similar to: meet me

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

I created Master.csv in /var/log/asterisk/cdr-csv ,even did not work,do freepbx do this problem,and how can I trouble shoot it. Thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ken Williams Sent: Wednesday, May 16, 2007 11:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

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

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:

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 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 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

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

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

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, 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

Dear I am using a2billing accounting software, how can I charge on the destination target not at the caller side Ex: if user A have 10$ and user B have 10$ ,and the onnet call charge cost 1$ When user A call user B for 1 minute ,user A amount remains 10$ and user B amount be 9$ Regards ********************************************* No employee or agent is authorized to conclude

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 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 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

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

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)

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 am using a2billing as billing software ,and I make an 800 call service which means that the destination extension should be build I put this code at extensions.conf exten => 99909994,1,SetAccount(2704714849) exten => 99909994,2,Wait,2 exten => 99909994,3,DeadAGI(a2billingp.php) exten => 99909994,4,Wait,2 exten => 99909994,5,Hangup its not stable ,its works for 3 times

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]