similar to: AT&T access code entry by Asterisk

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

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)

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"

Can anyone recommend who we can consult with that could provide advice on the legality of a proposed VoIP service. Specifically, we would provide VoIP termination in the USA to clients in Spain, Nigeria and Guana. The termination service would connect the VoIP clients to the PSTN through a carrier like MCI, Verizon, etc. The calls placed would connect anywhere in the world via the USA carrier.

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

Helo WipeOut, I have found a solution for sending dtmf after dial. I use spooling. Take a look at the sample.call file inside asterisk dir. You need to edit this file and dump it in /var/spool/asterisk/outgoing. Asterisk will precess this file automaticlly I create the sample.call do something like this: Channel: OH323/4324324324 #dial the access way MaxRetries: 3 RetryTime: 60 WaitTime: 30

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

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

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

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

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031201/c9e420c5/attachment.htm -------------- next part -------------- Hello all, I have 1 IP 7940 with the following Firmware versions App Load ID: P00303011201 Boot Load ID: PCO303010001 Version 3.1(12.1) Could you please confirm, if my IP phone has the correct SIP image. My asterisk

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

Here's my situation. I have two receptionists that answer incoming lines. Each has a 7960G with 5 incoming lines each. I'm trying to set this up so each line on each phone doesn't utilize call waiting. My problem seems to be that ChanisAvail(Sip/cisco1&Sip/cisco2&Sip/cisco3&Sip/cisco4&Sip/cisco5) always returns cisco1. Here are the sip.conf entries: (mind you,

Hello, My Cisco phone has software: Boot Load: PC030300 Ver: 3.2(7.0) And I want to upgrade it to SIP 6.0 Is it possible or I have to upgrade to ealier then 6.0 and then to 6.0 ? bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031121/7331dff5/attachment.htm