Re: [Asterisk-Users] Nortel i2004 firmware upgrade.
[EMAIL PROTECTED] wrote: I've been trying to look up information on upgrading firmware on a nortel i2004 ip phone. I have this phone leftover from a trial, and it's supposed to be upgradable to current firmwares. Since I also run a DMS I was able to login to nortel's site and get all the firmware files, but All the NTP's regarding firmware upgrading these are how to tell you BCM to send the file to it. I'm trying to use this with asterisk, and was wondering if any of you reading would have information like that. Mark It looks like you set a flag on your unistim server, which then sends a message (using unistim) to the phone, which then (T)FTP's it off the specified location. In other words, good luck. Jeff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Soft Phone
Bill Ford wrote: Does anyone have any experience with an Asterisk compatible softphone application which meets the following criteria: 1) Is able to use touch screen rather than mouse for on-screen functions. Most touch screens have a pseudo-mouse driver. Simply set up the pseudo-mouse, and you're done. 2) Has an API which can be used to export Caller ID info to another App on the same compuer. Thanks Bill Jeff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A good SIP receptionist phone
Pedro wrote: What I did once was create an announcement that got played to the receptionist announcing who the call was for based on the number that was called. This allowed the receptionist to know which greeting to recite. Why not turn that around? Have the receptionist record a greeting for each company, IE: Hello this is company X, how may I help you? Welcome to The Y Corporation, how may I direct your call? Z Corp! How can I help you? that then gets Play()ed to the customer when the receptionist picks up the call. Saves wear and tear on the receptionists voice, so they'll thank you, and it gives them a hint as to whicch company has been called. Jeff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk POE
Mike Robinson wrote: If you are moving from a PBX to * and you are buying POE just to power IP phones, you should at least check out another option called a Handset Gateway. It converts your existing PBX phones and existing wiring into SIP so the PBX phones look and work just like SIP phones running on *. The old PBX goes away and * becomes your IP-PBX, but you don't have to go buy new IP phones and install all the POE gear to power them. Just go Google for Handset Gateway and you'll find some suppliers. It might have been nice to have let the guy know you work for Citel, the manufacturer of the product you're pushing. Kinda changes perspective from advice to advertising, ya know? Jeff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nortel Option 11
Friend, George E. wrote: Question...I'm fairly new to Asterisk, but one location I'm looking at deploying Asterisk has an Option 11 in place already (it's actually in someone's HOME - long story). Does anyone know if it's feasible to interconnect the two and use Asterisk to interface with the other offices and lines, and merely use the Option 11 as a gateway to use the existing digital handsets (Nortel)? George I am currently using a TDM04B connnected to 4 2500 lines, and a T100P connected to a NT5D14 line side T1 card(program them like 2500 sets). I will shortly be implementing a new T1 trunk to split our office side voice off onto asterisk. The TDM04B and T100P service multiple ACD queues, some of which handle Voicemail for external employees, and some of which provide IVR services for human operated ACD queues. You can also use a 4 port T1 card as a go between for your CO T1s and your switch. Simply set up one port as CPE (this connects to the telco T1s), and one port as NET (this connects to your switch). Then set up asterisk to send any calls that you don't want asterisk to handle down the appropriate pipe (inbounds would go up the NET T1, and outbounds up the CPE T1). If you look around voip-info.org you should find where someone documented doing that with a Norstar. Jeff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nortel i2004
Stefan Gofferje wrote: Hi folks, has anybody knowledge about the Nortel i2004? Nortel calls it Internet Phone. I'm curious, which protocols it may understand... Regards, Stefan UNISTIM. It's a nortel proprietary protocol. Jeff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: How do I match a D? (Was: RE: [Asterisk-Users] In-band disconn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
I just have one problem with that. I have _never_ (that I know of) seen a phone with A-D on it! They were (are?) mostly used for military installations. Jeff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nortel i2004 support asterisk?
Ing. Ignacio Ortega A. wrote: Hello everyone i simply just asking if the Nortel i2004 telefhone can work with asterisk if it so HOW? Thank You Simple Answer: No. the i2004 uses the proprietary nortel UNISTIM protocol. Asterisk uses SIP, IAX, SCCP, H.323, but not UNISTIM. Complex answer: It depends on how much you really want it. There has been an open-sourced implementation of a UNITSTIM server done by Cedric Hans. It is located at http://www.mlkj.net/UNISTIM/voi.tar.bz2 (Note: I have not tried it myself yet). With some work it could be modified to provide a channel driver, but Cedric would have to disclaim his code in order for it to be entered in the Digium CVS. Jeff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 in aging Dell Optiplex
Ronan Mullally wrote: I've got an old Dell Optiplex (Pentium-II, 1998 Vintage) which is successfully running an X100P card. I'm hoping to upgrade to a TDM400. Has anybody tried running these cards in old Optiplex machines? I'm not particularly worried about horsepower - more about the motherboard having a PCI bus that's able to power up the card... -Ronan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Got one running in an Optiplex GX100. Works fine. Jeff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Prompt Info
[EMAIL PROTECTED] wrote: I am trying to put together a list of 'departments' to request as voice prompts. I have the biggies (sales, accounting, shipping, etc...) but I want to make sure I do not miss any. If anyone anyone has some suggestions (Ha... that is like going to an NRA meeting ans asking if anybody has a gun :-) ) please forward them to me (and / or post here although, with the volume of this list I do not always have time to read every digest so the 'and' option may be best.) so that I can compile a single list, verify that they are not already available, group them, and send them on. Please put 'voice prompt' in the subject line of anything you forward me so that I am less likely to miss it. I am looking for titles that fit into the string: press 1 for the DEPT department or press 1 for DEPT but if you have other suggestions, let me know. I will be collecting these for about a week so please try to get them to me in that time frame. I am hopeful that, with these prompts, it will be possible to make a complete (albeit fairly generic) tree, all with the same voice. Thanks; James alspachfam at charter dot net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Operations Call Centre Large Order(s) Government Order(s) jsp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem flashing zap channel.
Hello All, I've got Asterisk CVS-HEAD-11/03/04-14:36:44 installed and running. I have a TDM04B (wildcard with 4 FXO modules) using fxs_ks signalling (I'm under the suspicion that my lines (2500 lines from a Nortel Option11 PBX) are merely loopstart, but that's a side issue (which, if anyone knows for sure, would be nice to know)). I have a fairly simple extensions.conf consisting of: [open] exten = s,1,Wait(1) exten = s,2,Answer exten = s,3,MP3Player(/var/lib/asterisk/sounds/P73_hoho.mp3) exten = s,4,Flash() exten = s,5,SendDTMF(1300) exten = s,6,Hangup [closed] exten = s,1,Wait(1) exten = s,2,Answer exten = s,3,Background(beep) exten = s,4,Flash() exten = s,5,SendDTMF(7112) exten = s,6,Hangup [default] include = closed|00:00-16:00|mon|*|* include = closed|02:00-16:00|tue-wed|*|* include = closed|02:00-16:00|thu|*|* include = closed|02:00-11:00|fri|*|* include = closed|04:00-11:00|sat|*|* include = closed|04:00-11:00|sun|*|* include = open When I make a call during the open hours, no problems. and I get this output from the console: Connected to Asterisk CVS-HEAD-11/03/04-14:36:44 currently running on asterisk1 (pid = 1421) Verbosity is atleast 3 -- Remote UNIX connection -- Starting simple switch on 'Zap/3-1' -- Executing Wait(Zap/3-1, 1) in new stack -- Executing Answer(Zap/3-1, ) in new stack -- Executing MP3Player(Zap/3-1, /var/lib/asterisk/sounds/P73_hoho.mp3) in new stack -- Executing Flash(Zap/3-1, ) in new stack -- Flashed channel Zap/3-1 -- Executing SendDTMF(Zap/3-1, 1300) in new stack -- Executing Hangup(Zap/3-1, ) in new stack == Spawn extension (default, s, 6) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' asterisk1*CLI HOWEVER; when I call during the closed period, it doesn't go so well: Connected to Asterisk CVS-HEAD-11/03/04-14:36:44 currently running on asterisk1 (pid = 1521) Verbosity is atleast 3 Asterisk Ready. -- Remote UNIX connection -- Starting simple switch on 'Zap/3-1' -- Executing Wait(Zap/3-1, 1) in new stack -- Executing Answer(Zap/3-1, ) in new stack -- Executing BackGround(Zap/3-1, beep) in new stack -- Playing 'beep' (language 'en') -- Executing Flash(Zap/3-1, ) in new stack Nov 10 16:09:40 WARNING[1535]: app_flash.c:82 flash_exec: Unable to flash channel Zap/3-1: Device or resource busy == Spawn extension (default, s, 4) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' asterisk1*CLI I put the Background(beep) in because asterisk wasn't even picking up the line previously. I'd tried a NoOp before that, and it made no difference. Can anyone tell me how to make this work? Any patches (in or out of CVS) to app_flash.c that relieve this issue? I'll be trying a 1/4 second silent mp3 shortly. Thanks. Jeff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
