similar to: How to MessageSend to a SIP from AMI Or CLI?

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

After upgrading from asterisk-10.5.0 to asterisk-11.1.2, I am getting a Segmentation fault. [root at localhost asterisk-11.1.2]# asterisk -vvvvvvc Asterisk 11.1.2, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components

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

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

Hello, After Upgrade to Asterisk 11.1.0-rc1 I keep getting == Using SIP VIDEO TOS bits 136 == Using SIP VIDEO CoS mark 6 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [603 at DLPN_AlDimnaDialPlan:601] Dial("SIP/601-00000002", "SIP/603") in new stack [Nov 16 06:42:33] WARNING[15547][C-00000004]: app_dial.c:2433 dial_exec_full: Unable to

Hello My provider allows to activate/deactivate a forwarding rule by sending a SIP MESSAGE. This is done outside a call. That is, while there is no ongoing call, a SIP client just sends the following message: MESSAGE sip:543951354657 at callfwd.sip.providerx.com SIP/2.0 Call-ID: b9ba106e-613a-46b9-8a4d-0efb4dc0a0f2 CSeq: 1 MESSAGE To: <sip:543951354657 at

ive installed asterisk but below is what am getting proces gets killed.please help [root at localhost sounds]# asterisk -vvvvc Asterisk 13.13.1, Copyright (C) 1999 - 2014, Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under

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

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

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, 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

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

hello all, my vm (windows server 2008) keep crashing? 2011-09-21 15:00:45.924: 1355: info : libvirt version: 0.9.6 2011-09-21 15:00:45.924: 1355: error : virSysinfoRead:465 : internal error Failed to find path for dmidecode binary 2011-09-21 15:05:43.873: 1349: warning : qemuDomainObjTaint:1128 : Domain id=1 name='IIS' uuid=a02e41a9-9f4d-d94f-43a6-995b1ca6d37d is tainted: