similar to: Why I get repeat messages many times

Thanks again. How do you create that message context in extensions.conf? On Mon, Nov 16, 2015 at 9:44 PM, Thyda ENG <engthyda at gmail.com> wrote: > According to what I have done , I add the message_context to the > pjsip.endpoint_custom.conf in /etc/asterisk and then I create that > message_context in the extension.conf, and it works. > > On Tue, Nov 17, 2015 at 9:34 AM,

Hi, when sending IMs from endpoint to endpoint with the MessageSend() application, I can check the MESSAGE_SEND_STATUS and send another message to the sender of the message to notify them that their message was not sent when the status indicates it. This works fine with chan_sip. With chan_pjsip, this works differently in that MESSAGE_SEND_STATUS is "SUCCESS" after sending the

Hi, I'm running into a strange problem with commas is variables. I have the following contexts: [messages] exten => _+.,1,Noop(External SMS) same => n,Set(ACTUALTO=${CUT(CUT(MESSAGE(to),@,1),:,2)}) same => n,Macro(goip_sendsms,${ACTUALTO},"${MESSAGE(body)}") same => n,Hangup() [macro-goip_sendsms] ;Call Macro(goip_sendsms,number,message) exten => s,1,Noop(SMS

On Thu, Jan 30, 2020 at 9:35 AM hw <hw at gc-24.de> wrote: > Hi, > > when sending IMs from endpoint to endpoint with the MessageSend() > application, > I can check the MESSAGE_SEND_STATUS and send another message to the sender > of > the message to notify them that their message was not sent when the status > indicates it. > > This works fine with chan_sip.

Hello Mongrels, Release candidates for Mongrel 1.0.2 and its dependencies are now available. Note that gems are now signed, so please add the Mongrel public certificate via: $ wget http://rubyforge.org/frs/download.php/25325/mongrel-public_cert.pem $ gem cert --add mongrel-public_cert.pem Now you can verify and install the candidates via gem install: $ sudo gem install mongrel

Hi all, We''ve installed RoR on Solaris 9, and are attempting to install Mongrel using the following command:gem install mongrelgetting error "SSL is not installed on this system".All other gems (including Rake) installed fine.Any suggestions?Thanks,Brian _________________________________________________________________ Boo!?Scare away worms, viruses and so much more! Try

I have many endpoints and each endpoint has some parameter in common so i wonder is there any way to config one for all endpoints? Like in my example I have two endpoints and I repeat the same thing, [100] type=endpoint aors=100 auth=100-auth allow=ulaw,alaw,gsm,g726 context=from-internal callerid=device <100> dtmf_mode=rfc4733 use_avpf=no ice_support=no

So, the only thing that is needed in the endpoint definition in pjsip.conf (there is no such file pjsip.endpoint_custom.conf) is *message_context=astsms* Is that correct? Anything I need to do in extensions.conf? I see that the messages are received at Asterisk (when I turn on pjsip set logger on) but they are not delivered to the other endpoint. What gives? Any help appreciated. Thanks! On

The Asterisk Development Team would like to announce the release of Asterisk 13.32.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.32.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 13.32.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.32.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Hello, I am looking for documentation support for enabling instant messaging between endpoints using Asterisk 13.1.0 and vanilla VoIP clients such as Zoiper. Where do I enable this support on the server side and does it need anything on the client side? I see plenty of online help for chan_sip, but nothing for chan_pjsip. I imagine there is both pjsip.conf configuration and extensions.conf

The Asterisk Development Team would like to announce the release of Asterisk 16.9.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.9.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 16.9.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.9.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 17.3.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 17.3.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 17.3.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Hi All, I have configured WebRTC according to the install document. The clients register correctly. I'm use SIPjs. The clients are able to send messages to the server. The SIP debug shows the messages being received. However I'm stumped for directions on how to route the messages between the clients. Asterisk 11.11.0 Here is my client sip config: [1060] type=friend username=1060 ; The

how if I use the auto generate once from freepbx ? On Tue, Sep 22, 2015 at 10:12 PM, Ishfaq Malik <ish at pack-net.co.uk> wrote: > > > On 22 September 2015 at 16:04, Thyda ENG <engthyda at gmail.com> wrote: > >> I have many endpoints and each endpoint has some parameter in common so i >> wonder is there any way to config one for all endpoints? Like in my

Hi all, Release candidates for *nix 1.0.2 and dependencies are up at http://mongrel.rubyforge.org/releases/ . Public cert is at http://rubyforge.org/frs/download.php/25325/mongrel-public_cert.pem . Install with: sudo gem install mongrel -P HighSecurity --source=http://mongrel.rubyforge.org/releases/ etc. Luis is working on win32 builds. Evan -- Evan Weaver Cloudburst, LLC

tengo la siguiente pagina pero no se como seguir despues del punto 22 http://highsecurity.blogspot.com/2012/12/webrtc-and-asterisk-11-using-sipml5.html gracias! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141112/59751a87/attachment.html>

Hi all, I am trying to enable SIP SIMPLE communication in my test environment. I have the following env : - one server (192.168.50.126) with Asterisk 11 - 2 clients using pidgin : demo-bob and demo-alice on my 192.168.50.143 I successfully had a phone call between clients. I used the following link to enable SIMPLE messaging between my clients :