similar to: Free g729

I'm having problem powering Cisco phone's (7940 and 7905) on Linksys SRW224P switch (with PoE functionality). I have tested three phone's, one is working (7905) and two aren't (7905 and 7940). I have plugged all three phones on same switch port with same cable! Do I need to change anything in phone configuration? Is there something wrong with Linksys switch? How can I troubleshoot

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

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

- 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>

When agent tries to transfer a phone call (*2 - att transfer) he hangs up. Why? When a phone call isn't from queue then att transfer works fine. In features conf I have *1 for recording, *2 for att transfer and #1 for blind. In queue blind transfer works. For disconnect I have #0. I guess that * is somewhere defined as for hang-up the call, but where? I can't find it anywhere. Any help

Hello, I have been using FreeBSD-7.0STABLE with BIND-9.4.2 ( i guess, forget to check before upgrade) up to 2008-01-26 (yesterday). But after upgrade FreeBSD-7.0STABLE-->FreeBSD-7.1STABLE everything goes wrong. 1.BIND can't start anymore and giving me following message at /var/log/messages: . . . Jan 27 12:30:20 ns kernel: ad4: 152587MB <WDC WD1600AAJS-75PSA0 05.06H05> at

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

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

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"/> >

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

Where can I find alaw, ulaw, gsm, g729 formats for native music on hold? I have some mp3 files and I have tried to transcode them to above, but it seams that SOX can't do that. Please, tell me where to download some MOH files (in above formats) or how to transcode mp3? Thank you for your time! -- Tomislav Parcina tparcina#lama.hr

Does anybody use UTStarcom F1000G Wi-FI VoIP phone? http://www.utstar.com/Solutions/Handsets/WiFi/ I'm planning to buy one and I need to know did you have any problems with phone. What is the sound quality? How close you need to be to the access point? Please, any information's are useful to me. -- 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