search for: pezhman

...p this number to some ip-phones , base on received Caller-id. it is my database's view: 456 | DID | 14193016880 | 2 | hangup | | 455 | DID | 14193016880 | 1 | Dial | H323/1169#989181310524@66.152.61.66|60 | didx.org for test by pezhman it's work good. but for routing by caller id: 456 | DID | 14193016880/2085838 | 2 | hangup | | 455 | DID | 14193016880/2085838 | 1 | Dial | H323/1169#989181310524@66.152.61.66|60 | didx.org for test by pezhman this extens...

...------------------------------------------------------------------------------------------------------------------------------------------------- 460 | DID | 441216182112 | 2 | hangup | | 459 | DID | 441216182112 | 1 | Dial | SIP/8684831@sipgate|60 | pezhman in this mode, i dont have any problem, but I want to route by caller id, from a system that supports caller id, new value in dbase is: id | context | exten | priority | app | appdata |...

Dear for having an stable system which limit option is good for ulimit comand ? 2-is any option for making asterisk crash-free? Best -- Pezhman Lali -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110810/365d9d56/attachment.htm>

...------------------------------------------------------------------------------------------------------------------------------------------------- 460 | DID | 441216182112 | 2 | hangup | | 459 | DID | 441216182112 | 1 | Dial | SIP/8684831@sipgate|60 | pezhman in this mode, i dont have any problem, but I want to route by caller id, from a system that supports caller id, new value in dbase is: id | context | exten | priority | app | appdata |...

Dear, I added new field to cdr table , named "service" and type varchar(20), but in extensions.conf with the following command, nothing to be saved. exten => _X.,1,Set(CDR(service)=OUT) does asterisk support this ability ? is any setting must be changed, before that ? best Mani

Dear, Faxter is an opensource email to fax gateway, please check it, let me know if any bug. best -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110130/0f418a92/attachment.htm>

hello is any web based sip phone? for example: a user after logining in, view a configured sip phone, and ...... best MAni ____________________________________________________________________________________ Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel sites to find flight and hotel bargains. http://farechase.yahoo.com/promo-generic-14795097

...-+---------------------+--------- 9706015 | 9706015 | friend | 5056ed3c | | | no | | | md5 | | | 9706015 | GPHONE | | dynamic | | | | | yes | NULL | all | 0.0.0.0 | 0 | 0 | pezhman_lali at yahoo.com | 2009-01-31 11:33:10 | 9706015 (1 row)

Dear, do you have any successful experience for installing SHT-8C/PCI/FAX (synway) with asterisk ? is it compatibe with asterisk (dahdi/zaptel)? best -- Pezhman Lali -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110525/9df2050a/attachment.htm>

Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. ? thanks in advance Mani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/c1b2ed9d/attachment.htm

Hi dears, I am from saudi arabia and using asterisk 1.6.2.13,Dahdi-2.3.0 and Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) . I am facing problem with detecting caller id before first ring.I recorded the dahdi channel using dahdi_monitor command. Where I am able to see and hear caller-id dtmf tones. Pl tell me the procedure to upload recorded file if you needed. Something I want

Hi, I would have thought that when spawning a channel using the Originate() dialplan command, variables prefixed with two underscores would be preserved. However this does not work in the following case. Dialplan code: [intern] exten => 200,1,Set(__myvar="foo") exten => 200,n,Originate(Local/123 at test_orig,exten,dummy) [test_orig] exten => 123,1,NoOp(${myvar}) exten =>

Dears my Internet Provider , prevents , sip connections, between sip client(sip phone) and sip server, (asterisk + ser) . both of client and server are mine. is there any solution for tunneling the sip packets? best Mani ____________________________________________________________________________________ Don't pick lemons. See all the new 2007 cars at Yahoo! Autos.

hi After compiling and installing pwlib and openh323 , the asterisk, give the folloing error. please tell me where the problem is ? Best Mani *CLI> -- Executing Dial("SIP/2.2.2.2-086f5ac0", "H323/652#150388590962@1.1.1.1|60") in new stack Mar 28 14:17:23 WARNING[11985]: channel.c:2576 ast_request: No translator path exists for channel type H323 (native 4) to 256 Mar 28

hi xten.de produced an activex for web phone. but I can not find any link for download. can u help me ? best Mani ____________________________________________________________________________________ Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit. http://farechase.yahoo.com/promo-generic-14795097

dear can asterisk dial two numbers, then bridge them.(like jah jah) best Mani ____________________________________________________________________________________ Looking for earth-friendly autos? Browse Top Cars by "Green Rating" at Yahoo! Autos' Green Center. http://autos.yahoo.com/green_center/

Dear I am using Ser+Asterisk, for sip providing. there is a problem, the asterisk does not return back the busy tones to the sip phones. for example, if the destination number is busy, we are hearing waiting ring from sip phones, and after 60sec(timeout) the call will be terminated. thanks in advance for all help Best Mani __________________________________________________ Do You Yahoo!?

Hi now, I am using, realtime connection(mysql) for dialplan, but the following line must be added ,manualy to extensions.conf, before reloading.for each new context. [NEW_CONTEXT] switch => Realtime/@extensions is there any idea, to add this line to dbase too? thanks in advance Best MAni ____________________________________________________________________________________ Never miss an

Hi: I've configured an asterisk server with A102d sangoma's card and the E1 link.I want to dial outside number using the E1 Link.How can I dial a phone number?Is this true? exten => 123,1,Dial(ZAP/1/phone number) I'd appreciate any help. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear, the sip phones that registered, in to the asterisk 1.4.x have the echo in their callings to pstn. how this echo can be canceled? Best -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081120/d440509f/attachment.htm