similar to: stop the MOH since asterisk knows that channel is ringing

Dears -How can I stop MOH when status of the dial is ringing and let the user hear the Ring Back Tone from the termination Gateway. Even I can see in the CLI debugging the status is ringing -my idea is to add music on hold stop when asterisk detect -- SIP/OPNS-096456c0 is ringing line In which script this line located? -- Executing [97130245612 at default:1]

i installed skype for asterisk i can send and recieve calls normaly how can i receive messages from another skype user i Succeed to send only using for example: exten => 2233,1,SkypeChatSend(fSkypeBcp,User,message text) how to receive messages using this code SKYPE_CHAT_RECEIVE(<account>,<from>,<timeout>),and where and how I should add this code in extensions.conf

Dears What's the major deference between Asterisk 1.6.0.6 and Asterisk 1.4.23 Regards Khaled Chehab NGN Eng. Untitled Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: <mailto:bsara at mg-tel.com> Kchehab at Xplorium.com MSN ID :KhalidChehab at hotmail.com Web Site:

Dears, I have two questions 1-Is there a way to export fax tiff file image from .pcap captured file . In other words i am trying to backup all faxes that are passing on my network,and export the fax file later on. Is this feasible and how . i am using sip protocol and fax protocol is bypass ,G711 i can use T38 if that will solve the matter. 2-I tried using asterisk to receive the

when I enter asterisk-addons-1.4.1 and make menuselect ************************************* Asterisk-addons Module Selection ************************************* Press 'h' for help. XXX 1. app_addon_sql_mysql

Dears I am using centos 4.4 wirh asterisk 1.2.17 What I want is to make a customized packages to be installed from cd with no need every time to install packages and my personalized web interface , Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without

Dears Any one knows a standalone voip transcoder software name,not an ip pbx. What I want is to transcode the incoming sip calls from g711 to g723 or ilbc or g729 ..... and forward it to a media gateway .. Regards Khaled chehab ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by

Dears its too urgent Can anyone guide me ?? I want to put my asterisk system on an iso image like trixbox ,or how to make a. how can I do that ,I am using centos 4.4 final Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written

I am using centos 4.4 ,when I am compiling zapltel using l make linux26 ,error accrued ,what s missing [root@localhost zaptel]# make linux26 grep: /include/linux/autoconf.h: No such file or directory make: *** No rule to make target `linux26'. Stop. Regards ********************************************* No employee or agent is authorized to conclude any binding

I want to have a PC-based real-time VoIP bulk call generator (including both SIP signaling and RTP generation) for stress testing and precise analysis of the VoIP network equipment. Do any one knows a free program can do that . Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with

Dear All I installed a Digium card TE405P with zaptel and its running successfully with no alarms, but asterisk is not running . Any one have a cure or advice 03:09.0 Communication controller: Digium, Inc. Wildcard TE405P quad-span T1/E1/J1 card 5.0V (rev 02) Nov 14 07:56:58 localhost kernel: wct4xxp: Clearing yellow alarm on span 2 Nov 14 07:56:58 localhost kernel: wct4xxp: Clearing

Dears, I configured an account on my asterisk pbx to record the outgoing calls. When the asterisk pbx user make a call and send a fax the call recorded to wave file format. I searched the internet and found a software that can play the recorded wave file and export from it the tiff fax document sent. Is there a way that asterisk can play the wav file and export the tiff document ???

Hi, Dear How can I limit the number voicemail messages for a user by 10 messages only, I am using asterisk@home Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an

Dears I already read most of post on asterisk group and (http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning) But I could not find a calculator 1-Is there a calculator I can download for that 2-What I the maximum simultaneous calls that can asterisk handle using CPU 3.0 MHZ and 4GB ram With rtp g729 and there is no codec transcoding , 3-And what is the number of

Any one knows how to make Meet Me video conferencing room. Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily

Hi, I have a digium card (igium, Inc. Wildcard TE405P quad-span T1/E1/J1 card 5.0V (rev 02)) 4 ports I want to make a loop test between digium card E1 to test the configuration of dahdi What I want to do scenario is I connect port 1 and port4 in the digium card with E1 cable SIPcall-->E1 Digium port 1--->(Loop)E1 port 2---->sip extension local. kindly can any can help me to

Calling from a Cisco FXO port to an ATA-186 (SIP 3.1 image) via * (either CVS-HEAD-06/28/04-11:43:41 or CVS-HEAD-07/12/04-15:49:58) I didn't hear any ringing sound & get the following on the console: -- Called 5503 -- SIP/5503-f6b5 is ringing WARNING[-1323201616]: channel.c:1375 ast_indicate: Unable to handle indication 3 for 'SIP/10.10.2.250-9903' -- SIP/5503-f6b5 answered

I am using centos 4.4 server cd ,when I am trying to compile zaptel 1.4.2 ,the following error appears >From where I can get the missing rpms .or kernel source grep: /lib/modules/2.6.18-8.el5/build/include/linux/autoconf.h: No such file or directory grep: /lib/modules/2.6.18-8.el5/build/include/linux/autoconf.h: No such file or directory make[1]: Entering directory

I have a 'CONGESTION' Status with R2 protocol. While testing this scenario sip GW--?Asterisk ?Digium E1 R2 Protocol?Cisco E1 R2 protocol?sip Gw Find below my error and configuration ,where are the errors in my configuration ? ========================================================================= Connected to Asterisk SVN-branch-1.6.2-r235775 currently running on

Questions: Does anyone have a really STABLE asterisk system running about one year without need to restart the service or the SERVER ? Does anyone have a production Call Centre saled that don't lockup and is stable for 6 months ? I'm asking this questions because we have choose Asterisk for our call centre solution but, since the bugtracker only grows and people still want to stuck more