similar to: RE: Re: Re: Compatible Asterisk Connectivity Cards :Sangoma

One simple question. I'm using asterisk 1.2.1, can one agent be defined in more than one group? Example: group=1 ; queue1 agent => 401,401,Tomislav Parcina agent => 402,402,Katarina Ivanisevic agent => 403,403,Sasa Juginovic group=2 ; queue2 agent => 401,401,Tomislav Parcina agent => 402,402,Katarina Ivanisevic agent => 404,404,Marija Bilic agent => 405,405,Ana

How can I send recordings, that I have recorded with One Touch Record, to e-mail address that is defined in voicemail.conf? Thank you for your ideas. -- Tomislav Parcina tparcina#lama.hr

- How to restart the phone? (On 7960 it is *+6+Settings) - How to setup working dtmf? - How to setup hinting? For line is <line button="4"> <featureID>9</featureID> ... For speeddial is <line button="5"> <featureID>2</featureID> <featureLabel>341</featureLabel> <speedDialNumber>341</speedDialNumber> </line>

Hi list! How to send a call directly to voicemail recording? When I put this exten => 313,n,VoiceMail,u221 Or this exten => 313,n,VoiceMail,b221 In my dial plan, calling person first hears greeting message (busy or unviable). I would like to avoid greeting message (I would play something with Playback application). Is it possible? -- Tomislav Parcina tparcina#lama.hr

Hi, I just edited this page: http://wiki.rubyonrails.org/rails/pages/DarwinPorts I thought I''d post something to this mail-list in hopes of an answer.. I''m running Tiger. On my Mac, I cannot get past installation of readline: mac:~ mac$ sudo port install readline Password: ---> Fetching readline ---> Attempting to fetch readline51-001 from

I have several Cisco 79xx phones (7905, 7920, 7940, 7960, 7970, ATA 186) and I need to buy firmware for them. I have contacted http://www.cdw.com and http://www.insight.com/ but they didn't respond. Can anybody tell me where can I buy SCCP and SIP firmware for my phones? BTW, I'm in Croatia (Hrvatska). I heard that location does matter. P.S. My local Cisco reseller wants to sell me

I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/ I have some wav files (with 755 permission). In musiconhold.conf I have [native] mode=files directory=/var/lib/asterisk/moh-native And in sip.conf I have musicclass=native When I put call on hold this is what I get at CLI. -- Executing Dial("SIP/341-5931", "SIP/344|20|wWtT") in new stack

Hi group. I have huge problem. My pickup exten #8 isn't working. This is what I have configured. pbx*CLI> show features Builtin Feature Default Current --------------- ------- ------- Pickup *8 #8 In sip.conf I have callgroup=2 pickupgroup=2 For called party and same for person that is trying to pick up the call. The person that is trying

In article <Pine.LNX.4.64.0603211635320.7043@ab1-1-246.shsu.edu>, amdtech@shsu.edu says... > You have to set up a dialplan.xml file in your tftpboot directory for the > phone to pull: > > <DIALTEMPLATE> > <TEMPLATE MATCH="9,59....." Timeout="0"/> > <TEMPLATE MATCH="9,29....." Timeout="0"/> >

In article <02a201c64f16$7376fb10$0201000a@JACK>, balgaa@micom.mn says... > Hello, > > I installed Asterisk from CVS on Redhat Linux 9 and working with chan_h323 module and g729/g723 free codecs too. Can you send us more information about this free g729 codecs? -- Tomislav Parcina tparcina#lama.hr

Hi group! I have been using asterisk 1.2 for quite some time and now I need to go back on asterisk 1.0 (because of oh323 channel driver). One thing that I can't remember is does asterisk 1.0 support native music on hold? If it does, how can achieve it, because it doesn't work the same way as it does on asterisk 1.2. Thank you for your help. -- Tomislav Parcina tparcina#lama.hr

When I try to make att transfer (*2) of call that was in queue the call get's disconnected. Blind transfer (#1) works fine. In dial plan I don't have any h or H (hangup call with *). In features.conf I have this line disconnect => *0. What could be the reason why call hang's up? -- Tomislav Parcina tparcina#lama.hr

Situation. I call out from SIP phone over h323 trunk and called person decides not to pick up (on mobile phone they press red button - NO - hang-up). Until the called person press the NO button, I can hear ringing. When called person press the button, I don't hear anything. Asterisk waits until timeout and than ends the call. How can I get busy or some other appropriate signal on SIP phone

Does ooh323 from asterisk-addons 1.2.1 support alaw codec? This is what is written in h323.conf.sample that can be found in asterisk-addons dir. The codecs to be used for all clients.Only ulaw and gsm supported as of now. Default - ulaw ONLY ulaw, gsm, g729 and g7231 supported as of now disallow=all allow=gsm allow=ulaw So, it shouldn't support alaw, but I manage to establish calls with

How to disable silence suppression (or Voice activity detection - VAD) on Cisco 7905 phone? On Cisco 7940 I use "enable_vad: 0", but I can't find anything similar for 7905. -- Tomislav Parcina tparcina#lama.hr

Hi group! How to set language for queue? I have several queue's. In every queue, agents speaks different language. I need to announce queue-youarenext and similar on different languages. This is what I have in my extensions.conf and it does set language, but when calls enters queue, it doesn't use that language. exten => 313,1,Answer exten => 313,n,Set(LANGUAGE()=de) exten =>

How to upgrade h323 from Asterisk add-ons (from version 1.2.1 to 1.2.2)? In INSTALL they don't say anything about upgrade... Thank you for your time! -- Tomislav Parcina tparcina#lama.hr

Hi group! Does anybody knows about any news server that works the same way that Gmane www.gmane.com/ does it? I was satisfied with Gmane for few months, but now it seams that it doesn't work any more (no new posts in past few days). Now I'm looking for alternative. -- Tomislav Parcina tparcina#lama.hr

HI Group! I have strange problem. Since I started to use H323 with my VoIP provider when I dial the person that is currently busy, I can't hear busy tone on my handset. What could be the problem? What should I look for? How is this exactly called (because I even don't know what to look for). Hopefully someone will be able at least to give me some starting point. -- Tomislav Parcina

I use asterisk 1.2.5 and h323 that comes with addons 1.2.1. Incoming call comes on h323 trunk. Person A (local SIP phone) receives call and tries to make attendant transfer to person B (local SIP phone). They speak. Then A hangs up. Call form h323 trunk doesn't get to person B. This is what I get on CLI. -- Attempting native bridge of OOH323/xxx.xxx.xxx.xxx-5381 and SIP/307-5663 --