[asterisk-users] Two lines for outgoing calls

Dear All, I'm using Asterisk 1.4.16.2 with Zaptel 1.4.7 on Debian with kernel 2.6.18. I have two analog lines Zap/1 and Zap/2 as group 1 in zapata.conf. I'm using below context for dialing out. [outbound-local] exten = _9XXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten =

[asterisk-users] Two lines for outgoing calls

Dear All, I'm using Asterisk 1.4.16.2 with Zaptel 1.4.7 on Debian with kernel 2.6.18. I have two analog lines Zap/1 and Zap/2 as group 1 in zapata.conf. I'm using below context for dialing out. [outbound-local] exten = _9XXX,1,Dial(Zap/g1/${EXTEN:1},30,tTr) exten =

Re: [asterisk-users] Two lines for outgoing calls

On Wed, 2007-12-26 at 11:36 -0500, Steve Totaro wrote: Dominik Zalewski wrote: Dear All, I'm using Asterisk 1.4.16.2 with Zaptel 1.4.7 on Debian with kernel 2.6.18. I have two analog lines Zap/1 and Zap/2 as group 1 in zapata.conf. I'm using below context for dialing out

[asterisk-users] Dialout Macro and transfer call in progress

Dear All, I can not transfer call in progress. What's wrong with my macro? I think tT flags is enough right? [macro-stdexten] exten = s,1,Set(temp=${DB(CFU/${ARG1})}) ; Get CFU key exten = s,2,Set(DNDStatus=${DB(DND/${ARG1})}) ; Get DND key exten = s,3,GotoIf($[${temp} = ]?5); If

Re: [asterisk-users] Call transfer in asterisk

me suggestion for my solution Regards Satish Patel - Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://www.voip-info.org/wiki/index.php?page=PBX+CallTransfer -- Dominik Zalewski | System Administrator OpenCraft t- +2 02

[asterisk-users] Transfer outgoing call - macro

Dear All, I have a problem with call transfer. When I dial a number and then I want to transfer current call to an extension, I'm getting disconnected. Transfering incoming call works fine. I'm using macro for dialing. extensions.conf: [from-internal] ignorepat = 9 exten =

Re: [asterisk-users] VPN on Asterisk

mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dominik Zalewski | System Administrator OpenCraft t- +2 02 336 0003 w- http://www.open-craft.com ___ --Bandwidth and Colocation provided

Re: [asterisk-users] VPN on Asterisk

I'm in Middle East also and don't have problems with SIP and IAX2:) Try OpenVPN. It's easy to setup and has many features. -- Dominik Zalewski | System Administrator OpenCraft t- +2 02 336 0003 w- http://www.open-craft.com ___ --Bandwidth and Colocation

Re: [asterisk-users] WiFi SIP phones

-541 (g729) with asterisk 1.2.x without any problems. Battery life is poor but the range is acceptable. I tried also Zyxel WiFi SIP phone but I wasn't able to register to my pbx. It restarts everytime. Probably a firmware problem or something. Dominik Zalewski | System Administrator OpenCraft

[asterisk-users] Asterisk 1.2.17 and BRIstuff - SOLVED

Hi All, I'm using Asterisk 1.2.17 and I would like to install BRIstuff. Problem is that current version of BRIstuff is for Asterisk 1.2.14. BRIstuff 0.3.0-PRE-1y (* 1.2.14) If I'm misunderstanding how to apply patches for 1.2.17? Thank you in advance, Dominik I found it:)

[asterisk-users] Asterisk 1.2.17 and BRIstuff

Hi All, I'm using Asterisk 1.2.17 and I would like to install BRIstuff. Problem is that current version of BRIstuff is for Asterisk 1.2.14. BRIstuff 0.3.0-PRE-1y (* 1.2.14) If I'm misunderstanding how to apply patches for 1.2.17? Thank you in advance, Dominik

[asterisk-users] MP3Player

Hi All, I'm having problem with MP3Player app. I want the caller to hear mp3 when he is waiting until I answer my phone. -- from extentions.conf -- exten = 200,1,Answer() exten = 200,2,MP3Player(/home/user200/mp3/hanna-hais.mp3) exten = 200,3,Dial(SIP/200|20|tTrR) exten = 200,4,Hangup() --

[asterisk-users] Asterisk PPPD with analog lines

Hi All, Is it possible to use asterisk as a internet link backup callback solution? I mean when my main DSL link is down at my server room I would like to dial to asterisk , then it will call back me and provide a connection to a LAN network. Regards, Dominik

[asterisk-users] Call forwarding

Hi All, I'm using asterisk 1.2.15 and call forwarding doesnt work for me. from my extensions.conf: ; Unconditional Call Forward exten = _*21*X.,1,NoCDR exten = _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4}) exten = _*21*X.,3,Playback(vm-saved) exten = _*21*X.,4,Hangup exten =

Re: [asterisk-users] Call forwarding

On Thursday 15 February 2007 03:22:58 pm Stefan Wintermeyer wrote: Am 15.02.2007 um 14:06 schrieb Dominik Zalewski: exten = _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4}) Just use ${CALLERID(num)} and not ${CALLERID(NUM)}. Stefan it didnt help :( Is there is other way

Re: [asterisk-users] Call forwarding

On Thursday 15 February 2007 04:00:52 pm Pavel Jezek wrote: you just post only call forward activation part of dialplan, but you must also make dialplan part, that reflect, how is set this callforward mark, ie. if callforward is set, dial that number, if not, dial peer... Do you have any

[asterisk-users] Can Asterisk handle 7000 SIP users?

Hi All, One of my customer asked me if Asterisk can handle 7000 SIP users. They want anyone that have access to wireless hotspot to make voice calls to the office using software phone or SIP cordless phone. Does anybody did such a setup? What are hardware requirements for server and how

[asterisk-users] Can not compile latest zaptel -1.2.13

I'm trying to compile latest zaptel-1.2.13 and I'm getting following errors: /usr/src/Asterisk-1.2.14/zaptel-1.2.13/xpp/xbus-core.c: In function ‘debugfs_open’: /usr/src/Asterisk-1.2.14/zaptel-1.2.13/xpp/xbus-core.c:171: error: ‘struct inode’ has no member named ‘i_private’ make[5]: ***

Re: [asterisk-users] Can not compile latest zaptel -1.2.13 - SOLVED

If you look a number of lines above the function debugfs_open, you'll see: /* * As part of the inode diet the private data member of struct inode * has changed in 2.6.19. However, Fedore Core 6 adopted this change * a bit earlier (2.6.18). If you use vanila kernel (or Debian Etch) *

[asterisk-users] Asterisk 1.2.14 and bristuff 0.2.0-RC8s

Hi All, How to install bristuff on asterisk 1.2.14? install scripts are trying to download and compile those versions: asterisk-1.0.10 zaptel-1.0.10 libpri-1.0.9 and I'm running: asterisk-1.2.14 zaptel-1.2.12 libpri-1.2.4 I only need Pickup application from bristuff to be able to pickup

[asterisk-users] Enhanced PickupChan

Hi All, I've installed Enhanced PickupChan on asterisk 1.2.14 using howto from http://www.thorsten-knabe.de/linux/asterisk/pickup.jsp . from extensions.conf: exten = 0,1,Dial(SIP/eosoiris|20|tTrR) exten = 200,1,Dial(SIP/dzalewski|20|tTrR) exten = _7.,1,Pickup2(${EXTEN:1}) When I try to

[asterisk-users] Asterisk 1.2.14 bristuff app_pickup.so

Hi All, I'm using Asterisk 1.2.14 with Wildcard TDM400P. I need app_pickup.so application so I can pickup channel-independent calls from any IP Phone headset. How to compile and install only this application from bristufff package? Thank you in advance, Dominik

[asterisk-users] Pickup() ringing extension and call waiting

Hi All, I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would like to be able to pickup ringing extention from any SIP phone using Pickup() application. from my dial plan: [incoming] exten =

Re: [asterisk-users] Pickup() ringing extension and call waiting

On Monday 29 January 2007 03:20:16 pm Steve Davies wrote: On 1/29/07, Dominik Zalewski [EMAIL PROTECTED] wrote: Hi All, I'm using Asterisk 1.2.14 under openSuSE 10.2 with kernel 2.6.18. I have Wildcard TDM400P card and D-Link DPH-120S and DPH-140S SIP phones. I would like to be able