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Hi, I've been trying to dial out to a cellphone, but all my calls get redirected to 066 (the emergency number at my city, like 911) does anyone know how to fix this, any ideas,? does anyone from mexico has done this? Any comment will be highly appreciated, Regards, Claudio -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if i can fill this 30 channels with REAL traffic for 2 or 3 days I can find new bugs on chan_unicall or I can see how stable it can be. Im using R2/MFC with chan_unicall the patch that Steve Underwood wrote. I will let anyone make FREE LOCAL calls to Mexico City till saturday or maybe until monday to see how stable this can

...) exten => 20,2,Goto(menu,s,1) exten => 20,3,Hangup() exten => 492774XXXX,1,Answer() exten => 492774XXXX,2,Goto(menu,s,1) exten => 492774XXXX,3,Hangup() [outgoing] exten => _0.,1,WaitforDigits(5000) exten => _0.,2,Dial(SIP/${EXTEN}@sip.1und1.de <mailto:SIP/$%7bEXTEN%7d@sip.1und1.de> ) /// Normal Way --> outgoing call to SIP-Gateway (TIMEOUT) exten => _0.,3,Dial(misdn/1/${EXTEN}) exten => _999.,1,WaitforDigits(5000) exten => _999.,2,Dial(misdn/1/${EXTEN:3}) /// 999-Way --> outgoing call at ISDN-Interface 1 (everything fine) exten...

...an to turn on/off reinvite capability or will every call on this channel be forced to use the SIP peer context for the duration of the call? Is there maybe a new feature in 1.6 that does this? exten => 5551212,1,Set(canreinvite=yes) exten => 5551212,2,Dial(SIP/${EXTEN}@othersippeer<SIP/$%7BEXTEN%7D at othersippeer> ,,) Something like that. Thanks. JR -- JR Richardson Engineering for the Masses -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091116/c41cd915/attachment.htm

Hello I am new in Asterisk Java, i am working on Asterisk 1.6.2.0 , i started the first example Hello AGI in this tutorial http://asterisk-java.org/development/tutorial.html I put the jar file and the proparty file in folder i wrote in extensions.conf this exten => 1300,1,AGI(agi:// 192.168.50.127/hello.agi,${EXTEN},${UNIQUEID},${CALLERID(name)}) I started AGI server ,then when i call

FWD (Free World Dialup) allows any SIP call to US toll free numbers via * 18xxzzzyyyy at fwd.pulver.com This works WITHOUT the need to be registered at FWD so in my dialplan I have something like: exten => _8.,1,Dial(SIP/fwd.pulver.com/*${EXTEN:1},60,r) exten => _8.,2,Hangup And I just dial 8-1-8xxyyyzzzz and presto ... calls go through just fine 99% of the time. I'm wondering if

Hi All, I am a newbie to Asterisk. Presently I am working with Asterisk 1.4.17 and using it to make SIP calls. I have a configuration of Asterisk which serves the users in a particular domain, say internal.com I would like to make a SIP call from bob at internal.com to charles at external.com I have added the following lines in extensions.conf exten =>

Hi all, I need to support this feature. When caller dial if the dial fail or no answer from the called number then play a music. So how to achieve that? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110509/c3bb9124/attachment.htm>

I have just installed * from the latest CVS and I can make calls via X-Lite to outside numbers but for only 30 to 35 Seconds at a time then Broadvoice will hang up on me but X-Lite will not know it. Once I hang up X-Lite then I will get the following error message: Spawn extension (default, Number-I-Dialed, 1) exited non-zero on 'SIP/200-6a22' -- Got SIP response 404 "Not Found"

I cant seem to get outgoing calls to be placed properly .. No matter what I try I get an error from the PSTN company saying that the "call can not be completed as dialed" or "you need to dial a one..." The asterisk debugging seems to show the correct number being dialed out of the zap interface... the "9" is being stripped and there is a "1" where it is

...164 [dundi-e164-lookup] include => dundi-e164-local include => dundi-e164-switch [macro-dundi-e164] exten => s,1,Goto(${ARG1},1) include => dundi-e164-lookup [iaxtel700] exten => _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel<http://IAXINFO%7D at iaxtel.com/$%7BEXTEN:1%7D at iaxtel> ) [iaxprovider] [trunkint] exten => _9011.,1,Macro(dundi-e164,${EXTEN:4}) exten => _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) [trunkld] exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1}) exten => _91NXXNXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}}) [trunklocal...

Hi everyone! I'm quite a newbie at this Asterisk stuff so please bear with me. We've recently decided to start training in Asterisk via AsteriskNow! Asterisk version is 1.4.18.1 through AsteriskNow! 1.02 The box we have is paired with a Digium TE110P and we've managed to get it to the point where incoming calls via a single DID (from NTT Japan) can be received and answered