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I am able to get this script to dial, but I am unable to talk or hear anything. The script asks for the number to call and the the caller id to display (if user is not at their normal extension). Once submitted, the external extension receives a call, once answered the call is then placed to the dentition number. The script works as the call is place, but I cannot hear or say anything. Any one

I have a gateway, which I call from my mobile phone (free of charge, since it is the same phone company). This gateway gives me a dial tone. I can than dial to any extension number or even other gateways, .... It is getting more a trouble to remember all the numbers, or to key in all the long phone numbers when you got the dialtone. I was thinking of using for this Sphinx2. How can I

Hi, I have these extensions: exten => 101,1,Dial(SIP/101,15) exten => 102,1,Dial(SIP/102,15) exten => 0,1,Dial(SIP/101&SIP/102,15,r) They work fine and I get the ringing sound if I dial them directly. However, I also have this extension: exten => s,1,Answer() exten => s,2,Background(viagenie) exten => s,3,WaitExten() The ringing sound doesn't work for any extension

...driving me crazy. Is there anyone who can help? - -- Kind Regards, Matt Riddell Director _______________________________________________ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHAGLUDQNt8rg0Kp4RAgh6AJ98NQauv8Ze60lNhjRYgL4PNe66AwCdGZIv gMwh8FIagmPFT20ohzV+M7I= =diDo -----END PGP SIGNATURE-----

The x86 and x86_64 Digium G.729 codec binaries have been updated for use with the Asterisk 1.4 beta (which should also work on current svn trunk). Anybody that is using the older modules with the 1.4 beta (or svn trunk newer than several days ago) is strongly encouraged to upgrade immediately, to avoid potential issues. -- Jason Parker Digium

The x86 and x86_64 Digium G.729 codec binaries have been updated for use with the Asterisk 1.4 beta (which should also work on current svn trunk). Anybody that is using the older modules with the 1.4 beta (or svn trunk newer than several days ago) is strongly encouraged to upgrade immediately, to avoid potential issues. -- Jason Parker Digium

Hi all, Just ran into some issue with the originate AMI command. It seems that there is a limit of around 120 calls I can place with the originate command simutanously. By that I mean sending Asterisk a lot of originate command very fast. Anyone know if there is a limitation? Thnx. -------------- next part -------------- An HTML attachment was scrubbed... URL:

...put them all together and get a framework going. - -- Kind Regards, Matt Riddell Director _______________________________________________ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG1yKhDQNt8rg0Kp4RAv5UAJ48tW28T5lWCQIPTwVimyvlhEPJowCgpnE6 OF3L2M/6Hc+YBNL1NFx6dzA= =OXNn -----END PGP SIGNATURE-----

...> > > Matt Riddell > > Director > > _______________________________________________ > > > > http://www.sineapps.com/news.php (Daily Asterisk News - html) > > http://wap.sineapps.com (Daily Asterisk News for your cellphone) > > http://feeds.feedburner.com/AsteriskNews (Daily Asterisk News - rss) > > -----BEGIN PGP SIGNATURE----- > > Version: GnuPG v1.4.2 (MingW32) > > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org > > > > iD8DBQFF+GeUS6d5vy0jeVcRAhfCAJ4oG+PItrOEoZEDhuzNf0dzOykllACfbI67 > > NV4lAmOkaISR79fBTjajG...

Earlier this week 'refresh' releases of these two projects were put on our FTP servers, but due to some miscommunication on our end no announcements were sent out... so here they are :-) Asterisk 1.2.12.1 fixes one significant bug that was introduced after 1.2.10 but which hadn't been corrected yet when 1.2.12 was released (this bug is related to setting up transcoding paths between

Sorry for asking this question again but before I blow 100 dollars on a X100P I need to know this info: So does "SET EXTENSION <new extension>" allow for you to set which extension the rest of the call will occur over? So if a call comes into the switch and I could make the AGI script check the DID or DNIS which is really in the variable agi_dnid? After that I can do a database

Earlier this week 'refresh' releases of these two projects were put on our FTP servers, but due to some miscommunication on our end no announcements were sent out... so here they are :-) Asterisk 1.2.12.1 fixes one significant bug that was introduced after 1.2.10 but which hadn't been corrected yet when 1.2.12 was released (this bug is related to setting up transcoding paths between

On an Asterisk 1.2.12.1 system using a TDM400P with two FXO and two FXS modules, servicing two POTS lines: When dialing a number, such as a bank, or pharmacy, where it is required to enter a long series of numbers via the phone's keypad, an unexpected hangup occurs. The hangup does *not* occur when entering the numbers, as one might initially expect, but after the called end begins to

Is there anyone with any experience of using the AMD app and the settings that worked for them in the UK ? Any help would be appreciated. Julian

We have decided to allow our tech's to do support for non-clients of voicemeup.com You can head to http://support.voicemeup.com/ and one will be in touch 8 to 6pm business hours. 3 levels of support are offered for Asterisk/compiling Trixbox , Ivr's etc. -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 -------------- next part

Hi Folks. I have installed two sip phones and two PCs in a network. The later with iaxComm. Calls are made between the sip phones and between a sip phone and a PC. When calling from one PC to the other the iaxComm shows ??? in the status column and the call can't be answered. The same goes if anyone calls from the outside and tries to reach a PC. Any ideas? Regards. -- IEE Jos? Hugo

Hi folks! I'm having a problem where my music on hold is just blaring to my callers. I've tried several different formats (converting using mpg123 and sox) and adjusted my musiconhold.conf to use quietmp3, to no avail. Every file plays way too loud. I did notice that sox has a -v flag for adjusting volume, but danged if I can find documentation online that'll tell me what

Hello All We have developed an application for Home Delivery Restaurants using Asterisk, Java (JSP/ JSF) and MySQL. Here is listing of its features. If someone is interested then we can provide him more details. - POP up window with caller data containing his/her name, address and transactions history. - In case of new customer, Pop up window with blank form to add customer data and

Any recommendations for an affordable SIP conference bridge unit? I mean one that isn't crappy; something where the duplex and cancellation functions that are traditionally built into such devices actually work. I am referring to something that looks like this . . . http://www.hardware.com/products/cnet/I212272.jpg But not necessarily that. -- Alex Balashov Evariste Systems Web :

I've been trying to post a specific message for the last four or five days. It's on a specific topic, and I suspect the topic is the reason it is not being published to the list. Which would suggest that some kind of keyword filtering is being done, though I've rephrased the message several different ways without success. I'm sending this message to see if my new posts even make