similar to: Asterisk Differences

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

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 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

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 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

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>

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

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

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 Please how can create an independent group of users on asterisk ,in which user on group A cant dial user on group B. Thanks 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

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

Hi I have cisco 7960 connected to asterisk ,using tftp xml config file,my problem is it can receive any call but it cant call any extension. Please can you send me ,how to solve this issue 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

HI, My system been attacked from someone I guess, kindly check the link below How can I stop the ircd attack http://pastebin.com/tbjh5qzP 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

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

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 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

Dears Any one succeeded to make Redundancy / Failover with asterisk 1.4.9 on centos with kernel 2.6.9-55.EL. Can you please send me the documentation link on how to or write down how to . Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without

Hi All I am using asterisk 1.2.4 Please see the results when I execute Sip show channels X X X X x 192.168.8.106(None) 04cddc1f5a0 00101/00000 unkn No 215.96.142.83 (None) caac0846-cf 00101/00000 unkn No 192.168.8.106(None) 94910146-46 00101/00000 unkn No 192.168.8.106(None) 793ed1eb0f2 00101/00000 unkn No 85.219.172.253 (None)

I installed asterisk 1.4.4 final ,but the cdr is not written any patch or tweaking can be done 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

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]