Re: {Scanned} [Asterisk-Users] Asterisk Voice mail-reg
nr k wrote: HI all How to configure voice mail in asterisk . pls do the needful. regards ramakrishnan.n here is the wiki page on voicemail http://www.voip-info.org/wiki-Asterisk+config+voicemail.conf Tom -- This message has been scanned for viruses and dangerous content by Cache Communications, and is believed to be clean. Thank You For Choosing Cache Communications ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: {Scanned} [Asterisk-Users] aastra.cfg mac.cfg examples Firmware version 1.3
Lists wrote: I have gotten the tftp server working and the 9133i is doing a firmware update and finds the aastra.cfg file as well as the 00XXX.mac file. The issue is that I can't figure out what is wrong in the configuration files that it is not loading the extension, proxy, etc. info. Could someone post their aastra.cfg file and mac.cfg file for a 9133i and/or 480i phone as well? I would like to see a working copy of each and compare it to what I am doing wrong. Thanks. Here are my files I have a 480i i think the files should be the same aastra.cfg dhcp: 1 tftp server: xxx.xxx.xxx.xxx sip silence suppression: 2 # 0 = off, 1 = on, 2 = default sip line1 proxy ip: xxx.xxx.xxx.xxx # IP of proxy server. sip line1 proxy port: 5060 # 5060 is set by default. sip line1 registrar ip: xxx.xxx.xxx.xxx # IP of registrar. sip line1 registrar port: 5060 # 5060 is set by default. sip digit time out: 3 time server disabled: 0 # Time server disabled. time server1: # Enable time server and enter at time zone name: US-Mountain time format: 0 # 0=12 hr, 1=24 hr date format: 0 # 5=dd;mm;yy web interface enabled: 0 #sip dial plan: [2-9]11|9911|1[01]xx|[2-9]xx|1[2-9]xxx|011x+#|xx*|*xx+#|x# #sip dial plan: X+#|XXX+* #sip customized codec: payload=18;ptime=10;silsupp=on,payload=0;ptime=10;silsupp=off sip customized codec: payload=18;ptime=10;silsupp=off,payload=0;ptime=10; silsupp=off directory 1: xxx.csv mac.cfg make sure the mac is all uppercase sip line1 auth name: # SIP Registrar request authorization name. sip line1 password: # SIP Registrar request passcode. sip line1 user name: sip line1 display name: # Name used for SIP messages. sip line1 screen name: # User's name seen on the idle screen of the user's phone. sip line1 vmail: vmext directory 2: xxx.csv softkey1 type: blf softkey1 label: name softkey1 value: (ext ) softkey1 states: idle softkey1 line: 1 softkey2 type: dnd softkey2 label: DND softkey3 type: speeddial softkey3 label: test softkey3 value: (ext) softkey3 line:1 Hope this help Tom -- This message has been scanned for viruses and dangerous content by Cache Communications, and is believed to be clean. Thank You For Choosing Cache Communications ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: {Scanned} Re[2]: [Asterisk-Users] format_mp3 uninstalling mpg123
Alessio Focardi wrote: GK How did you install mpg123? If you installed it with the package GK management system, then use the package management system on your GK OS to remove it. If you installed it manually, you'll need to remove GK it manually. GK To actually allow format_mp3 to work you also need to change GK musiconhold.conf from mode=quietmp3 to mode=files. Regarding this issue: anyone knows how to setup streaming music on hold (from webradios) with the new native syntax ? Previously I was using this as suggested by the wiki: radiowazee= mp3:/var/lib/asterisk/sounds/pbx/webradio,http://grace.fast-serv.com:9206/ where in the webradio dir there was just a dummy mp3 file I would like to reproduce this using native mp3 ... any idea ? Tnx ! google for icecast and you will also need ices. i also found a howto on the wiki it runs great on my 1.0.12 box . hope this help Tom Tom -- This message has been scanned for viruses and dangerous content by Cache Communications, and is believed to be clean. Thank You For Choosing Cache Communications ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: {Scanned} Re: [Asterisk-Users] Caller ID ?
Stijn Jonker wrote:
Hello Tom,
On 26-Aug-2005 7:50, Tom wrote:
Most of the time i can find answers to my questions on the wiki, google,
or searching the list now i am stuck .
I have a small * box at my house running 1.0.9 stable and a devlite kit.
Every thing is awesome VM, IVR, Echo canceling, and Meetme are all
working great.
Nice isn't it?
But on Incoming caller id i need to add a 9 as a prefix to make it
easier to return call from my cordless phone (cheap vtech phone). I have
tried to search the list and also google but i think i am searching of
the wrong thing. If i could get a kick in the right direction that would
be great.
This is what I came up with: (Watch out for linewraps on the second line.)
; Incoming on normal line
; Incoming on normal line
exten = ${EDN_MAIN},1,LookupCIDName(${CALLERIDNUM})
exten = ${EDN_MAIN},2,GotoIf($[$[${CALLERIDNUM} = ] |
$[${CALLERIDNUM} = CID withheld]]?5:3)
exten = ${EDN_MAIN},3,SetCIDNum(9${CALLERIDNUM})
exten = ${EDN_MAIN},4,SetVar(__NETWORK=KPN-Prive)
exten = ${EDN_MAIN},5,Goto(int-dest,${EDN_MAIN},1)
Stijn
Stijn,
Thanks based on my reading on the wiki i thought the the cmd SetCIDNum()
was only for forcing Caller id on a PRI.. :-[ once again thanks for the kick
BTW:: this list rocks :-) so much good info
Tom
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
Thank You For Choosing Cache Communications
___
--Bandwidth and Colocation sponsored by Easynews.com --
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
[Asterisk-Users] re:Digital Phones
Anton, Nortel and Avaya are not ADSI phones. I can only see two options run * on ATA port on PBX or buy new phones. I have seen dialogic cards that act like a nortel digital ext but there are no * Drivers as far has i know. My two cents Tom B -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. Thank You For Choosing Cache Communications ___ 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
