similar to: Asterisk users Calculation

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, 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 ???

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

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

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 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

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

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

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

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 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

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

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

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

How to configure cisco 7905 with asterisk ,if you please can send me step by step configuration steps . Thanks Regards Khaled Chehab System Integration Engineer Xplorium Offshore. Sakiet Al Janzir Postal Code: 1102-2080 Tel: (961) 1- 868 686 Fax :(961) 1-808 810 GSM: (961) 3-979 343 ********************************************* No employee or agent is authorized to conclude

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

What I am asking for is something to take an incoming SIP INVITE, change the codecs listed in the SDP, forward the (new) INVITE to a media gateway, perform the reverse codec handling for the 200 OK and perform RTP transcoding on the resulting 2 legs of the call. -How can asterisk do that ! -do any one know a distribution contain asterisk have solution like that ? Regards -----Original

Hi, Can anyone suggest which is the best scripting language for Asterisk or any telecom device? Thanks in advance. -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:saigop at gtalk2voip.com -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110401/051f68d3/attachment.htm>

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