similar to: New Release asterisk 1.6 Beta

Hi, I use asterisk 1.4.11 version for making outbound calls. Running it on linux(fedora core 7) machine. Recently purchased the g729 codec, got it registered with my asterisk box. I have two queries for you to help me. 1. How do i know when an outbound call is placed that it makes use of the g729 codec. when i use the command "show g729" i get the following: 0/0 encoders/decoders of

Hi, I have been working with asterisk to make ivr calls (outbound and inbound). I have the functionality - Read(variable|file_name) used in my dialplan. Now i need to pass the variable to my ruby file to compare the data entered with the database (mysql). How can i pass the arguments from my dialplan to the ruby file. Is there a way i can do it with the agi script? Any one has any clues on

Hi, I have the asterisk-1.4.11 set up installation on my Ubuntu machine. When i try making a simple incoming call using xlite softphone. I get the following message when i try calling to the number. *CLI> [Feb 19 13:35:40] NOTICE[4137]: chan_sip.c:5331 process_sdp: No compatible codecs, not accepting this offer! Which codec is it talking abt here. How can i resolve this. My dialplan is as

Hi, I am using Asterisk-1.4.11 version to make outbound calls and deliver the swift text to audio. My functionality is as for example i make this text to audio deliver the person called. Eg. swift -o /tmp/test.wav -p audio/channels=1,audio/sampling-rate=8000 "Press 1 to confirm. Press 3 to cancel." extension.conf dialplan: [dialout] exten =>

Hi, I have been using asterisk-1.4.17 version. Have a SIP registration from bandtel sip providers. Use DID numbers for the incoming calls which works fine when i dont use any peer setting in my sip.conf file. But when i use a peer and make calls thru the DID number it doesn't reach asterisk at all. Doesnt give me any errors as well. peer in my sip conf is as given below: [proxy2_bandtel]

Hi, I have been working with Asterisk for the ivr functionalities in the past. I am interested to implement the Jabber - Gtalk in asterisk. For which i installed the iksemel but this didnt help me out. I couldnt find the res_jabber.so file any where in the asterisk source directory. Still when i run the command "make menuselect" the channel driver "chan_gtalk" shows xxx

Hello, I am wanting to close a specific channel for example; SofthangUp(SIP/EXTEN-UNIQUEID) but the problem is the channel is assigned a unique id as well. The need fits into the idea of receiving a call from a higher status user and thus closing a specific channel to allow the higher priority call to route through the dial plan to the freed extension. Any ideas welcome. Many thanks

Hi, We've run into an interesting (to us) problem with SpeechBackground. Inside a AGI script, we're playing some extended audio?basically, like a podcast?and we want playback to stop if and only if the speech recognized matches something in our grammar. If there's speech that doesn't match, we just want to go right on playing. (We're using LumenVox as our speech recognition

I'm having a wierd problem. Somewhere around 1-2% of the time, the first DTMF digit dialed gets dropped. This is occurring during a SpeechBackground application call. If the caller reenters the digits when given a second chance, all is OK. Any suggestions how to debug this intermittent problem?

Hi All, I am using Speex for encoding/decoding the audio stream for my streaming application. I have used almost the same code In the sampleenc.c and sampledec.c, except that I have used The buffers from the mic as input for encoder and encoded audio As input for decoder, instead of files input. My Problem is the audio played on the receiving side after decoding Is only a "hush" or it

HI, I got this error when i am testing my rails application. I dont have the table named ''instructions''. But it shows a error like "ERROR: relation "instructions" does not exist". Totally, I got same error for 64 tests as 64 errors. I am using rails 3.0, Ruby 1.9.2, Netbeans 6.8. PS: I didnt creat Instruction manual for rails application. *Error:

Hi, I have a column of data with repeated as mentioned below. *column_name* Acer Lenova HP Lenova Acer Acer Lenova I need to populate this column in a combo box without repeated data as mentioned below *combo box* Acer Lenova HP I tried as <%= collection_select(:column_name, TableName.all)%> "table_names" is a table and "column_name" is a column needed to

The Asterisk.org development team has released Asterisk-1.6.0-beta3. This release contains a number of bug fixes over beta2, as well as a few new features. * Added an 'n' option to SpeechBackground to request that the channel not get answered * Added a number of new manager actions to improve configuration management over the Asterisk Manager Interface, including the ability to:

The Asterisk.org development team has released Asterisk-1.6.0-beta3. This release contains a number of bug fixes over beta2, as well as a few new features. * Added an 'n' option to SpeechBackground to request that the channel not get answered * Added a number of new manager actions to improve configuration management over the Asterisk Manager Interface, including the ability to:

As Raffael suggested, you would need to use an instance variable (@temp) rather than a local variable. The local variable would only be available from the context within which it is called, while the instance variable would be available instance-wide. This sort of situation is where a basic understanding of Ruby comes in handy. A question that I won''t go into, but one that you might want

The Asterisk Development Team would like to announce the release of asterisk-18.21.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.21.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-18.21.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/18.21.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-20.6.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.6.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-20.6.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/20.6.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!

The Asterisk Development Team would like to announce the release of asterisk-21.1.0. The release artifacts are available for immediate download at https://github.com/asterisk/asterisk/releases/tag/21.1.0 and https://downloads.asterisk.org/pub/telephony/asterisk This release resolves issues reported by the community and would have not been possible without your participation. Thank You!