similar to: registration ip address

Hey all, after hooking up some Plantronics to some Snom's (3 320's 1 360), I noticed my client is having some form of feed back on the phone. Because of Snom's "inner oddities" this is how I got it to work. Plantronic --> RJ11 --> SnomHandset Port (on Snom Base) Handset --> Plantronic jack (bottom base in the front) If I placed Plantronic(RJ11) --> Snom's

Looking at my log file I found the following error: May 2 12:00:45 debian kernel: Zapata Telephony Interface Registered on major 196 May 2 12:00:45 debian kernel: No ISA tormenta card found at d0000 May 2 12:00:45 debian kernel: Zapata Telephony Interface Unloaded May 2 12:00:45 debian insmod: /lib/modules/2.4.20-8smp/misc/torisa.o: init_module: Input/output error May 2 12:00:45 debian

Hi. We have people with two or more sip phones. One wireless and one wired. So this is the case: Person A with two phones wants to have a queue for his incoming calls. So when he answers one of the two phones, the other phone should not ring. But when he isn't talking in any of the phones, they both should ring. Does that make any sense? This what I have for people with only one

Hi guys. This is my 1st post here (after much reading). I have a test asterisk system setup using X-Lite Soft Phones, and the issue I am running into is that caller id shows up as "asterisk" on all incoming calls and on all local to local calls (internal). I have showcallerid, etc. configured in zapata.conf, but I'm drawing a blank. When I check my voicemails it tells me

I'm registering 5 lines on my asterisk box from one voip provider. Lines; 4040.0000 4040.0001 4040.0002 4040.0003 4040.0004 All lines is registered in 5060 port so when someone call to 4040.0001 the call arrive on asterisk but arrive to last number registered 4040.0004becouse it is listening on same port as all others. How i make each number register in one different port, like

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, 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 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

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, 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

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]