similar to: How to escape characters in Dialplan

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 Tinc group, I have recently got an adsl connection at home, which only provides a single static ip address. I want to set up a tinc vpn from home to my workplace. I am trying to decide what hardware I need to do the above. At the moment, I am aware that I could use an Alcatel Speedtouch USB modem directly connected to a Linux vpn box at my home. ( Wall socket---adsl cable---Speedtouch---usb

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 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

I have sent a Polycom IP500 phone to an overseas remote user who has a Speedtouch adsl modem/router. The phone is connected to the 4 port router which performs NAT for the network. The phone rboots, find the server, downloads it's config and tried to register. Then it immediately becomes UNREACHABLE. Jul 26 17:44:41 NOTICE[21026]: chan_sip.c:9138 handle_request_register: Registration from

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

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

>> No I am using the EN5861, just that we have two ADSL lines and only >> one >> EN5861 and rather than buying another I''d like to use linux to do the >> same job as the en5861, if you know what I mean. > >You can, but since you have two ADSL lines, you''ll need two ADSL devices of some kind, whether it''s a pair of 5861s, or a 5861 and an

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:

Our users are experiencing some unacceptable delay when trying to have a conversation. The delay is so noticeable that they keep stepping on each others words and resort to calling the customers via cell phone. Here is the setup SDSL Connection (PPPoA) Speedtouch 610 SDSL Modem 3Com 2224PWR Plus Switch (phones on separate VLAN) 8 Cisco 796 Phones All connecting to a remote Asterisk Server. We

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

Refering to pppoe i have next problem : I asked my isp if i can buy a class of real ip`s to be routed by them. They said elegantly it can`t be done . I want opinions . I am using an ADSL connection through a Speedtouch 510 configured in bridge. About Bgp : i asked someone if i can peer 2 different locations on 2 different ip`s using private asn number and he said yes , and what i don`t

We have newly installed ADSL/speedTouch 510 modem. After the installation, we are getting problems with out Win2K machines. On out NT workstation everything is OK, but on the Win2K we are getting following error, "The semaphore timeout period has expired" whne trying to connect to the domain (GARAGET), My Network Places Entire network MS Windows Network Garaget (our

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