similar to: Re: Re: Cisco 7960 - Have to press a menu button to dial

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"/> >

Does anybody have Digium TE212P interface card on Dell PowerEdge 1850? I'm planning to install * on that configuration so I'm looking for any positive/negative experience. Best regards, -- Tomislav Par?ina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: tomo@sip.lama.hr e-mail: tparcina#lama.hr http://www.lama.hr

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

I too had a server room fry and need to replace h/w. So what specific Dell servers did/do you deploy? Where is the link w/Digium/s Dell caveats? I'm using the Digium TDM400 card w/* > Date: Thu, 13 Apr 2006 19:19:13 -0500 (CDT) > From: Aaron Daniel <amdtech@shsu.edu> > Subject: Re: [Asterisk-Users] Digium cards, so disappointing ! > To: Asterisk Users Mailing List -

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

Cisco 7960 has six buttons/lines. Can some of them be configured for fast dialing? If it can't be configured on the phone, how can I configure it on Asterisk? -- Tomislav Par?ina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: tomo@sip.lama.hr e-mail: tparcina#lama.hr http://www.lama.hr

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 <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

Hey all, Has anyone had any issues with phones having multiple lines that are in different contexts? We've got a couple phones that we're testing intercom functionality for, and I'm noticing that for some strange reason, no matter what line we use, the phones tend to be completely in one context or another, not segregated like I would expect. Our contexts look like this: context

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

Has anyone successfully gotten two separate servers to handle voicemail on an NFS shared mount? The main thing I'm concerned about at this point is keeping both systems from writing the voicemail file to the same filename... any thoughts? -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

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

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