similar to: Sending XML over the asterisk PJSIP

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

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,

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

Hmm, most phones I've used wouldn't have the capability of displaying a bitmap image due to only having minimal monochrome displays. What sort of end device do you perceive to display these images? Can you give links to any devices with support for such things? I'm assuming you mean only a still-photo image, not video image. Perhaps you could use a video channel for this and simply

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

Yes, I mean sending image file. On Tue, Aug 25, 2015 at 10:56 AM, Pete Mundy <pete at fiberphone.co.nz> wrote: > Thyda, > > The term 'image' can be quite ambiguous in computing. For example you > could be referring to a firmware image for a phone or you could be > referring to some form of live video channel support. Or something else. > > Can you be more

Did you open a Jira issue for this yet? I can actually work on this this week. On Fri, Oct 16, 2015 at 9:44 AM, George Joseph <george.joseph at fairview5.com> wrote: > On Fri, Oct 16, 2015 at 4:00 AM, Bryant Zimmerman <BryantZ at zktech.com> > wrote: > >> Is there a way to limit the items returned by pjsip show [type] using like >> > > There isn't but

I just get started with it so my question maybe not well catch. Anyway to do the VOIP call and IM we need to use two difference servers? which one is asterisk for VOIP ? and other one for IM that is openfire ? or we can have other choice better than this ? Thank you for your help, I am waiting for your reply. Thyda On Wed, Jul 8, 2015 at 12:43 PM, Kristof Van Den Ouweland < kvandenouweland

I mean by sending image by using sip channel just like we can send text message and what about sending image file ? On Wed, Aug 12, 2015 at 6:37 PM, Joshua Colp <jcolp at digium.com> wrote: > On Sat, Aug 8, 2015, at 07:41 AM, Thyda ENG wrote: > > Dear Sir, > > Kia ora, > > > > > I current have done successfully with sip message over asterisk server , > >

Actually, I am using the openfire and I create two users with the SIP mapping on the openfire to the asterisk server. I can register one user with the openfire client(Spark) and yes it is connect to asterisk SIP also. But with the other one user, I register it with the SIP client(Zoiper/ or Linphone) and then I can make the call over these two SIP but they cannot reach the chat. I wonder what

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

Hi, I was wonder is there any way to custom the message on the call busy or no answer I actually get the error code from asterisk server on busy or no answer. Can I custom the text message or custom the message to sound ? Anyone have any idea could u please share me ? Thank, Thyda -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am currently, I create the VOIP server which enable the user to make the call over the asterisk server, Additionally now I want the user to be able to chat to each other too. I found some suggestion of using the openfire with asterisk but not much said on it, Anyway could you please share me how can I config the IM server over asterisk? I am waiting for your reply, Thyda -------------- next

From: Matthew Jordan <mjordan at digium.com> > > > >> If the INVITE request is not shown in the CLI with 'pjsip set logger > >> on', then Asterisk is not actually receiving the request. > >> > >> Does a pcap show the message being sent to the correct IP/port? If you > >> change the transports to bind to port 5060, does that change

Well, it breaks audio for all NAT endpoints, how can I fix this? > On 18 Mar 2015, at 15:48, Matthew Jordan <mjordan at digium.com> wrote: > > On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome <jleed at me.com <mailto:jleed at me.com>> wrote: >> Hey guys, >> >> have issues with reinvite, no matter what endpoint is calling asterisk >> always tries

NAT endpoint calling local endpount - switching to native_rtp then no audio, both of them have direct_media=no, Verbose log: -- Executing [99 at dialmap:1] AGI("PJSIP/304-00000022", "/pbx/agi.php") in new stack -- Launched AGI Script /pbx/agi.php -- AGI Script Executing Application: (Dial) Options: (PJSIP/99/sip:99 at 192.168.1.73:5060,20) -- Called

MessageSend is command for send message, however I don't know what the context for sending message. I create a pjsip with the context 'from-internal' then when i config the extension for context 'from-internal' it works but then the my call dialplan does not work. Because they both sms and call are coming to the same context 'from-internal', as I notice. I wonder how

Yes, sorry actually in asterisk 13, anyway how could i do that ? On Tue, Sep 22, 2015 at 5:43 PM, Joshua Colp <jcolp at digium.com> wrote: > On 15-09-22 03:34 AM, Thyda ENG wrote: > >> I am using the asterisk 12 with pjsip, I wonder how could I config the >> instance meesseging for pjsqip in asterisk 12 ? What is the default >> message context for pjssip ? I use the

I am using the asterisk 12 with pjsip, I wonder how could I config the instance meesseging for pjsqip in asterisk 12 ? What is the default message context for pjssip ? I use the default extension.conf from the installation and I successfully could make the call over each but when I try to send message, it does not receive by the client. -------------- next part -------------- An HTML attachment

Nick- Are you wanting to recreate the dialplan Echo() application in stasis? Why not just send the call to Echo() instead of Stasis()? On Fri, May 22, 2015 at 11:25 AM, Matthew Jordan <mjordan at digium.com> wrote: > On Fri, May 22, 2015 at 4:41 AM, Nick Awesome <jleed at me.com> wrote: > > Can anyone tell me how can I create echo test using ARI stasis > application?