similar to: displaying call progress with SendText on a Snom

Hi, Is is possible to "translate" non-english text into ASCII text so that SIP phones would correctly display non-ASCII characters received from SendText() ? I think SIP MESSAGE (rfc3428) on which SendText() currently relies, defines "text/plain" Content-type but googling, I can't find a source describing what text/plain can or cannot be. Regards -------------- next part

Hi, we've proof of conecpt system for speech recognition on Asterisk. We would like to send results of recognition back to user in standard way. Currently we're considering using sendtext command and it works with Firefly. But I'm curious what soft or hard ip phones that can connect to Asterisk support such feature ? Also what softphone would be most suitable for further work in

Hiya, i've just noticed that chan_misdn writes the AOCD information into a logfile. Has someone done a patch that sends this information via sendtext() to the active channel? At least some phones (like Cisco with chan_sccp and the snom-phones with SIP) can show this information on the display, what is kinda the idea with AOCD ;-) Regards, Andeas

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 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, I'm using Asterisk 1.6.2.0 and I like to use escape characters with SendText, because I can just delete the message from my phone (Thomson Speedtouch ST2030) display by sending a return-char (\n). But \n is not escaped: I tried already: exten => 222, n, SendText(\n) exten => 222, n, SendText("\n") exten => 222, n, SendText('\n') exten => 222, n,

Hi, I understand SendText() sends text on the current channel. Is there a way to manipulate this feature to SendText toward another SIP device? I use Polycom IP600's. Local sendtext works fine. Would be nice to drop an instant message on another user's phone. thanks! Mark

Hello! I tried using sendtext() in the Asterisk dialplan to send a SIP MESSAGE to Bria or a recent Eyebeam on my mac. I know it used to work, but right now I get "100 trying" and nothing else from the softphone. Anyone that knows what's going on here? Thanks, /O

Hi, I'm using a SIP phone (Thomson ST2030) which is able to display text received though Asterisk's SendText() application. I'm using this to display from Asterisk "Forwarded to 0123456789" whenever a user forwards his calls to another number or extension. Test is displayed with white letters on black background. What I can't do at the moment is erasing this

Hi, I'm having trouble configuring Asterisk to respond to an incoming out of call SIP MESSAGE. The transport protocol is TLS and the Asterisk version is 10 (it's old, but I'm kind of stuck with it at the moment). Currently I have roughly the following configuration and handling: sip.conf: [general] accet_outofcall_messages=yes outofcall_message_context=sip-im and extensions.conf

Someone posted a message suggesting someone try sendtext() and so I thought I'd see if it was useful. Much searching through help at the CLI has failed to find any help for sendtext, but I did find that: "core show function vmcount" fails but: "core show function VMCOUNT" works. Is that a bug and if so, has it been reported? Ira

Hello all, I am running Asterisk 10.10.0 and I can send Message between SIP's no problem. However, I would like to be able to send send Message to a SIP from AMI Or CLI. I check the ListCommands On the AMI and it don't have MessageSend. Therefore, I try the SendText. AMI: Action: SendText" Channel: SIP/600" Body: This is a test. Message: This is a test. Extension: 600";

Sorry for the delay here. For some reason the mail from Joshua Colp failed to deliver to my mailbox. So, anyway, I've set up a local scenario on my computer a PJSIP client and Asterisk 11.17.1 (On a fedora linux workstation) with the settings listed below. In this scenario I've used UDP, but I want a configuration that can be used with any transport protocol. I can see that the context

On Mon, Sep 21, 2015 at 9:45 AM, D'Arcy J.M. Cain <darcy at vex.net> wrote: > On Mon, 21 Sep 2015 06:48:52 +0000 > Emil Ohlsson <emo at svep.se> wrote: >> [sip-im] >> exten _X!, 1, NoOp(Got message) >> exten _X!, n, Answer() >> exten _X!, n, Agi(agi://localhost/messagehandler.agi?...) >> exten _X!, n, SendText(Message received) > > I am not

How do I receive text sent from SendText() application? Asterisk lists text capability, so SendText() is successful. But I don't see an application to actually use it. Yuan Liu

Hi, I'm trying to pass a single carriage return (0x0d in hexa) to Sendtext in an AEL2 file like this : SendText(${BASE64_DECODE(DQ==)}); Value sent (8 bytes long) is very strange : Content-Type: text/plain;charset=UTF-8 Content-Length: 8 ?ez?== Any workaround ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've been working with the old-style (pre-SIP) MultiTech MultiVOIPs, trying to make them work against chan_h323. With the voips in rfc2833 mode, Asterisk can detect DTMF fine from them, but when it sends DTMF to them, they lock up on the second digit, crying about an incoming fax. Has anyone encountered that problem (and found a solution)? Taking the other route, switching DTMF to inband on

Hi, can a call a php script wich is located in a remote server, someting like calling www.theserveraddress.com/scripts/validate?code=234234swq and get the result which this script generates (a 0 or a 1) back in the dial plan in a direct way or should I create a script which in turn does this? I'm using * CVS HEAD. Also I searched for this for I while but didn't manage to find anything

Hello, I'm experimenting with Asterisk for possible use in a call center. I'm trying to figure out how to run applications when an agent answers a call in the queue. I see that the queue itself supports a very limited range of applications; for example, I can give a URL to the Queue() application to SendURL(), or an announcement to read to the agent. I'd like to do some slightly

All, I am intersted in having 2 extensions for the same sip phone... The reason is when I am using the outgoing spool file to initiate a SendText() command to my SIP phone I dont want voicemail to answer the call and my text message go there. I want my outgoing call to FAIL and then I can try again some time later to deliver the message. Any suggestions on how to do this is appreciated.