similar to: Configuring CISCO IP 7940 for *

Just a quick note to people looking for SIP firmware images for Cisco phones: To access these files from Cisco's website, you need to have a Service Contract (SmartNet) on at least on of your phones. I though a contract was several hundred dollars, but it is way cheaper! Their lowest level, 1 year Next Business Day 8x5 SmartNet program, costs around $70.00, and gives you unlimited access to

Does anyone know how to send a message to a Cisco 7940/7960 phone running SIP images 6.3 telling it to light up one of its LED's when new voice mail arrives? I found alot of web based solutions http://www.voip-info.org/wiki-Asterisk+GUI and easy ways of getting email or getting paged of a new voice mail - but nothing where you can just look at the phone and see a blinking light or

Here's the problem, I upgraded all of my 7960 phones to SIP. Now my boss wants to carry his phone with him between offices. The other office has CCM which is set up for Skinny. Now I have to put SCCP back on a 7960 phone and it won't take it. Does anyone have an example of config files for sccp. Are they the same as config files for SIP? I've never had to go back to Skinny once I

Hi, An Cisco 7960 ipphone has been set to SCCP firmware by one of our students. I want to set it to 7.5 SIP firmware and I've been unsuccessful yet. Firmware versions are SCCP 3.0 (Source: http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/mgcp/frmwrup.htm#wp1045789) ie: Application Load P003F300 Boot Load ID PC030300 When I browse, phone settings, I see

I am trying to figure out the correct syntax for the CLI command "SIP SHOW CHANNELS". I have tried SIP SHOW CHANNELS SIP/200 and I've even tried to do this when a call is connected such as: -- Zap/15-1 is ringing -- Zap/15-1 answered SIP/206-4299 asterisk*CLI> sip show channel SIP/206-4299 No such SIP Call ID 'SIP/206-4299' I always get the "No such SIP

hi, i saw the cdr file called Master.csv and i want to know what these represent. examples "","","4","incoming","","Zap/1-1","Zap/4-1","Voicemail","u8888","2003-11-07 17:43:04","2003-11-07 17:43:04","2003-11-07 17:43:22","ANSWERED","DOCUMENTATION"

Does anyone have a document I can use as a guide on how to load a SIP image on a cisco 7960 phone? Ryan

Hi All. I'm working on an * configuration. We require 8 inbound POTS lines, and CT1 or PRI seems like it will be quite expensive at that level. I've read that a T1 Channelbank plus the T100P would be a (the?) way to go for this situation. What is the recommended channelbank for use in this scenario? From searching the archives I see a lot of suggestions to get "a

I'm trying to fix a problem with the GrandStream Budgetone 102. I've been reading the source code, mailing lists and other resources. Here's the scenario and the approach I have been pursuing. I'm having some problems with the AGI calls and I hope someone can give me some clarification. PSTN <---> T1,PRI * <---> Grandstream BT 102 (12)

I have a dialplan that requires that we use * to send the long distance access code to AT&T. I have found in the list that the `w` command can be used to inject a pause, I have tried the following: exten => _91NXXXXXXXXX,1,Dial(ZAP/g1/${EXTEN}www5555555,70) There `5555555` is the ld access code. I tried various quantities of `w`s but I never got * to dial the ld access code. Allof the

We've got one of the Budgetone phones here, and we can call from any SIP phone, or an outside line TO this phone and the conversation sounds great for bothways, not a bad delay, no echo problem, etc. But when we pick up the Budgetone and dial an outside line or another SIP phone the person on the Budgeton just sounds really choppy and there is a slight delay. We've messed with

To recap: T100P card wouldn't sync with the telco using line side clocking ( span=1,1,0.........) Had to use internal clocking (span=1,0,0.......) zttool showed Tx/Rx Levels as 0/ 1 For the grins of it I replaced the T100P card with another newer card from inventory. This newer card has the same rev on the ASIC / FPGA but doesn't have any of the various jumper headers installed

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040112/e9dceeb4/attachment.htm -------------- next part -------------- Hello One of the option in VoicemailMain() is "Adavance Options". Could anyone explain what are these ?. Because whenever I select Advance Options, it repeats the same process of asking "Change Folders,Advance

Should FAX transmission generally work through Asterisk and a TDM400P connected through a PSTN gateway? At first blush I'd think that if they're all g.711uLaw encoded that it would work. But experience shows otherwise. Is there a better way to do FAX? -brian

Hi, can anyone recommend a Linux based TFTP server to go on an asterisk box? Thanks in advance Robb

Hi. Can anyone help me with the following: [globals] OFFICEHOURS .................................... [internal] exten => *80,2,SetGlobalVar(OFFICEHOURS=100) exten => *80,2,SetGlobalVar(OFFICEHOURS=200) .................................... [incoming] exten => s,1,GotoIf($[${OFFICEHOURS} = 100}]?incoming-officehours:incoming-officehours-off 1. Am I using the right sytanx when

Does anyone have a working way of having a Cisco 7960 reload its config remotely. I have tried some of the scripts that I have found on the web, but to no avail. Thanks for the help. B. J. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040116/aa4eda3c/attachment.htm

Dear all, I have the following lines in my extentions.conf file; ;All US Calls exten => _9001XXXXXXXXXX,1,Dial(IAX2/dornoch:xxxx@10.xx.xx.xx/${EXTEN:1}@outbound) ;Dial 9 for outgoing numbers exten =>_9.,1,Dial(Zap/g1/${EXTEN:1}) ;include Brunswick switch => IAX2/dornoch:xxxx@xx.xx.xx.xx/sip What I'm trying to do is to send any calls starting with 9001 out through

I've managed to use GhoustScript (gs) to take a postscript file and convert it to tiffg3, but I CANNOT seem to make it merge multiple files. Here is the output from tiffinfo on the file that SG generates: fteTYGeh2v.tif: TIFF Directory at offset 0x8 Subfile Type: multi-page document (2 = 0x2) Image Width: 1728 Image Length: 1056 Resolution: 204, 96 pixels/inch Bits/Sample: 1

Hi all, Having fixed my problems with the call waiting ringing on the GS phones, I needed to extend that with a campon facility (available on some legacy systems - sort of callwaiting without phone ringing). I've managed to implement that by adding/modifying app_queue.c. Basically, when calling the SIP phone, and if busy, I can camp the call onto that extension, with MOH, allowing the caller