Displaying 20 results from an estimated 10000 matches similar to: "How to send Image over asterisk sip"
2015 Aug 25
2
How to send Image over asterisk sip
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 ,
> >
2015 Aug 25
2
How to send Image over asterisk sip
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
2015 Aug 25
2
How to send Image over asterisk sip
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
2015 Jul 08
3
How to enable IM over the asterisk server
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
2015 Jul 07
2
How to enable IM over the 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
2015 Aug 20
2
Changing volume via dialplan
Greetings everyone,
I am attempting to adjust the volume of a call using Set(VOLUME) in my extensions.conf file. I am finding that Set(VOLUME(TX)=x) and Set(VOLUME(RX)=y) have no discernable effect on my endpoints (Snom 300 IP phones). I have tried setting x and y to -30, -10, -3, -2, -1, 0, 1, 2, 3, 4, 10, and 100 and there appears to be no change on the phone volume. I can see that the
2016 Jul 13
3
PJSIP defaults for endpoints when using realtime
Until Asterisk 11 I could use sip.conf to set defaults for all
phones (language, dtmf, vmexten, etc) and just leave many fields in the
database as NULL. What would be the proper way to do this for Asterisk
13 and PJSIP?
--
Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Carlos Ch?vez
+52 (55)9116-91161
2015 Aug 12
2
Busy level in Asterisk 11
Hi
I need to set the number of incoming calls to one, but the outgoing calls
should be unlimited. I think the busylevel parameter is for it(incoming
calls), but not works. My config is:
cat sip.conf
[general]
[template](!)
qualify=yes
cc_agent_policy=generic
cc_monitor_policy=generic
call-limit=2
busylevel=1
callcounter=yes
subscribecontext = hint
allowsubscribe=yes
[100](template)
2016 May 09
2
voicemail: duration while leaving a message
<!DOCTYPE html>
<html><head>
<meta charset="UTF-8">
</head><body><p>Hello list,</p><p><br></p><p>I am asking when a caller want to leave a message to a mailbox with the application voicemail</p><p>How i can limit the duration for exemple 30 seconds.</p><p>exten =>
2015 Feb 18
2
Asterisk 13 - sorcery realtime for pjsip publish objects
Hello,
I am currently trying to set up pjsip realtime and would like to have
outbound-publish, inbound-publication, and asterisk-publication sorcery
object types in ODBC realtime. Is that currently supported? I know that
some object types are known working and others are not. I was curious
what the status of those objects are.
Thanks!
Matt Hoskins | NPG Corp | Systems Architect
2015 Sep 22
2
How to config instance messaging for asterisk 12
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
2015 Sep 22
2
How to config instance messaging for asterisk 12
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
2014 Dec 10
2
PJSIP configuration question
I'm working with a SIP provider to try and transition our sip connection with them to PJSIP. I thought I had transitioned the settings correctly, but whenever I attempt an Originate it never even tries to send any PJSIP messages.
I'm currently running Asterisk 13.0.0.
Anyone have any suggestions as to what I am doing wrong?
The SIP provider says the latest version of Asterisk they have
2015 Nov 20
2
How to custom the message on call busy or no answer in asterisk
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
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2015 Jul 07
2
How to enable IM over the asterisk server
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
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2015 Nov 17
2
How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?
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,
2015 Sep 22
2
How to set the global setting for each pjsip endpoint
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
2016 Feb 11
3
Unexpected termination of the call when pick up (res_pjsip)
The call initiated from internal extension.
I have made two test call:
Successful call from device on res_pjsip via endpoint on chan_sip:
http://pastebin.com/LWeDYstj
Unsuccessful call from device on res_pjsip via endpoint on res_pjsip:
http://pastebin.com/hepVb6Nu
And ones again i don't see anything that would make asterisk send BYE.
I would be grateful for any ideas.
11.02.2016 1:47,
2014 Dec 10
1
PJSIP configuration question
Thank you for the speedy reply.
My originate string is something like the following where
xxxxx is really the sip provider's supplied IP address
1234567890 is really the phone number I am dialing
PJSIP/outbound.vitelity.net/1234567890
In the chan_sip based solution, it's...
SIP/outbound.vitelity.net/1234567890
Have a great day!
Dan
-----Original Message-----
From:
2015 Nov 17
2
How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?
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