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...einvite=no nat=1 disallow=all allow=ulaw allow=alaw Extensions.conf exten => 203,1,Dial(SIP/lcs-sipura1) exten => 204,1,Dial(SIP/lcs-sipura1) ----------------------------------------------------------------- \ \____\_ Ariel Batista / / IS Director / Avionica, Inc. ------------------------------------------------------------------ abatista@avionica.com Ph: 786-544-1114 Fx: 305-574-0212

...ines! Here is a sample on how we use this settings in extensions.conf exten => _91NXXNXXXXXX,1,SetCallerID(305XXXXXXX) exten => _91NXXNXXXXXX,2,Dial (${LDTRUNK} / $ {EXTEN:1}) exten => _91NXXNXXXXXX,3,Congestion Thank you in advance for any help with the 2 above problems. Ariel Batista Avionica, Inc. 14380 SW 139 Ct. Miami, FL 33186 Ph: 305-256-0429 x114 Fx: 305-574-0212 web: http://www.avionica.com email: abatista@avionica.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030905/4fb67280/atta...

...#39;s. So we need to be able to give the correct address to the 911 call! This is just for our locations and not for reselling our Asterisk server! ----------------------------------------------------------------- \ \____\_ Ariel Batista / / IS Director / Avionica, Inc. ------------------------------------------------------------------ abatista@avionica.com Ph: 786-544-1114 Fx: 305-574-0212

...sterisk? They seem simple enough to config with there web interface. Thanks ----------------------------------------------------------------- \ \____\_ Ariel Batista / / IS Director / Avionica, Inc. ------------------------------------------------------------------ abatista@avionica.com Ph: 786-544-1114 Fx: 305-574-0212 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031205/a40...

...k. Is there any other way to update them without restarting the system? Since the system is used allot it's hard to find a time to restart it. ----------------------------------------------------------------- \ \____\_ Ariel Batista / / IS Director / Avionica, Inc. ------------------------------------------------------------------ abatista@avionica.com Ph: 786-544-1114 Fx: 305-574-0212

...her Sip phones working fine. Cisco 7960'g, IpDialogs They all work fine. ATA 186 and Sipura-2000 are also working fine they all can park a call and pick them up. ----------------------------------------------------------------- \ \____\_ Ariel Batista / / / Avionica, Inc. ------------------------------------------------------------------ abatista@avionica.com Ph: 786-544-1114 Fx: 305-574-0212

...y to have there own context. But I am using standard_macro for all the extensions. My extension.conf file is extermly large so it is not a good idea to post it here. ----------------------------------------------------------------- \ \____\_ Ariel Batista / / / Avionica, Inc. ------------------------------------------------------------------ abatista@avionica.com Ph: 786-544-1114 Fx: 305-574-0212

...ard to take IAX connections? Is there any way to get this working via two paths? There is only one bindipaddr=10.1.1.1 for internal trunk but outside address section? ----------------------------------------------------------------- \ \____\_ Ariel Batista / / / Avionica, Inc. ------------------------------------------------------------------ abatista@avionica.com Ph: 786-544-1114 Fx: 305-574-0212

I have a Cisco 7960 & IpDialogs that I am not able to use the transfer button on it. What happens is that it puts the call on hold and then it gives you a dial tone. You can dial but it will not transfer the call. What we are trying to do is transfer to extension 700 for parking so another person can pick up the line. We can not use the # key to do this due to we have several IVR's

Yes folks it's me a Newbie. Remember I am also a non-Linux person trying to learn. I have a production Server running Asterisk .5 12/02/03 CVS, and would like to upgrade it to the new .71. Has anyone come up with instructions (Documentation for us newbie) on how to do this? My server is running Mandrake 9.0 which I know nothing about! Sorry if this sounds stupid but all the instructions I

Hi I am a user of Asterisk-0.5.0. I am a final year student of MCA in IGNOU.All the system are running in Red Hat-7.3 OS. I am able to transfered call in the following procedures: PSTN(INDIA)>>Mediatrix 1204>>Asterisk server>> VOCAL server>>Mediatrix 1204>>PSTN(USA) Now I want to save the cdr data in my Asterisk box.I am using RedHat-7.3 OS. I am using the

Ok here is a question that has gotten me stumped. I have an Asterisk system up and running. I need to connect it via the Internet to a Sip Cisco system. This is what they have. I have there IP address's and login. X-lite is able to connect to them and make a call! So I have the name right! CISCO router model: 2621 VoIP module: NM-HDA-4FXS I have done Google lookup and at the Wiki about

I need to get 2 Asterisk servers working together. I have been reading and doing just about every example I have been able to find here on the list and the Wiki. It's now gotten to the point that nothing on box2 seems to be working. I seem to have a major problem understanding the format. Here is what I have so far. It's 3 days of hair pulling and nothing seems to work! Asterisk box 1

I would like to prevent * from dialing 900 and 976 numbers. I setup the following settings in extensions.conf. But this does not seem to work! I don't know what I am doing wrong please help! exten => 1900XXXXXXX,1,Congestion exten => XXX976XXXX,1,Congestion exten => XXX976XXXX,1,Congestion exten => 1XXX976XXXX,1,Congestion exten => 91900XXXXXXX,1,Congestion exten =>

I am trying to get the following phones for testing. Is there a distributor in the US that is able to sell me these Sip phone and ATA adapters? I can not afford the Cisco phones there too hard to configure and too expensive! 1 - Sipura SPA-2000 2 - Grandstream Sip phone BT-102 1 - Grandstream HT-286 1 - Snom 105 Sip phone. I have called and emailed chagres but they have not reply. Nor

Are you looking for local or Long distance? What will these T1s primarily be used for(inbound/outbound, domestic, local, long distance, international) How important are per minute rates to you? how many minutes do you expect to use per month? We are in Tampa Florida and have 15 T1s from several different providers so I may be able to refer you to one if it's a match to what you're

Hi all, DIAX 0.9.3 is available for download from the same place: http://www.laser.com/dante or http://www.geocities.com/tdanro The new DLL contain the latest updates made by Steve in the iaxclient library. Still just IAX1 is supported (for the moment). What's new in 0.9.3? - accept blank passwords; - accept for registration/calls host names, not only IP Address; - password no

I have a problem that sometimes lines will go into what I call never never land. The Asterisk system will put a line with <Zombi> on it when you type show channels it will make the analog phone line dead. And on the CLI it says: astsvr*CLI>Zap/1-2<ZOMBIE>(macro-twoline-exten,s,1)Up Dial Zap/1-2|20|r I have tried to release it with soft hangup Zap/1 & also soft hangup

Hi, I have this scenario Cisco 5300 (public ip. 200.47.xx.xx) <---> Asterisk (public ip: 64.76.xx.xx) <--> Cisco 3600 (public ip: 64.76.xx.xx , same network than * ) When a calls comes in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome message and resend call to Cisco 3600 that have 4 analog lines connected... but after cisco play welcome message and when

I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway working with asterisk. The Idea is the following. PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} -- Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic system has FX0