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Hello, I'm trying to configure Asterisk with Radius cdr support. Asterisk version 1.6.2.13 Server Radius: Freeradius version 1.X Radius client: radiusclient-ng version 0.5.5 With the Asterisk core debug on 1 when a call terminate, on the console appear this error: Unable to create RADIUS record. CDR not recorded! My cdr.conf is: [radius] usegmtime=yes ; log date/time in GMT

Hi , Is it possible to have two meetme room in asterisk 1.6 which each one have a different language? I mean, one room the annoucement is in Portuguese an another in english? Today I can change over the sip.conf and it is valid for all room. regards! Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda -------------- next part -------------- An

Hi it's possible into Asterisk 1.6.x to limit a call at 120 mn ? after 120mn, hangup and the customer call a new time thanks olivier

Hi Everyone, We use the top buttons on Aastra 55i to login and logout from Queues. This is the order: Button 1 = Login to English Queue Button 2 = Login to Spanish Queue Button 3 = Logout of English/Spanish Queues There are indicator LEDs on each of these buttons. Is there anyway we can send a SIP request or some other communication to get the Aastra 6755i phone to keep the LED for login set

Hello all, We would like to "inform" the caller of the reason for a failed call. For example, when we get a "486 Busy Here", the system accepts it and in the CLI we see "Everyone is busy/congested at this time". Can we use this data to play an announcement to the caller? Thank you in advance for your help. Michael -------------- next part -------------- An HTML

Hi, I'm trying distributed events with Openais but don't work. I made the test with two asterisk box in the same LAN box A: 192.168.142.246 asterisk 1.6.2.13 BoxB: 192.168.142.248 asterisk 1.8.0 openais.conf: # Please read the openais.conf.5 manual page totem { version: 2 secauth: off threads: 0 consensus: 4800 interface { ringnumber: 0 bindnetaddr: 192.168.142.0 mcastaddr:

Hello, I'm trying to use SIP_HEADER function on my dialplan but I receive this message (on the console): pbx.c:3367 ast_func_read: Function SIP_Header not registered Why? Thank's - Bakko

hello, I'm trying to connect Asterisk to Google Calendar. The connection work fine but Asterisk don't retrieve any programmed event present on the calendar. Asterisk version 1.8.20.1 Any hint? Thank you - Bakko

hello, I' trying to use Asterisk 12 Alpha. Compilation and instalation without issues. When I try to start asterisk with: asterisk -cvvvvvvvvvvvvvvv i see this error on the console: 17:09:43.559 sip_endpoint.c !Module "mod-refer" registered asterisk: ../src/pjsip-simple/evsub.c:415: pjsip_evsub_register_pkg: Assertion `mod_evsub.mod.id != -1' failed. Any hints? Thank you

Hi, I'm trying to send Voice mails from my existing Windows application to an Asterisk system. I'm new to Asterisk and to VOIP. Could you please guide me with this? Regards, Shyamala -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101108/3238c1d5/attachment.htm

Hi, I created a extension in Asterisk, the extension has been configured in Android softphone 3cx. When I tried to call from Andorid phone to some other IP extension which is registered in Asterisk, I am not able to hear the voice, when I check the asterisk log or wireshark there is only one way RTP traffic, from Android I am connecting to Asterisk via 2G GSM network. Any idea would be

Hello, I have written an AGI script for asterisk that uses google translate for text to speech synthesis. It supports a variety of different languages, local caching for the voice data and wideband audio. The voice in most languages is female and the quality of the synthesized speech is very high. More info about the script can be found here: http://zaf.github.com/asterisk-googletts/ the first

Hello, For those customers with only analog lines, who ask for fax2email and email2fax, whats the most reliable solution available and tested with Asterisk? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120529/3e28b56e/attachment.htm>

Hello, I'm trying to conect two 1.6.2.13 Asterisk server with IAX. This is my configuration: Asterisk A: iax.conf register => coiax:pass1 at 69.164.207.166 [smiax] type=friend host=dynamic trunk=yes secret=pass2 context=phones deny=0.0.0.0/0.0.0.0 permit=69.164.207.166/255.255.255.255 qualify=yes Console: iax2 registry 69.164.207.166:4569 N coiax 69.164.197.105:4569

Hi I have configured IAX2 realtime in Asterisk 1.6.2.13. when I cannect a client to realtime extension, always the state of extension is "UNKNOW" like: * Name : marco Secret : <Set> Context : phones Parking lot : Mailbox : 2345 at default Dynamic : Yes Callnum limit: 0 Calltoken req: Auto Trunk : No Encryption : No

Hello, maybe is a stupid question but I'd like know if I'm doing some mistake on IAX trunk configuration. I have two Asterisk (1.6.2.14) with this configuration: A (iax.conf) register => serverb:password at 192.168.142.246 [servera] type=friend host=dynamic trunk=yes secret=password context=phones deny=0.0.0.0/0.0.0.0 permit=192.168.142.246/255.255.255.255 qualify=yes B

Hello, I'm trying to send a fax with sendafax aplication and receive the fax with the receiveFax aplication on the same Asterisk Server (1.8..8.2). All work fine but the PBX always use T30 protocol. Is thes a variable or setting to configure Asterisk to send and receive this fax with T38 protocol only? Thank you Regards

Hello, during DAHDI 2.7.0.1 compilation on CentOS 6.5 64bit, I have this error: In file included from /usr/src/dahdi-linux-2.7.0.1/drivers/dahdi/dahdi-base.c:66: /usr/src/dahdi-linux-2.7.0.1/include/dahdi/kernel.h:1407: error: redefinici?n de 'PDE_DATA' include/linux/proc_fs.h:328: nota: la definici?n previa de 'PDE_DATA' estaba aqu? make[2]: ***

Hello Is there a way to slow down or speed up the speed at which SayDigits rattles off a series of digits? Reagards