search for: sendimag

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 pri...

I tried really quickly the other day to send an image to these phones from the dialplan like this: exten => 2821,n,SendImage(/var/lib/asterisk/images/asterisk-intro) or exten => 2821,n,SendImage(asterisk-intro) It didn't work for me. Should this work? Is anyone else using this with Polycom Phones? Bob

...a. 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 I''m doing in the controller: def sendImage biFiltered = BI.new(100, 100, BI::TYPE_INT_RGB) big = biFiltered.createGraphics() $w = 0 $h = 0 begin filename = "#{RAILS_ROOT}/public/images/kids.jpg" imagefile = java.io.File.new(filename) bi = javax.imageio.ImageIO.read(i...

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

...fault -- dnid = unknown -- enhanced = 0.0 -- extension = 147 -- language = en -- priority = 2 -- rdnis = unknown -- request = agi-test.agi -- type = H323 -- uniqueid = 1065783976.0 Answering1. Testing 'sendfile'...PASS (0) 2. Testing 'sendtext'...PASS (0) 3. Testing 'sendimage'...PASS (0) -- Playing 'digits/1'. -- Playing 'digits/hundred' -- Playing 'digits/90' == Spawn extension (default, 147, 2) exited non-zero on 'H323/ip$192.168.0.100:3016/2107' PASS (-1) 5. Testing 'waitdtmf'...FAIL (unexpected result '...

...ntext = sip-gw -- dnid = unknown -- enhanced = 0.0 -- extension = 126 -- language = en -- priority = 5 -- rdnis = unknown -- request = agi-test2.agi -- type = SIP -- uniqueid = 1074011198.0 1. Testing 'sendfile'...PASS (0) 2. Testing 'sendtext'...PASS (0) 3. Testing 'sendimage'...PASS (0) -- Playing 'digits/1' (language 'en') -- Playing 'digits/hundred' (language 'en') -- Playing 'digits/90' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/million' (la...

...ct to the current channel xi. Directory: Provide directory of voicemail extensions xii. Exec: Exec(Appname(arguments)) xiii. MeetMe: MeetMe conference bridge xiv. MP3Player: Play an MP3 file or stream xv. MusicOnHold: Play Music On Hold indefinitely xvi. Record: Record to a file xvii. SendImage: Send an image file xviii. SendText: Send a Text Message xix. SendURL: Send a URL xx. System: Execute a system command xxi. Transfer: Transfer caller to remote extension xxii. VoiceMail: Leave a voicemail message xxiii. VoiceMailMain: Enter voicemail system d. Scripting i. DialPlan...

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

Hi all. I've been reading through Wi-Ki and at the extensions.conf file description (http://www.voip-info.org/wiki-Asterisk+config+extensions.conf) The author says this: "One day, someone is going to write a proper scripting language for Asterisk that can understand a simpler, easier (and more traditional) scripting syntax. All it would need to do is translate the "high

...der.) == Registered application 'HasVoicemail' == Registered application 'HasNewVoicemail' [app_ices.so] => (Encode and Stream via icecast and ices) == Registered application 'ICES' [app_image.so] => (Image Transmission Application) == Registered application 'SendImage' [skipping app_intercom.so] [app_lookupblacklist.so] => (Look up Caller*ID name/number from blacklist database) == Registered application 'LookupBlacklist' [app_lookupcidname.so] => (Look up CallerID Name from local database) == Registered application 'LookupCIDName'...

....so] => (GSM/PCM16 (signed linear) Codec Translator) == Registered translator 'gsmtolin' from format gsm to slin, cost 1 == Registered translator 'lintogsm' from format slin to gsm, cost 1 [app_image.so] => (Image Transmission Application) == Registered application 'SendImage' [chan_local.so] => (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) [format_sln.so] => (Raw Signed Linear Audio support (SLN)) == Registered file format sln, extension(s) sln|raw [app_lookupcidname.so] => (Look up CallerID Name from...

Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in