[Asterisk-Users] Asterisk 2.0 Where to download

Hello All I read in www.sineapps.com have Asterisk 2.0 rewritten C# and run on windows, any body could be mail or send to me URL to download.ThanksTin Trung NguyenTechnical TeamMobile: 084-91.365.4857website: www.daivietcontrol.net___

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 15, Issue 28

Hello All Anybody had used ooH323 for asterisk i using ooH323-0.7.2 and asterisk CVS may 2005. OpenH323 1.17.1 and pwlib 1.9.0 and GNUGK 2.0.2 audio is very good, better than SIP and IAX, but i have problem. how to router call from openh323 to outside PSTN. my h323.conf setting ; Objective

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 14, Issue 70

Hello All any body have used SS7 run with asterisk. could you like tell me how to download driver of SS7 and how to use it. Thanks___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 14, Issue 22

Hi All I have problem with LIBMFCR2 for once Exchange I using Sangoma card, the firstly. my ssystem run successful with MFCR2, connected to E10 (Acatel Exchange), after that, i move connection connect to EWSD (Siemens), my system don't work. error protocol R2. my system: Asterisk CVS 1.1.X

[Asterisk-Users] Sangoma card problem with EWSD Exchange

Hello All. I'm using sangoma card A-101. tested successful with E10 (ACATEL) Exchange,connection with E1, CAS, (using unicall-0.0.3pre4). my systemrun success,incoming call and call out are good. when iswitch to EWSD (SIEMENS) R-15. my asterisk faill, cannot connect with EWSD. (E10 and EWSD

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 14, Issue 1

Hello All. I'm using sangoma card A-101. tested successful with E10 (ACATEL) Exchange, connection with E1, CAS, (using unicall-0.0.3pre4). my system run success, incoming call and call out are good. when i switch to EWSD (SIEMENS) R-15 . my asterisk faill, cannot connect with EWSD. (E10 and EWSD

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 13, Issue 131

Hello I have problem with transfer call if using ACD When i using ACD with agent and queue setting, i cannot monitor call and transfer call. this's my setting - i have 2 IAX phone (phone number as 201, 202), agent.conf agent = 1001,4321,member 1agent = 1002,4321,member 2agent = 1003,4321,Tin then,

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 13, Issue 131

Hello how to calculator billing exactly when IAX accept the call, my configure customer -- telco --- asterisk -- ACD -- IAX at time, for example: 11:00 i dial to asterisk 11:01 asterisk answer channel and dial to IAX phone (11:02) ring 20 second (at 11:22). when IAX answer call (11:22) and talk 10

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 13, Issue 123

Hello All I want to setting my asterisk with features as belows: - When dial to system, caller hear music and message "welcome to asterisk Open PBX". - then when extension (IAX, SIP) is available, ringing on channel. - when channel answer call (offhook), at the time, i want to Host Exchange

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 13, Issue 86

Hello All i need to transfer CDR data from linux to MS SQL Serever (on Windows). writing by Perl. I have download and install UnixODBC, DBI, DBD from CPAN, when i tested isql -DSN -UID - PWD, that's successful, but when run by perl, message alert could not loaded driver database, anybody

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 13, Issue 7

Hello ALL SS7 for asterisk release http://www.footnotess7.com/. but i not yet account to download. any body have SS7. could you like send to me. thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 12, Issue 143

Hello ALl i need context to do: record to wave file and receive DTMF when recording wave file. for example: exten = s,1,Record(test:wav) exten = s,2,hangup when recording, press # to hangup and i want to receive others DTMF (while recording), max DTMF to received as 7 and when received enough 7

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 12, Issue 65

Hello All Any body have NVDialDetect module using for dialing out. (www.newmantelecom.com). How to solve problem, when dial out, i want to remote call answer, then asterisk play wave file. current, dial out, play wave file when remote call not yet answer. any advice ? Thanks

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 12, Issue 63

Hello ALL When i dial out to outside, how to detected remote call offhook. i append in extensions.conf [ext-callout];exten = s,1,NVLineDetect(60,d);exten = s,1,NVLineDetect;exten = s,1,NVBackgroundDetect(custom/aa_1);exten = s,1,MachineDetect(7000,2,2200);exten = s,1,NVFaxDetect(10) exten =

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 202

Hello Any body was tested LIBISUP. and price of LIBISUP packet ?. how much to purchased it from digium. if posible, tell me where are LIBISUP beta release to test with asterisk and my postoffice of my country (vietnam). Best regards.___ Asterisk-Users

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 183

Hello All I'm using TxFAX and rxFax. this's work when my system connected with PABX. then when i connect my card with PSTN, my system don't work. it's don't send and receive any thing fax document. Thanks Please help me___ Asterisk-Users mailing list

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 183

Hello All how to install functions allow called record current call by pressed any key to wave file. for examples. the caller call to asterisk, then press 123 tone to switch to saler person. then two persons conversation, the saler want to save current call. saler may be press any key as *5 or

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 181

Hello All How to detect remote called offhook. i make a context as below i created call file. copy to /var/spool/asterisk/outgoing. Channel: vpb/g0/9050718MaxRetries: 1WaitTime: 10Context: ext-calloutExtension: sPriority: 1 then when i copy to /var/spool/asterisk/outgoing. the asterisk auto call

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 131

Hi All I wan to get DTMF while voicemail recording sounds. DTMF received save to contents field of mail attach with wave sounds file. Please help me___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 136

Hello All i have 2 problems, please help me 1. How to implenment record call at called side. i want to record the call by called press the DTMF key. 2. how to implement call out functions, for example: i create .call file and copy to /spool/outging, then when asterisk call out, i want that: when

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 68

Hello All i have big problem for unicall. my system work successful with sangoma card, E1 and CAS signalling (vietnam). when at the some time. i have trouble then my system is half (CPU instructions = 100) i tested for some case as belows: - When i dial, then my system became answer, the caller

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 77

Hello All I'm settup my asterisk as belows: sangoma card, connected with E1, CAS Signalling. I have two problem. 1. The asterisk don't received any DTMF when caller input to 2. when i dial to system, the caller hear bad sounds. monitor on console. asterisk show error. Jun 11 12:15:45

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 69

Hi All i used cangoma card, connected with E1, using unicall. asterisk 1.1.x. when i dial to asterisk server. asterisk show error as belows: -- Unicall/9 extension '9' in context 'from-pstn' from '71811242' does not exist. RejectingcallJun 10 16:47:59 WARNING[28159]: chan_unicall.c:2655

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 48

Hello I'm using H323 channel and client used ohPhone-1.4.1 (with gatekeeper). when at client side dial to asterisk server (dial , test mode). ohPhone don't hear any thing sounds (no audio). i dial between ohphone (with gatekeeper). sounds are good. my current setting. Asterisk-1.1.x, GNUGK

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 17

Hello All How to measure the time when i click hold button on softphone. i need to save the time when user choice hold call for example: when user answer the call, the time is: 13:37:35 PM, after that, user choice hold call (time: 13:37:37 PM), then release hold call button (13:18:00 PM). the

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 10, Issue 234

Hello All I'm using asterisk 1.1.X and MFCR2 lib version 0.03pre2. when i call to E1 (connected with asterisk), chan_unicall don't detected event incoming call and show error. error messages:*CLI Warning, flexibel rate not heavily tested!Rx CAS bits 0x9 [ 1/ 0/ 0]Line unblocked -- R2 Channel

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 10, Issue 215

Hi All i'm using sangoma card. connected to E1, my wanpipe file as ## WANPIPE1 Configuration File### Date: Fri May 27 00:25:04 GMT+7 2005## Note: This file was generated automatically# by

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 10, Issue 114

Hello All I need to use Asterisk with an E1 sangoma card with CAS R2 signalling for Vietnam what is difference between libr2 of CVS and libmfc2 of soft-switch.org ? how to compile chan_unicall.c on asterisk. asterisk update CVS-head- May 27 2005.

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 10, Issue 221

Hello all. How to compile chan_unicall.c i have problem when compile chan_unicall.c, error message please help gcc -c -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 10, Issue 114

Hello All Any body used sangoma card A101. I have problem with this card. My system are: Linux Redhat 8.0. asterisk 1.07 and libpri, zaptel,... i connected E1 and MF/R2 signalling. i configure HDLC and TDM Voice. this is my configure as belows: ##

[Asterisk-Users] Sangoma card !

Hello All ! i'm purchased sangoma card A-101. i connect to E1 with MF/R2 signalling. but card don't work. negotiation with E1 fail. please help me to correct it. i dont' know some parameters such as: MTU, BAUDRATE Thanks Tin Trung ___

[Asterisk-Users] H323

Hello all i'm build success H322 in channel/H323 of asterisk. but don't know how to use it. i run GNUGK on server and client using ohphone. when i dial to asterisk server. the connection accept and disconnect. please help me to configure in H323.conf and extensions.conf.