asterisk console

Hi all, in witch way can I put in a log file the asterisk console? I have tried with some settings in file logger.conf but the log not contain the same debug that I can see with asterisk -rvvv. I need it in debugging purpose for tracking some bug. Thanks Enrico. smime.p7s Description: S/MIME

[asterisk-users] Asterisk BLF to Cisco CME

Hi all, I'm searching for a way to inform my Cisco CME that a number on Asterisk server is busy. I have a SIP trunk between Cisco and Asterisk and some Cisco ip phone have a speed dial with a number registered on Asterisk. How can I exchange busy information between two PBX? Thanks Enrico.

[asterisk-users] app transfer problem

I all, I'm trying to transfer a iax2 channel trought dialplan app transfer to another extensions (IAX). The variable TRANSFERSTATUS report SUCCESS but the call isn't trasfered. I haven't other information, in console I see only hangup of a channel. My scenario is 3 asterisk box connected with iax

[asterisk-users] Fax issue over cisco gateway

Hi all, I'm trying to send fax from Hylafax to a remote fax machine through Asterisk and cisco 2801 as E1 gateway. This is my architecture: sendfax - HylaFax - iaxmodem - Asterisk - (SIP) 2801 with E1 card For incoming fax I don't have any problem, but I'm not able to send fax out of 2801. My

Re: [asterisk-users] Asterisk Realtime with MySQL Registration Failed

On Mon, 2008-08-04 at 16:48 +0300, Abid Saleem wrote: May be. If somebody has experience this problem before, then only he/she can guide about this. I am not sure whats going on. Abid Saleem Try to set debug verbose option in logger.conf, then check all query from asterisk to mysql for see

[asterisk-users] wait pickup

Hi all, One question I have set in the extensions.conf of my asterisk that all incoming call go in the wait application because I need to not connect the caller but remain in the ringing state. After that the call is on the wait exten for a N second I need from other sip phone to pickup this

Re: [asterisk-users] wait pickup

On Thu, 2008-07-03 at 09:31 -0500, Eric ManxPower Wieling wrote: chan_iax2 does not support pickup (callpickup=, pickupgroup= and *8). Wow! It's a very nice problem And for redirect a call in wait state to a sip phone? Without pickup ... Channelredirect don't work with ringing channel for

[asterisk-users] Conference Hangup

Hi all, I have a question on asterisk conference. Now I use appl Meetme with option A x because when a marked person hangup I want to close all the conference. But what I have to do if I want two marked person and kill the conference when one of two hangup? Is possible? Thanks. Enrico

[asterisk-users] Realtime context

context I have to modify the extensions.conf with: [newcontext] switch = Realtime/@ but I have about 50 asterisk that read one database, now if I want to change/add a context I have to change 50 extensions.conf file :( Thanks Enrico. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web

[asterisk-users] Asterisk translator issue?

to keet one file but is interesting to know who want to translate who. Thanks Enrico. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org http://www.linkedin.com/in/epasqualotto smime.p7s Description: S/MIME Cryptographic Signature

[asterisk-users] Park problem on IAX2 channel

came across the iax channel and arrive to other asterisk. The are a way to block this dtmf across the IAX trunk? Thanks Enrico. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org http://www.linkedin.com/in/epasqualotto smime.p7s Description: S/MIME

[asterisk-users] Dial and option G

(,qdxAa) exten = ,2,MeetMe(,qdx) exten = ,3,Dial(other-user,,G(from-iax,,4)) exten = ,4,MeetMe(,qdx) but not work. Any suggestion? Thanks Enrico -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org skype: epasqualotto

[asterisk-users] mISDN problems

Hi all, we're buildin an Asterisk box based on an Intel IXP425 board. The board uses a Beronet BN2S0 ISDN card, mISDN 1.1.4 and asterisk 1.4.2. hfc_multi has been patched to compile under big endian cpu, and so also capi kernel files. All the modules seem to load correctly (configuration was made

[asterisk-users] Beronet card - issue?

jitterbuffer_upper_threshold=0 hdlc=no [isdn] ports=1 context=from-pstn msns=* This is the first time that I configure this type of card Link of some good docs is ok too. :) Enrico. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org skype: epasqualotto

Re: [asterisk-users] Using meetme like call

Rob Schall wrote: One easy way to get close to this affect: Create a group dialDial(SIP/1000SIP/1001) then have a dynamic meetme room generating extension. This way, you can put them on hold for a brief second, dial that extension, create a room, then transfer them into it. This keeps

[asterisk-users] Using meetme like call

a way to make ring two phone and enter in the conference in the same time? Thank Enrico. -- Pasqualotto Enrico Netspin srl mail: [EMAIL PROTECTED] cell: 347 3292620 web: www.netspin.it smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth

[asterisk-users] Using meetme like call

a way to make ring two phone and enter in the conference in the same time? Thank Enrico. -- Pasqualotto Enrico Netspin srl mail: [EMAIL PROTECTED] cell: 347 3292620 web: www.netspin.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk

[asterisk-users] IAX Realtime - show peers works?

hi all, I'm trying to set up some iax2 trunks in Realtime architecture with the same backend. All work better (make call, receive etc etc) but when I do iax2 show peers some asterisk don't show anything and other show the iax2 peers but with status unknow. Name/UsernameHost

[asterisk-users] Asterisk CME integration using h323

]: chan_h323.c:1479 cleanup_connection: Avoiding H.323 destory deadlock on ip$192.168.99.2:53716/21 localhos== H.323 Connection deleted. I don't understand why the call goes down only from cisco to asterisk any ideas? Thanks Enrico -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web

[asterisk-users] *****SPAMZ***** Conference Page question

Spam detection software, running on the system placebo, has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see [EMAIL PROTECTED] for details. Content

[asterisk-users] Conference Page question

Hi. I'm currently setting up a particular conference: 3 members (a,b,c), a can speak with b and c, b and c can speak only with a and not between them. I found my possible solution with paging/intercom using option d (full-duplex), but I need to make ringing the phone in intercom. Now I set

[asterisk-users] *****SPAMZ***** Asterisk cluster - keep up connection?

Spam detection software, running on the system placebo, has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see [EMAIL PROTECTED] for details. Content

[asterisk-users] PHP sip client

Hi all, I want to write a simit sip client in PHP with asterisk API, in this moment I'm able to compose a number on my browser and call between 2 hw sip phone. I digit a number, my phone ring and after hanging up the cornet the second phone ring. But I want to add a features I want to

[asterisk-users] Asterisk HA

Hi all, I have to make for a client an asterisk system for process up to 250 calls between conference and normal call. At disposition I have 4 xserver 346 with dual xeon 3.0Ghz and the client require a failover system. Anyone have experience for this type of solution? Is better ultramonkey,

[asterisk-users] qualify=yes

hi all, how can I set the interval in second from retrasmit the magic packets when qualify is set to on? I want to view whitch voip-phone is connected but I don't want to DOS my adsl connection ;) Thanks Enrico P. -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web:

Re: [asterisk-users] qualify=yes

Enrico Pasqualotto wrote: hi all, how can I set the interval in second from retrasmit the magic packets when qualify is set to on? You have to set qualify=second instead of qualify=yes|no. Eheheheh -- Pasqualotto 'Pasqu' Enrico enrico AT pasqualotto DOT org web: http://www.pasqualotto.org

Re: [asterisk-users] [OT] Nokia E60/61/70 and SIP

Martin Joseph wrote: For all of us using these devices, I have some good news. There is a self installable firmware update available from Nokia here (requires windows box to install): http://www.nokia.co.uk/nokia/0,1522,,00.html?orig=/softwareupdate This seems to radically improve the

[asterisk-users] Bad number - is not in inbound speed dial

Hi, what mean this voice message that asterisk say when I try to call an extension of another asterisk connected by IAX2 trunk? This problem exist only if I call from asterisk1 to asterisk2, vice versa all work. ___ --Bandwidth and Colocation provided

[asterisk-users] IAX2 trunk problem

Hi at all, I have make a IAX2 trunk over openvpn between [EMAIL PROTECTED] and trixbox. [EMAIL PROTECTED] have extension 200 to 299 and Trixbox 300 to 399 In my 2 box I set in outbound routing that if I call 7|XXX I want to use the IAX trunk. The call from Trixbox (ext 301) to [EMAIL PROTECTED]