s <jul...@jsansonnens.ch>
wrote:
> Hi,
> Check the DIALSTATUS variable.
> http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS
>
> Regards, Julien
>
> --
>
>
>
> 2015-11-20 2:15 GMT+01:00 Thyda ENG <ength...@gmail.com>:
> > Hi,
> >
>
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
--
The default message context for the pjsip is the same the call context, so
to set the new message context for the pjsqip you need to modify your
pjsip.endpoint_custom.conf and add the message context as in the example
below :
[100]
type=endpoint
aors=100
auth=100-auth
allow=ulaw,alaw,gsm,g726
I turn on pjsip set logger on) but
> they are not delivered to the other endpoint. What gives?
>
> Any help appreciated. Thanks!
>
> On Mon, Nov 16, 2015 at 9:16 PM, Thyda ENG <ength...@gmail.com> wrote:
>
>> The default message context for the pjsip is the same the call context
Tue, Nov 17, 2015 at 9:58 AM, Sonny Rajagopalan <
sonny.rajagopa...@gmail.com> wrote:
> Thanks again. How do you create that message context in extensions.conf?
>
> On Mon, Nov 16, 2015 at 9:44 PM, Thyda ENG <ength...@gmail.com> wrote:
>
>> According to what I h
Hi,
I found an api to get all the extensions from asterisk however I wonder is
there any api to create the extension on asterisk or not ?
Thank you, I am waiting for your reply.
--
_
-- Bandwidth and Colocation Provided by
Hi,
I found an api to get all the extensions from asterisk however I wonder is
there any api to create the extension on asterisk or not ?
Thank you, I am waiting for your reply.
Thyda Eng
--
_
-- Bandwidth and Colocation
Hi,
I wonder about free pbx does it has any api to get the register extensions
or the api to create the extensions or not ?
thank you, I am waiting for your reply.
Thyda Eng
--
_
-- Bandwidth and Colocation Provided by http
No, It directly goes the context astsms when we send the message. but it
still repeats the message sometimes.
On Mon, Oct 19, 2015 at 3:25 PM, jg wrote:
>
> I am using the asterisk 13 and I config my dialplan for the SIP messaging
> as the following :
>
>
I am using the asterisk 13 and I config my dialplan for the SIP messaging
as the following :
http://highsecurity.blogspot.com/2012/03/asterisk-10-110-sms-messaging-or-sip.html
[astsms]
exten => _.,1,NoOp(SMS receiving dialplan invoked)
exten => _.,n,NoOp(To ${MESSAGE(to)})
exten => _.,n,NoOp(From
,
however I don't see any notify information on the server at all. I wonder
do we need to config anything on the server to enable it accept the xml
text ?
On Sat, Oct 17, 2015 at 11:24 PM, Matthew Jordan <mjor...@digium.com> wrote:
> On Sat, Oct 17, 2015 at 2:32 AM, Thyda ENG <ength...@gmai
Can i send XML data over the asterisk PJSIP ?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
I am using the asterisk 13 and I config my dialplan for the SIP messaging
as the following :
http://highsecurity.blogspot.com/2012/03/asterisk-10-110-sms-messaging-or-sip.html
[astsms]
exten => _.,1,NoOp(SMS receiving dialplan invoked)
exten => _.,n,NoOp(To ${MESSAGE(to)})
exten => _.,n,NoOp(From
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
My pjsip.conf is the auto_generated file from freepbx and it should not be
modified. I really cannot find where to set the messge_context in freepbx
UI at all. could you please show me where?
On Tue, Sep 22, 2015 at 10:22 PM, Thyda ENG <ength...@gmail.com> wrote:
> how if I use the auto
Yes, sorry actually in asterisk 13, anyway how could i do that ?
On Tue, Sep 22, 2015 at 5:43 PM, Joshua Colp <jc...@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
>> instan
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
are coming to the same context
'from-internal', as I notice. I wonder how could i custom the context for
the messaging ?
On Tue, Sep 22, 2015 at 8:52 PM, Joshua Colp <jc...@digium.com> wrote:
> On 15-09-22 10:48 AM, Thyda ENG wrote:
>
>> Yes, sorry actually in asterisk 13, any
I heard some talk about message_context too but I don't know to where to
put the message_context info in .
On Tue, Sep 22, 2015 at 9:02 PM, Joshua Colp <jc...@digium.com> wrote:
> On 15-09-22 10:57 AM, Thyda ENG wrote:
>
>> MessageSend is command for send message, however
how if I use the auto generate once from freepbx ?
On Tue, Sep 22, 2015 at 10:12 PM, Ishfaq Malik <i...@pack-net.co.uk> wrote:
>
>
> On 22 September 2015 at 16:04, Thyda ENG <ength...@gmail.com> wrote:
>
>> I have many endpoints and each endpoint has some para
Hi sir ,
How to enable SIP text messaging with PJSIP ?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Dear, sir
I have installed the Freepbx 12 on amazon could and it and asterisk run
successfully. I could registered the sips but I wonder why when we make the
call between those sip, each sip cannot hear the sound talking from each
side ? could you please tell me what I need to config more ?
I have installed Freepbx successfully on the Amazon Ec2 micro instance I
finally could access to the Freepbx and it show the state success. I create
the extension on this instance then I wonder why when i try to register my
sip client to this instance it seems like no any action.
Could you please
I wanna pass the message between the sip.
On Tue, Sep 1, 2015 at 10:49 PM, Kevin Larsen <
kevin.lar...@pioneerballoon.com> wrote:
> >
> > How to integrate Asterisk with XMPP ?
> >
>
> What you are asking for isn't a simple question to answer. What exactly do
> you want to accomplish by
How to integrate Asterisk with XMPP ?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
for this and simply display only a still
image instead? I do believe that Asterisk has video support, although I
haven't personally used it.
Hope this helps.
Pete
On 25/08/2015, at 4:11 PM, Thyda ENG ength...@gmail.com wrote:
I mean by sending the .jpg, or .png or . file.
On Tue, Aug 25, 2015
Does the asterisk support for sending image ? if it supports how to config
it ?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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 jc...@digium.com wrote:
On Sat, Aug 8, 2015, at 07:41 AM, Thyda ENG wrote:
Dear Sir,
Kia ora,
I current have done
I mean by sending the .jpg, or .png or . file.
On Tue, Aug 25, 2015 at 11:10 AM, Thyda ENG ength...@gmail.com wrote:
Yes, I mean sending image file.
On Tue, Aug 25, 2015 at 10:56 AM, Pete Mundy p...@fiberphone.co.nz
wrote:
Thyda,
The term 'image' can be quite ambiguous in computing
channel support. Or something else.
Can you be more explicit as to exactly what you mean by 'image file'
and/or what it is that you aim to achieve or what you want to see happen?
Pete
On 25/08/2015, at 3:47 PM, Thyda ENG ength...@gmail.com wrote:
I mean by sending image by using sip channel
Dear Sir,
I current have done successfully with sip message over asterisk server ,
and additionally now I want to send the image between sip using asterisk.
Could any one share me how to config the asterisk for sending image from
sip?
Thank, I am waiting for your reply.
Thyda
--
Dear Sir,
I would like to see how can we config the asterisk to enable calling to
multiple SIP number at the same time?
Thank,
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
Dear Sir,
Does the asterisk support instance messaging ? Thank,
Thyda
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Dear Sir,
Does the asterisk support SMS feature ?
If it does how can we config that ?
I am waiting for your reply,Thank.
Thyda
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
of it.
sorry
Hope this helps a bit.
kristof
Thyda ENG ength...@gmail.com 7/07/2015 11:28
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
Hi,
I am new to asterisk, I have set up the asterisk server and successfully I
could make the dialplan between 2 SIPs but when there are more than two
sips calling each other, my dialplan seems doing the wrong routing to the
sip. Do i need to config anything additionally to asterisk to handle
On Wed, Jul 8, 2015 at 4:24 AM, Thyda ENG ength...@gmail.com wrote:
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
allowed single sign so that users could use the same
password and log on automatically with the Jitsi client.
But if you have some specific questions, I will be glad to answer.
//Kristof
Thyda ENG ength...@gmail.com 7/07/2015 6:07
I am currently, I create the VOIP server which enable
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
39 matches
Mail list logo