similar to: Respond to an out of call SIP MESSAGE

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

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 15-09-28 10:19 AM, Emil Ohlsson wrote: > (Still no not receiving the mail, revisited the settings.) > > OK, so SendText doesn't work with this scenario. But can MessageSend > handle this, and respond even when the transport protocol is TLS? Or > do I need to modify Asterisk to add this support? MessageSend has no concept of TLS, it gets passed to chan_sip which then sends

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

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

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

Easy questions for you guys probably, I'd like to serve 10 parallell incoming calls at the same time, so I bought a lot of Zap-channel cards for analog phone lines. But I want all users to be able to use the same phone number to dial in, but I want the number to be switched to an avaiable zap-channel. Do I need some kind of switch for this? It sounds reasonable, but I'm not sure. :) Am

I'm trying to play a wav-file on a channel. This is what I see in the asterisk debug console AGI Rx << STREAM FILE "test.wav" "12345" [Aug 20 16:10:19] WARNING[25219]: file.c:602 ast_openstream_full: File test.wav does not exist in any format So it doesn't find the file, or it's in a wrong format? I can listen to it with windows media player... it's a

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";

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

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:

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

Hi there! Trying to originate and dial a number using Zap-8, used to work, but now it just fails. I enabled all debug I found in the source-code and this is the output from asterisk. Can someone understand something from the debug-output what is wrong and direct me to what the problem might be? The setup is correct, trust me, it worked some hours ago, haven't changed anything. Just dialing

Hi is it any tool like show.pl by Stef Coene to generate graph with classes but for HTB regards emil _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/mailman/listinfo/lartc HOWTO: http://lartc.org/

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.

Is there a list of phones (hard or soft) that support the Asterisk SendText or SendURL applications? I have been trying to make this work with a Snom 200 and a Cisco 7960, to display call progress information, such as which trunk a outbound call is routed to, but my attempts have been unsuccessful. The Snom claims to support SMS somehow, but searches reveal no useful information on how it is to

- we used the C4/C5 (first prefix) byte instead of the apparent ModR/M one as the second prefix byte - early decoding normalized vex.reg, thus corrupting it for the main consumer (copy_REX_VEX()), resulting in #UD on the two-operand instructions we emulate Also add respective test cases to the testing utility plus - fix get_fpu() (the fall-through order was inverted) - add cpu_has_avx2,