similar to: SendImage() to Polycom ip550 or ip670

SendImage() in 1.4: ---cut--- SendImage(filename): Sends an image on a channel. If the channel supports image transport but the image send fails, the channel will be hung up. Otherwise, the dialplan continues execution. The option string may contain the following character: 'j' -- jump to priority n+101 if the channel doesn't support image transport This application sets the

Anyone seen anything on the IP670 & the Color Expansion? -Matt

Hi I''m doing some server-side graphics with Java2D and I''m trying to stream the resulting image to the browser using send_data. When I try to view the image that I streamed, the browser just shows the URL in the window as a very small image (about .4 kb) when I am using jpeg format. For png, it tells me that the image has errors and can''t load it. Here''s what

Hi I've just started to play around with AGI scripts and have run into problems. When I run Asterisk in console mode everything works just fine. If I run Asterisk in 'regular' mode (not console) it crashes if I hang up on the script. I have used Python scripts to test this and also the "agi-test.agi" script. (the Asterisk code was compiled from CVS code just a few days

I have several Polycom IP550 phones running UC 4.0.3, connected to Asterisk 1.8. Setting forwarding for "Always" works as expected; the phone issues a 302 Moved Temporarily, and Asterisk shifts the call to the new location. Setting forwarding to "No Answer" means a 302 never gets issued. It just rings and eventually goes to voicemail. Watching with Wireshark, I never see a

Hi, I'm trying to use the say-ani agi asterisk-perl script and am experiencing crashes, I am also experienceing problems with the test-agi scripts shipped with asterisk. The clearest demonstration of the problem is that if I dial extension 125 configured as... exten => 125,1,Ringing exten => 125,2,Wait(3) exten => 125,3,Answer exten => 125,4,Wait(2) exten =>

We are having the strangest issue that I have seen for some time. A customer of ours with Polycom phones (4x ip335, 2x ip550) will occasionally (maybe 1 in 50 calls) hear ringing on the line along with the other party. It has happened on both incoming and outgoing calls across apparently all of the phones. We use ip550 in our office with Asterisk and have never had such a problem (we run the same

Anyone have some recommended equipment for alerting people to calls in a noisy environment? I have Polycom IP550 phones set up in some really noisy environments - our mine hoists - and they tend to drown out the ringers. I'm using Clarity WR100s now. They're analog devices, attached to Linksys PAP2T ATAs as part of a call group to get a loud (advertised as 95dB) ring out there, but it

hi, all is there any way to send image on sip channel ? Regards, -- Bhrugu Mehta Sr. S/W Engineer (D&D) VOIP,Telephony Team India -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100317/3cb8322a/attachment.htm

I'm looking for a way to online resize a Windows disk -- i.e., be able to resize the disk without shutting down, rebooting, or detaching the disk. Is this at all possible? Or am I just barking up the wrong tree? I'm not finding a way to do this and even Amazon has a weird workaround, in which the user must write data to the newly resized-drive in order to recognize the new size

Hello all, I've been invited to do a presentation on Asterisk for the Ohio Linuxfest in Columbus this weekend (http://www.ohiolinux.org). Rough estimates are that nearly 500 people will be attending. I've been working on an outline for a couple of weeks and I would like to have some peer review of the information presented. I am going to have to cut down the content to make it fit in

Hello I am required to do some thing like Dail in modem . User will have to call a modem just like we do in dail up connection ....now we need to handle that request and retrieve some parameters from that send a HTTp request to a web server and then after getting http response send user a feed back .. this is a requirement .. Is it possible ?? what is the way forward ?? please give me a

I'm looking for a nice Open Source web hosting Control Panel. Is anybody using VHCS with Centos4? Any other suggestions for good control panels? We'd really prefer a panel with a MySQL backend for user settings and such. Bob Pierce Systems Analyst Westman Communications Group pierceb at westmancom.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have run a pca and want to plot the samples multivariate space using the sample numbers. In addition, I would like to color code the sample numbers by group, but can't find a way to color-code the sample numbers by group. So, I have to do it in two separate steps: plot(bulbil.pca, col=as.numeric(group.bulbil[,3]), pch=as.numeric(group.bulbil[,3])) # group.bulbil[,3] is grouping

Hello to all I need send a data to sofphones screen when I use a Dial () . Thanks a lot Regards Andres Gomez -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070430/d26033b2/attachment.htm

Hello, fax/ata(ht286) -> asterisk/tdm04b -> pstn fax machine I can fax out from the sip side, but I can't fax in from the PSTN side. When I try to send a fax, asterisk sees the call and show me this: "Redirecting Zap/1-1 to fax extension" "Timeout on Zap/1-1" TCPDUMP doesn't show any activity to the extension that I configured to be the fax machine.

First, the scenarios: Call placed from Boston to locally configured Asterisk Meetme extension: Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Asterisk(Boston) Call placed from Boston to European Asterisk Meetme extension: Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Cisco 2821(CME,Europe) <-SIP-> Asterisk(Boston) In the 1st scenario, everything works

What are the HA options for Switchvox systems? Is it possible to set up redundant systems with DRBD? I know on the digium website they talk about "Optional cold spare failover" What does this mean? Is this an active spare ready for some sort of automated failover? Thanks for you help, Bob

Hi all, I've been googling for a solution here and haven't really come up with anything yet. We're doing an Asterisk install for a local radio station, and we're looking for a phone that they can use in their control room hooked up to their mixer board for recording calls. So, when you phone in for some contest or to request a song they record it and play it back a few minutes

Anyone know of a good package for reporting on Queue statistics from Asterisk? Bob