similar to: XML How To for Cisco 7960

There is no release note, just a text file that says AASTRA TELECOM INC. February 2006 FC-000046-01-07.st - 9133i Generic SIP Firmware 1.3.1.1095 for customer release. ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gareth Owen Sent: 15 February 2006 02:00 To: asterisk-users@lists.digium.com

Does anyone have a good resource to learn how to program the soft and hard buttons on a Cisco 7940 or 7960 phone? Using SIP Firmware...thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060303/e5e63834/attachment.htm

Hi, I'm having some problem with realtime: let's say I have a dialplan like this [globals] IPPHONES=_3XX [sip] exten=>${IPPHONES},1,Answer A call from ip phone 300 comes in, and it's been answered. Then I "switch" the sip context to realtime, putting the exten in the db and using the variable name for this as in the file version. [globals] IPPHONES=_3XX [sip]

hi.. i am try tu run vvvv, a visual language programm, that run only for wind.. but when i run the vvvv. exe the software don t go for this error ole error 80131700. and in the terminal this error> wine vvvv.exe* machine fixme:heap:HeapSetInformation (nil) 1 (nil) 0 err:ole:CoGetClassObject class {88d96a05-f192-11d4-a65f-0040963251e5} not registered

does anyone know howto set the softkeys of an Cisco 7960 running on an asterisk server via chan_sccp ? Mit besten Gr??en Dirk Rieger

I don't know if this is already common knowledge, and it's not specificly for Asterisk, but if you are using Cisco phones and want to roll XML applications, make sure you have "Connection: close" in your HTTP header. Without it screen loads are very sluggish. In PHP, do: header("Connection: close"); I whipped up quick-and-dirty PHP/MySQL/Cisco XML directory and

I encountered the following problem, can anyone help me? when i use "nmblookup -B 192.168.1.255 redmaple" on my linux box named redmaple, something like "(tdb(null)): could not open file /usr/local/samba/var/locks/unexpected.tdb. No such file or directory" is output, and the name lookup failes. my samba is 2.2.3a, IP address of redmaple is 192.168.1.5 rgds aloha

Hi, We are using Asterisk 1.2.1 with Cisco 7940 and 7960 phones. Most things are running fine ;-) But, when you are calling and you want to Transfer, you need to press first on the 'more' button (4th), then you have the label 'Trnsfr' to Transfer. these are the lables on the softkeys when having a phone call: "Holt / EndCall / Confrn / more" press more and you get

Hi I have two different IP addresses (in a block of /29), one is on port 0 and the other is on port 2 of a Ciso 888. I am doing this so I can have two different certs with two different ip addresses. I have tried: 1) one machine, two real interfaces, two cables (eth0 and eth2) 2) one machine, one real interface eth0 and one virtual interface eth0:1, one network cable Using number 2) I can

Hi, I tried set up a global var for an extension, like this [globals] IPPHONES=_3XX [sip] exten=>${IPPHONES},1,Answer What I would like to do is to make a dialplan without fixed extension numbers to change the entire dialplan according to the customer requests: what exten number do you want for your IP Phones ? let me change a variable and we are set! It seems that this is not supported,

Hello *'s, I have question regarding Quintum Tenor AXT800 VOIP gateway can anyone integrate it with asterisk if anyone what is the scenerio i have scenerio which is quite simple but i am confused about it whether it is possible or not : I integrate it with asterisk for interanet no PSTN at all just only IPphones and analog phones connected on FXS port.Is it's neccassary to cannect with

Aloha, Does anyone have a good source for Polycom SoundPoint? IP 600/500/300 phones? Everyone sells Cisco 79XX. Aloha, Matt

I've got a problem that keeps popping up with my reception phone. It is a IP 650 and the receptionist - on three occassions - has accidentally hit the "Forward" softkey just before she enters the "Page All" keystrokes and then all future calls get routed as an overhead page. I will admit, the first time it happened, I was totally stumped. Why the heck did I have

I found on the web that there is way to customize the softkeys for the 7941/7961 phones. In the SEP<mac>.xml there is a section called "softKeyFile" where you can specify an xml file for the softkeys. I couldn't find any examples of this softkey file or the format to this file. Does anyone have a call manager enviroment where they can look at this file and send me an

Aloha, We are looking to roll a solution that will have the following network layout: ISDN-PRI <--> Asterisk <--> T.38 <--> ATA <--> Fax Does version 1.8 with the Digium fax driver have this capability? I like 1.8 because it is a long term support version. What ATA's are people using? Any working solutions would be great! Aloha, Matt

Is there any way to program a soft key in ADSI to put a caller on hold. Then able to retreive that caller. Example - Softkey Hold Softkey Retreive Call Softkey End Call -gcc

Aloha, I have an application to build an eight conference bridge for a Norstar. I would have a converter to turn the digital ports into analog with disconnect supervision. On the digium site I only saw a single port FXO card. I would like to use a higher density FXO card like a D/41 or newer from Dialogic. I noticed in the handbook 2 (draft 3/30/03) that there is a price to use dialogic cards

Aloha, Are there any hard phones that support ILBC? I see lots of soft phones but not Cisco, Polycom etc. Aloha, Matt

Aloha, Did anyone ever get the formating manual for the XML brwoser on the Polycom IP600? Does anyone have a sample? Aloha, Matt

Aloha, Anyone know how to disable call forward on a Polycom Phone. Calls being accidentilly being forwarded somewhere is the #1 trouble that we have to respond to. The real issue is the 'end call' button becomes 'forward' when the call ends....therefore the user thinks they are pressing 'end call' but the call ends just before they press the button so they end up