Re: [asterisk-users] How to custom the message on call busy or no answer in asterisk

[Thyda ENG]

I found this but I don't know where the busy tone place, I wanna replace
this file, do you have any idea ?
​
 Screen Shot 2015-11-21 at 1.49.47 PM.png
<https://drive.google.com/file/d/0B2n_BStebemaWmtNR1JNSXY4STg/view?usp=drive_web>
​

On Fri, Nov 20, 2015 at 1:51 PM, Julien Sansonnens <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,
> >
> > 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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[asterisk-users] How to custom the message on call busy or no answer in asterisk

[Thyda ENG]

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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Re: [asterisk-users] How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?

[Thyda ENG]

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

context=from-internal

callerid=device <100>

dtmf_mode=rfc4733

use_avpf=no

ice_support=no

media_use_received_transport=no

trust_id_inbound=yes

media_encryption=no

rtp_symmetric=yes

rewrite_contact=yes

*message_context=astsms*



On Tue, Nov 17, 2015 at 8:35 AM, Sonny Rajagopalan <
sonny.rajagopa...@gmail.com> wrote:

> 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
> configuration?
>
> Any help is appreciated.
>
> Thanks,
> Sonny.
>
> --
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Re: [asterisk-users] How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?

[Thyda ENG]

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, Sonny Rajagopalan <
sonny.rajagopa...@gmail.com> wrote:

> 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 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,
>> 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
>>
>> context=from-internal
>>
>> callerid=device <100>
>>
>> dtmf_mode=rfc4733
>>
>> use_avpf=no
>>
>> ice_support=no
>>
>> media_use_received_transport=no
>>
>> trust_id_inbound=yes
>>
>> media_encryption=no
>>
>> rtp_symmetric=yes
>>
>> rewrite_contact=yes
>>
>> *message_context=astsms*
>>
>>
>>
>> On Tue, Nov 17, 2015 at 8:35 AM, Sonny Rajagopalan <
>> sonny.rajagopa...@gmail.com> wrote:
>>
>>> 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
>>> configuration?
>>>
>>> Any help is appreciated.
>>>
>>> Thanks,
>>> Sonny.
>>>
>>> --
>>> _
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>>http://www.asterisk.org/hello
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>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
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>>
>
>
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Re: [asterisk-users] How do I enable instant messaging support for PJSIP endpoints on Asterisk 13.1.0?

[Thyda ENG]

Here is my extension .
[astsms]
exten => _.,1,NoOp(SMS receiving dialplan invoked)
exten => _.,n,NoOp(To ${MESSAGE(to)})
exten => _.,n,NoOp(From ${MESSAGE(from)})
exten => _.,n,NoOp(Body ${MESSAGE(body)})
exten => _.,n,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)})
exten => _.,n,MessageSend(${ACTUALTO},${MESSAGE(from)})
exten => _.,n,NoOp(Send status is ${MESSAGE_SEND_STATUS})
exten => _.,n,GotoIf($["${MESSAGE_SEND_STATUS}" != "SUCCESS"]?sendfailedmsg)
exten => _.,n,Hangup()

exten =>
_.,n(sendfailedmsg),Set(MESSAGE(body)="[${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)}]
Your message to ${EXTEN} has failed. Retry later.")
exten => _.,n,Set(ME_1=${CUT(MESSAGE(from),<,2)})
exten => _.,n,Set(ACTUALFROM=${CUT(ME_1,@,1)})
exten => _.,n,MessageSend(${ACTUALFROM},ServiceCenter)
exten => _.,n,Hangup()
exten => _.,n,Hangup()

On 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 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, Sonny Rajagopalan <
>> sonny.rajagopa...@gmail.com> wrote:
>>
>>> 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 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,
>>>> 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
>>>>
>>>> context=from-internal
>>>>
>>>> callerid=device <100>
>>>>
>>>> dtmf_mode=rfc4733
>>>>
>>>> use_avpf=no
>>>>
>>>> ice_support=no
>>>>
>>>> media_use_received_transport=no
>>>>
>>>> trust_id_inbound=yes
>>>>
>>>> media_encryption=no
>>>>
>>>> rtp_symmetric=yes
>>>>
>>>> rewrite_contact=yes
>>>>
>>>> *message_context=astsms*
>>>>
>>>>
>>>>
>>>> On Tue, Nov 17, 2015 at 8:35 AM, Sonny Rajagopalan <
>>>> sonny.rajagopa...@gmail.com> wrote:
>>>>
>>>>> 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
>>>>> configuration?
>>>>>
>>>>> Any help is appreciated.
>>>>>
>>>>> Thanks,
>>>>> Sonny.
>>>>>
>>>>> --
>>>>> _
>>>>> -- 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
>>>>>
>>>>> asterisk-users mailing list
>>>>> To UNSUBSCRIBE or update options visit:
>>>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>>
>>>>
>>>>
>>>> --
>>>> _

[asterisk-users] Is there any api to create the extension ?

[Thyda ENG]

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.
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[asterisk-users] Is there any api to create the extension ?

[Thyda ENG]

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
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[asterisk-users] Is there any API on Free PBX to connect to the extensions?

[Thyda ENG]

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
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Re: [asterisk-users] Why I get repeat messages many times

[Thyda ENG]

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 :
>
> 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 ${MESSAGE(from)})
> exten => _.,n,NoOp(Body ${MESSAGE(body)})
> exten => _.,n,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)})
> exten => _.,n,MessageSend(${ACTUALTO},${MESSAGE(from)})
> exten => _.,n,NoOp(Send status is ${MESSAGE_SEND_STATUS})
> exten => _.,n,GotoIf($["${MESSAGE_SEND_STATUS}" !=
> "SUCCESS"]?sendfailedmsg)
> exten => _.,n,Hangup()
>
> With this configuration I could send message, but I don't know what wrong
> with it as sometimes I get the repeat messages many times. do you have any
> idea?
>
>
> Are the calls answered before jumping to astsms?
>
> jg
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] Why I get repeat messages many times

[Thyda ENG]

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 ${MESSAGE(from)})
exten => _.,n,NoOp(Body ${MESSAGE(body)})
exten => _.,n,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)})
exten => _.,n,MessageSend(${ACTUALTO},${MESSAGE(from)})
exten => _.,n,NoOp(Send status is ${MESSAGE_SEND_STATUS})
exten => _.,n,GotoIf($["${MESSAGE_SEND_STATUS}" != "SUCCESS"]?sendfailedmsg)
exten => _.,n,Hangup()

With this configuration I could send message, but I don't know what wrong
with it as sometimes I get the repeat messages many times. do you have any
idea?
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Re: [asterisk-users] Sending XML over the asterisk PJSIP

[Thyda ENG]

I am pretty new with asterisk and actually, I want to send the image to the
client and my process is that, first the image is uploaded to the server
and once the image uploaded it will return the xml tag that contain the
information about that image, Then the sip send that xml to the server,
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...@gmail.com> wrote:
> > Can i send XML data over the asterisk PJSIP ?
> >
>
> That's a fairly generic question. Can you be more specific about what
> you are trying to accomplish?
>
> --
> Matthew Jordan
> Digium, Inc. | Director of Technology
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
> --
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[asterisk-users] Sending XML over the asterisk PJSIP

[Thyda ENG]

Can i send XML data over the asterisk PJSIP ?
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[asterisk-users] Why I get repeat messages many times

[Thyda ENG]

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 ${MESSAGE(from)})
exten => _.,n,NoOp(Body ${MESSAGE(body)})
exten => _.,n,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)})
exten => _.,n,MessageSend(${ACTUALTO},${MESSAGE(from)})
exten => _.,n,NoOp(Send status is ${MESSAGE_SEND_STATUS})
exten => _.,n,GotoIf($["${MESSAGE_SEND_STATUS}" != "SUCCESS"]?sendfailedmsg)
exten => _.,n,Hangup()

With this configuration I could send message, but I don't know what wrong
with it as sometimes I get the repeat messages many times. do you have any
idea?
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[asterisk-users] How to config instance messaging for asterisk 12

[Thyda ENG]

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.
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Re: [asterisk-users] How to set the global setting for each pjsip endpoint

[Thyda ENG]

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 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 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
>>>
>>> media_use_received_transport=no
>>>
>>> trust_id_inbound=yes
>>>
>>> send_pai=yes
>>>
>>> rtp_symmetric=yes
>>>
>>> rewrite_contact=yes
>>>
>>> message_context=astsms
>>>
>>>
>>> [200]
>>>
>>> type=endpoint
>>>
>>> aors=200
>>>
>>> auth=200-auth
>>>
>>> allow=ulaw,alaw,gsm,g726
>>>
>>> context=from-internal
>>>
>>> callerid=device <200>
>>>
>>> dtmf_mode=rfc4733
>>>
>>> use_avpf=no
>>>
>>> ice_support=no
>>>
>>> media_use_received_transport=no
>>>
>>> trust_id_inbound=yes
>>>
>>> send_pai=yes
>>>
>>> rtp_symmetric=yes
>>>
>>> rewrite_contact=yes
>>>
>>> message_context=astsms
>>>
>>>
>>> how could I avoid duplicate thing like this ?
>>>
>>> --
>>>
>>>
>> From my brief look at pjsip.conf it uses the same template concept as the
>> sip.conf.
>>
>> Here's the relevant instructions from the sip.conf in asteris13
>>
>>  ;
>> ; Because you might have a large number of similar sections, it is
>> generally
>> ; convenient to use templates for the common parameters, and add them
>> ; the the various sections. Examples are below, and we can even leave
>> ; the templates uncommented as they will not harm:
>>
>> [basic-options](!); a template
>> dtmfmode=rfc2833
>> context=from-office
>> type=friend
>>
>> [natted-phone](!,basic-options)   ; another template inheriting
>> basic-options
>> directmedia=no
>> host=dynamic
>>
>> [public-phone](!,basic-options)   ; another template inheriting
>> basic-options
>> directmedia=yes
>>
>> [my-codecs](!); a template for my preferred codecs
>> disallow=all
>> allow=ilbc
>> allow=g729
>> allow=gsm
>> allow=g723
>> allow=ulaw
>> ; Or, more simply:
>> ;allow=!all,ilbc,g729,gsm,g723,ulaw
>>
>> [ulaw-phone](!)   ; and another one for ulaw-only
>> disallow=all
>> allow=ulaw
>> ; Again, more simply:
>> ;allow=!all,ulaw
>>
>> ; and finally instantiate a few phones
>> ;
>> ; [2133](natted-phone,my-codecs)
>> ;secret = peekaboo
>> ; [2134](natted-phone,ulaw-phone)
>> ;secret = not_very_secret
>> ; [2136](public-phone,ulaw-phone)
>> ;secret = not_very_secret_either
>> ; ...
>> ;
>>
>> Regards
>>
>> Ish
>> --
>>
>> Ishfaq Malik
>> Department: VOIP Support
>> Company: Packnet Limited
>> t: +44 (0)161 660 2350
>> f: +44 (0)161 660 9825
>> e: i...@pack-net.co.uk
>> w: http://www.pack-net.co.uk
>>
>> Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
>> 37 Ducie Street
>> Manchester, M1 2JW
>> COMPANY REG NO. 04920552
>>
>>
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>
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Re: [asterisk-users] How to config instance messaging for asterisk 12

[Thyda ENG]

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
>> 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.
>>
>
> The context can be configured (at least in 13) using the message_context
> option on endpoints but will fall back to the value of the context option
> otherwise.
>
> Just an additional reminder that 12 is in security fix only[1] and has not
> received bug fixes for quite some time. It will also go end of life at the
> end of this year.
>
> [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
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[asterisk-users] How to set the global setting for each pjsip endpoint

[Thyda ENG]

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

media_use_received_transport=no

trust_id_inbound=yes

send_pai=yes

rtp_symmetric=yes

rewrite_contact=yes

message_context=astsms


[200]

type=endpoint

aors=200

auth=200-auth

allow=ulaw,alaw,gsm,g726

context=from-internal

callerid=device <200>

dtmf_mode=rfc4733

use_avpf=no

ice_support=no

media_use_received_transport=no

trust_id_inbound=yes

send_pai=yes

rtp_symmetric=yes

rewrite_contact=yes

message_context=astsms


how could I avoid duplicate thing like this ?
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Re: [asterisk-users] How to config instance messaging for asterisk 12

[Thyda ENG]

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 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, anyway how could i do that ?
>>
>
> I've told you how the context is determined in my response. Otherwise the
> MESSAGE dialplan function and MessageSend dialplan application can be used
> to retrieve message information and send a message out of dialog.
>
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
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Re: [asterisk-users] How to config instance messaging for asterisk 12

[Thyda ENG]

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 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 could i custom the context
>> for the messaging ?
>>
>
> As I originally mentioned the "message_context" option can be used to send
> messages to a different context.
>
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
>
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Re: [asterisk-users] How to set the global setting for each pjsip endpoint

[Thyda ENG]

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 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
>>
>> media_use_received_transport=no
>>
>> trust_id_inbound=yes
>>
>> send_pai=yes
>>
>> rtp_symmetric=yes
>>
>> rewrite_contact=yes
>>
>> message_context=astsms
>>
>>
>> [200]
>>
>> type=endpoint
>>
>> aors=200
>>
>> auth=200-auth
>>
>> allow=ulaw,alaw,gsm,g726
>>
>> context=from-internal
>>
>> callerid=device <200>
>>
>> dtmf_mode=rfc4733
>>
>> use_avpf=no
>>
>> ice_support=no
>>
>> media_use_received_transport=no
>>
>> trust_id_inbound=yes
>>
>> send_pai=yes
>>
>> rtp_symmetric=yes
>>
>> rewrite_contact=yes
>>
>> message_context=astsms
>>
>>
>> how could I avoid duplicate thing like this ?
>>
>> --
>>
>>
> From my brief look at pjsip.conf it uses the same template concept as the
> sip.conf.
>
> Here's the relevant instructions from the sip.conf in asteris13
>
>  ;
> ; Because you might have a large number of similar sections, it is
> generally
> ; convenient to use templates for the common parameters, and add them
> ; the the various sections. Examples are below, and we can even leave
> ; the templates uncommented as they will not harm:
>
> [basic-options](!); a template
> dtmfmode=rfc2833
> context=from-office
> type=friend
>
> [natted-phone](!,basic-options)   ; another template inheriting
> basic-options
> directmedia=no
> host=dynamic
>
> [public-phone](!,basic-options)   ; another template inheriting
> basic-options
> directmedia=yes
>
> [my-codecs](!); a template for my preferred codecs
> disallow=all
> allow=ilbc
> allow=g729
> allow=gsm
> allow=g723
> allow=ulaw
> ; Or, more simply:
> ;allow=!all,ilbc,g729,gsm,g723,ulaw
>
> [ulaw-phone](!)   ; and another one for ulaw-only
> disallow=all
> allow=ulaw
> ; Again, more simply:
> ;allow=!all,ulaw
>
> ; and finally instantiate a few phones
> ;
> ; [2133](natted-phone,my-codecs)
> ;secret = peekaboo
> ; [2134](natted-phone,ulaw-phone)
> ;secret = not_very_secret
> ; [2136](public-phone,ulaw-phone)
> ;secret = not_very_secret_either
> ; ...
> ;
>
> Regards
>
> Ish
> --
>
> Ishfaq Malik
> Department: VOIP Support
> Company: Packnet Limited
> t: +44 (0)161 660 2350
> f: +44 (0)161 660 9825
> e: i...@pack-net.co.uk
> w: http://www.pack-net.co.uk
>
> Registered Address: PACKNET LIMITED, Duplex 2, Ducie House
> 37 Ducie Street
> Manchester, M1 2JW
> COMPANY REG NO. 04920552
>
>
> --
> _
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[asterisk-users] How to enable SIP text messaging with PJSIP ?

[Thyda ENG]

Hi sir ,

How to enable SIP text messaging with PJSIP ?
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[asterisk-users] Cannot hear the sound from the sips on called

[Thyda ENG]

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 ?

Thank you, I am waiting for your reply .
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[asterisk-users] Cannot register sip on asterisk running on amazon

[Thyda ENG]

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 give me any clue ?
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Re: [asterisk-users] How to integrate Asterisk with XMPP

[Thyda ENG]

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 integrating XMPP? Shared states among multiple
> extensions? Passing messages between extensions? Depending on what you want
> and what infrastructure you have in place will all influence the answer.
>
> Also, you will get better responses if you say what you have tried and
> what isn't working or say what you goal is and ask for pointer on how to
> get there. Depending on what you want to do, there are multiple tutorials
> available online, but I will say that I did find it was a bit of trial and
> error to get xmpp working in my organization. I use it for allowing
> extensions on remote sites to join in to some of our call queues, thus
> needing our (multiple) asterisk boxes to be able to share extension states
> with each other. It wasn't the easiest thing in the world to get working on
> the 11 series.
>
> Depending on what you want to do, the new pubsub features in PJSIP in
> Asterisk 13 series may do what you want. I know I am looking forward to
> investigating them and quite possibly getting rid of my xmpp setup.
>
>
> https://wiki.asterisk.org/wiki/display/AST/Exchanging+Device+and+Mailbox+State+Using+PJSIP
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[asterisk-users] How to integrate Asterisk with XMPP

[Thyda ENG]

How to integrate Asterisk with XMPP ?
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Re: [asterisk-users] How to send Image over asterisk sip

[Thyda ENG]

I am pretty with asterisk, and i would thank for your reply. You mention
that i could send a base 64 image with asterisk, so how could i config
asterisk to support that?

On Tue, Aug 25, 2015 at 7:39 PM, Jeppe Larsen j...@krugercorp.dk wrote:

 If you are just abusing asterisk as a proxy for text/images, i guess you
 could send a base64 encoded image (as a SIP MESSAGE), and decode it on the
 other end? A weird thing to do really, but i dont know your needs.

 - Jeppe
 Den 25/08/15 kl. 06.23 skrev Pete Mundy:


 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 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 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. 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 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 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 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?

 What do you mean by image?

 --
 Joshua Colp
 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 Check us out at: www.digium.com  www.asterisk.org

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[asterisk-users] Does the asterisk support for sending image ?

[Thyda ENG]

Does the asterisk support for sending image ? if it supports how to config
it ?
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Re: [asterisk-users] How to send Image over asterisk sip

[Thyda ENG]

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 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?

 What do you mean by image?

 --
 Joshua Colp
 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 Check us out at: www.digium.com  www.asterisk.org

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Re: [asterisk-users] How to send Image over asterisk sip

[Thyda ENG]

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. 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 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 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 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?

 What do you mean by image?

 --
 Joshua Colp
 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 Check us out at: www.digium.com  www.asterisk.org

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Re: [asterisk-users] How to send Image over asterisk sip

[Thyda ENG]

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. 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 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 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 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?

 What do you mean by image?

 --
 Joshua Colp
 Digium, Inc. | Senior Software Developer
 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
 Check us out at: www.digium.com  www.asterisk.org

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[asterisk-users] How to send Image over asterisk sip

[Thyda ENG]

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
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[asterisk-users] How to enable group call

[Thyda ENG]

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,
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[asterisk-users] Does the asterisk support instance messaging ?

[Thyda ENG]

Dear Sir,

Does the asterisk support instance messaging ? Thank,

Thyda
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[asterisk-users] Asterisk SMS

[Thyda ENG]

Dear Sir,

Does the asterisk support SMS feature ?
If it does how can we config that ?
I am waiting for your reply,Thank.

Thyda
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Re: [asterisk-users] How to enable IM over the asterisk server

[Thyda ENG]

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 
kvandenouwel...@vangenechten.com wrote:

 Hi Thyda,

 I think you should see these as two individual systems. (I'm not an expert
 so just thinking out loud).

 Since you mention that you did a SIP mapping on Openfire, may I assume
 that you have the Asterisk IM plugin?

 In case of yes:
 Yes, there is a plugin between OpenFire and Asterisk but it is not
 actively developed anymore since 2006
 http://www.igniterealtime.org/projects/asterisk/

 So I don't think the plugin is really realiable anymore  on current
 versions.

 --

 I consider them as 2 separate systems which have to work on their own.
 Unfortunatly this means that every softphone has 2 accounts: one is SIP to
 Asterisk, one is XMPP to Openfire.

 That way our users are able to call internal/external using Asterisk, but
 do IM and internal calling via Openfire. (They can choose which source they
 take)

 Openfire is connected to our AD so our users just can logon with their
 Windows credentials.

 Unfortunatly, if you want a real production connection between Asterisk
 and Openfire, I'm unable to assist since I don't have the knowledge 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 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 should I config between openfire and asterisk
 to enable chat over these two sip clients ?
 I am waiting for your reply, Thank.

 Thyda

 On Tue, Jul 7, 2015 at 3:17 PM, Kristof Van Den Ouweland 
 kvandenouwel...@vangenechten.com wrote:

 Good morning Thyda;

 Perhaps somebody has a solution for using it on Asterisk itself but after
 some trying I added the Openfire server as a IM server.

 I was a bit afraid that 'if' I got it working properly we had to maintain
 it and off course had to troubleshoot it in case it didn't work anymore.

 I've read something that you add a ams_msg context in extensions.conf but
 that didn't work for me unfortunaly. It did work for SIP Messages on phones
 but not for IM.

 I found Openfire easier to configure and it added a full integration with
 our LDAP which 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 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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[asterisk-users] How to handle multiple lines call

[Thyda ENG]

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 this?

Example:
we have 6 sips
-sip 1 is calling to sip 2
-sip 3 is calling to sip 4
-sip 5 is calling to sip 6

Here is my configuration,
exten = _.,1,Dial(SIP/${EXTEN})

I am waiting for your reply, Thank.

Thyda
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Re: [asterisk-users] How to enable IM over the asterisk server

[Thyda ENG]

Yes, I have though of setting them up on the same server(openfire, and
asterisk) and the problem come in mind that how can register the user to
openfire automatically when I register the user SIP on the asterisk server
? Do you have any idea? I am waiting for your reply.

Thank,

Thyda

On Wed, Jul 8, 2015 at 6:55 PM, James Cass jcas...@gmail.com wrote:

 You can have the openfire server installed on the same server as asterisk
 without any issue, just size your server appropriately.  Just keep in mind
 they are different services.

 James Cass http://goog_987864563
 jcas...@gmail.com


 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 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 
 kvandenouwel...@vangenechten.com wrote:

 Hi Thyda,

 I think you should see these as two individual systems. (I'm not an
 expert so just thinking out loud).

 Since you mention that you did a SIP mapping on Openfire, may I assume
 that you have the Asterisk IM plugin?

 In case of yes:
 Yes, there is a plugin between OpenFire and Asterisk but it is not
 actively developed anymore since 2006
 http://www.igniterealtime.org/projects/asterisk/

 So I don't think the plugin is really realiable anymore  on current
 versions.

 --

 I consider them as 2 separate systems which have to work on their own.
 Unfortunatly this means that every softphone has 2 accounts: one is SIP to
 Asterisk, one is XMPP to Openfire.

 That way our users are able to call internal/external using Asterisk,
 but do IM and internal calling via Openfire. (They can choose which source
 they take)

 Openfire is connected to our AD so our users just can logon with their
 Windows credentials.

 Unfortunatly, if you want a real production connection between Asterisk
 and Openfire, I'm unable to assist since I don't have the knowledge 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 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 should I config between openfire and asterisk
 to enable chat over these two sip clients ?
 I am waiting for your reply, Thank.

 Thyda

 On Tue, Jul 7, 2015 at 3:17 PM, Kristof Van Den Ouweland 
 kvandenouwel...@vangenechten.com wrote:

 Good morning Thyda;

 Perhaps somebody has a solution for using it on Asterisk itself but
 after some trying I added the Openfire server as a IM server.

 I was a bit afraid that 'if' I got it working properly we had to
 maintain it and off course had to troubleshoot it in case it didn't work
 anymore.

 I've read something that you add a ams_msg context in extensions.conf
 but that didn't work for me unfortunaly. It did work for SIP Messages on
 phones but not for IM.

 I found Openfire easier to configure and it added a full integration
 with our LDAP which 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 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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 Privileged Confidential Information may be contained in this message.
 If you are not the addressee indicated in this message (or responsible for
 delivery of the message to such person), you may not copy or deliver this
 message to anyone.
 In such case, you should destroy this message and kindly notify the
 sender by reply email.
 Please advise immediately if you or your employer does not consent to
 Internet email for messages of this kind.
 Opinions, conclusions and other information in this message that do not
 relate to the official business of my firm shall be understood as neither
 given nor endorsed by it.

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 -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] How to enable IM over the asterisk server

[Thyda ENG]

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 should I config between
openfire and asterisk to enable chat over these two sip clients ?
I am waiting for your reply, Thank.

Thyda

On Tue, Jul 7, 2015 at 3:17 PM, Kristof Van Den Ouweland 
kvandenouwel...@vangenechten.com wrote:

 Good morning Thyda;

 Perhaps somebody has a solution for using it on Asterisk itself but after
 some trying I added the Openfire server as a IM server.

 I was a bit afraid that 'if' I got it working properly we had to maintain
 it and off course had to troubleshoot it in case it didn't work anymore.

 I've read something that you add a ams_msg context in extensions.conf but
 that didn't work for me unfortunaly. It did work for SIP Messages on phones
 but not for IM.

 I found Openfire easier to configure and it added a full integration with
 our LDAP which 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 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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 In such case, you should destroy this message and kindly notify the sender
 by reply email.
 Please advise immediately if you or your employer does not consent to
 Internet email for messages of this kind.
 Opinions, conclusions and other information in this message that do not
 relate to the official business of my firm shall be understood as neither
 given nor endorsed by it.

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[asterisk-users] How to enable IM over the asterisk server

[Thyda ENG]

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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