search for: extesion

Hi, I get fascinated when I dial someone and get an IVR play " for accounts department press 1, for sales, press 2 or hold the line for an operator" and then have MOH play before the line is picked up at the desired extesion. Please, permit me as I know this will be one of the dumbest questions to ask in a group like this. I'll apprecaite any specific guide/instruction. Thanks in anticipation. Mike --------------------------------- Discover Yahoo! Get on-the-go sports scores, stock quotes, news & mor...

I have a very simple question . I have 2 internal extension 301 and 300 sip phone . I want to these extesion can call each other, and ext 300 can call outside to pstn, and ext 301 to call internatonal. How can I do that ? [x1] exten => 300,1,Dial(SIP/300) include => pstnlocal [x2] exten => 301,1,Dial(SIP/301) include =>international [pstnlocal] exten => _9xxxxxxx,1,Dial(Zap/g1/...

...Are you *really* sure that the 0 is transmitted in the CLI, and that it isn't stripped already by the phone company? I think the easiest thing for you would be to add the leading 0 before forwarding the call to your SIP client (ie. SetCallerID(0${CALLERIDNUM}) in your extensions.conf for each extesion where you'd like to add the 0). Regards Manuel ___________________________________________________ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com

hi, all I have configure tdm400p analog fxo card. that's ok. but how to chek that is working properly or not. i chek with ztcfg -vvvv and zttool . that's ok. i want to dial from my fxo port to another extesion. zaptel.conf ------------------ fxsls=1,2,3,4 defaultzone=in loadzone=in zapata.conf ---------------- context=mycontext signalling=fxl_ls group=1 channel=1-4 thanks' in advance. Bhrugu mehta

Has anyone else noticed that firefox is less stable after the recent security update? I'm getting several occurrences per day of "just goes poof," i.e. probably seg fault. -- Collins When I saw the Iraqi people voting three weeks ago, 8 million of them, it was the start of a new Arab world.... The Berlin Wall has fallen. - Lebanese Druze leader Walid

Hi! I need some help Well i have this cenario: 1 ip04 running asterisk [A] 1 pc running asterisk [B] I nedd to make calls from A to B, and B to A. Via sip The A-B calls are working. Now I need to configure the dial plan to call B-A either to sip numbers and Fxs. Anyone can help me? -------------- next part -------------- An HTML attachment was scrubbed... URL:

...BCE5-B09C0A99FD26@192.168.1.33 for seqno 52991 (Non-critical Response) --------- (192.168.1.44 is the Asterisk HOST) I can do outgoing calls with Zap interface without problems, only i __can?T__ do calls into my lan with SIP phone/protocol , i can listen voicemail because is the second action on extesion. These are my configuration files: sip.conf ------------- [203] type=friend qualify=yes username=203 secret=203 host=dynamic callerid=\"JuanI\" <203> canreinvite=no reinvite=no context = anurix transfer=yes mailbox=203 callgroup=1 pickupgroup=1 nat=never ---------- extensions.con...

...hone I can hit #, get the "Transfer" prompt and enter an extension I want to transfer to. No problem. I can do the same thing on the FXS port. My question is does anyone have a dialplan that will bring the call I transfered back to me if the transfer fails (i.e. busy extension or an extesion that does not answer)? I see many examples of going to voicemail but I have no idea how to get the call back to me. The general idea: - I call someone or I receive a call. - I want to transfer said call, so I hit # and enter the extension - if the extension answers, it's all good. - if th...

...=> s,n,AGI(myAGI.agi,2,${ARG1},${CALLERID(num)},001) <br><br> </p> <p style="margin-bottom: 0in;"><br> </p> <p style="margin-bottom: 0in;">The problem is that the macro is 'forked' (new UNIQUEID) and when it ends goes to _h_ extesion triggering unwanted actions..</p> <p style="margin-bottom: 0in;"><span></span></p><p id="__paragraph__1246635018000" style="margin-bottom: 0in;"><span></span></p><p id="__paragraph__1246635018000" style...

Hi Firstly could I thnk everyone who has helped me so far, I just have a couple of queries I have not had chance to debug this much yet but When using the tdm40p all extesions busy themselves out, and * cannot rint the extensions for incoming calls is this because I don't have a hangup statement at the end of the incoming context? if not has anyone any idea? does anyone have a quick and dirty IAX confiuration sample Thanks in advance Robb

Hi, asterisk gurus: My purpose is to key in some number after the call is finished. The number keyed in will be stored in the database with the phone number dialed. But whenever a key is pressed in/after h extesion, asterisk exits the call flow. Does anybody has a solution for this? Is DEADAGI is possible solution? Really need help! Thanks!Thanks!Thanks! Manny __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yaho...

hi, I'm with problems: 1- I'm a brazilian and dont speak english hehehehehe 2- heheh my asterisk dont speak in micprohone in virtual extension. while extension is local comunication is perfect, but while a extesion is virtual is possible just listen. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050524/c4a53ec6/attachment.htm

I've been using AMP to manage my * test system. I've been trying to activate an extension that I don't want AMP to manage. It would appear that the extesion definitions are placed in the appropriate "custom" files which are then added with an include command to the appropriate master file (sip.conf,extension.conf, etc). So far I've not been able to get it to work. Anyone had any experience with this? Thanks Bill

...meeting or anysoftphone ) how can i call any extension .... first of all tell me how to register a uid n password there .... let's say i've a user type=friend username=adeel secret=adeel context=incoming mailbox=31 plz plz plzzzzzz send me just a sample oh323.conf n related portion of extesions.conf containg above info .... --------------------------------- Start your day with Yahoo! - make it your home page -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050714/ceb9b6ae/attachment.htm

...meeting or anysoftphone ) how can i call any extension .... first of all tell me how to register a uid n password there .... let's say i've a user type=friend username=adeel secret=adeel context=incoming mailbox=31 plz plz plzzzzzz send me just a sample oh323.conf n related portion of extesions.conf containg above info .... --------------------------------- Start your day with Yahoo! - make it your home page -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050717/f9eaf8c4/attachment.htm

Hi i have installed AAH 2.6 and configured some extensions the calls are working fine. but if i dont answer a call then it says " the person at extension " and hangs up . it doesnt spell out the extesion number nor it goes to voice mail box. *************************** Asterisk CLI log **************************** dialparties.agi: Extension 200 is available...skipping checks -- dialparties.agi: DbSet CALLTRACE/200 to 208 -- AGI Script dialparties.agi completed, returning 0 -- Executi...

Hi, I have a queue with maxlen=1, and when i make a call, the call enters into the queue, but he doesn't exit from it after a reject signal received from the agent?? please, have you any idea how to remove calls after a reject signal??????? Thanks. Rachid

Hi List, I am making outgoing call from asterisk to GSM network with the help of VoIP trunk(SIP trunk) then I am not geting any caller ID at destination end. Is this the asterisk issue or VoIP trunk issue? Is this is due to asterisk then how we solve it? I already user Set(CALLERID(num)=XXXXXXXXXXX) in dialplan. ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer

Hi, I need to mount a UFS (Unix File System) partition from an external device (ie, live cd). I've seen the kernel-2.6.x-x.x.x.xxx.plus.c4 has UFS support and I've several questions about: ?How I know what FS are supported by my kernel? ?Is CentOS LiveCD edition builded with UFS support? ?Is there another way to get UFS support without the complete installation of a new kernel? Thanks.

...is I will grab any incoming SIP call form the IP address of my SIP trunk that matches XXX-XXX-XX00 and pass it to my XXX-XXX-XX00 context in extensions.conf [XXXXXXXX00] type=friend defaultip=69.67.248.51 host=69.67.248.51 fromuser=XXXXXXXX00 nat=no context=XXXXXXXX00 insecure=very And a look at extesions.conf: // // My thought is here I will route my incoming calls to a DID i haven't specifically routed to my default context (GoTo(XXX)) [sip-default-in] exten => s,1,Answer() exten => s,2,Playback(beep) exten => s,2,Ringing exten => s,3,Wait,1 exten => s,4,GoTo(XXX) // // My t...