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Dan Cropp

2015-Aug-27 21:54 UTC

[asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?

Thanks Scott.

I was able to get the basic concept to run.
However, it seems PJSIP INVITE for the Dial also does not support added headers.

The Local channel dial plan did have the channel variable values.  I added them
as SIP headers, then Dial(PJSIP/Agent).
The INVITE for the Dial on PJSIP continues to not include the SIP Headers I
added.

For chan_sip, I have no problem with this.  Even the original Queue code I had
includes the added SIP headers with it?s INVITE to the Agent.


From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Scott Griepentrog
Sent: Thursday, August 27, 2015 4:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior
to calling Queue and have it part of the INVITE packet?

Local channels:
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeeperDialplan_id324598.html

This explains adding members to queues, although it doesn't specifically
provide an example of using local channels in a queue:
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id289508.html

Basically, read that book, and if you get stuck ask for help.


On Thu, Aug 27, 2015 at 3:08 PM, Dan Cropp <dan at amtelco.com<mailto:dan
at amtelco.com>> wrote:
Thanks Scott.

I?m taking over for someone else?s code, so I must admit I?m still learning the
Agent and Queue concepts.  Local channels are something I have not used either. 
Would local channels essentially be an internal bridge?

How would I
?Register Local/number at agent in the queue on behalf of the agent (replace
number with the agent's extension number)?



From: asterisk-users-bounces at
lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>
[mailto:asterisk-users-bounces at
lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>] On
Behalf Of Scott Griepentrog
Sent: Thursday, August 27, 2015 1:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior
to calling Queue and have it part of the INVITE packet?

To add a header to the call leg that goes to the agent, try using a local
channel to activate dialplan on the outbound call:

Register Local/number at agent in the queue on behalf of the agent (replace
number with the agent's extension number)

In dialplan [agent], wild card match the number, add the header, and then
Dial(PJSIP/{$EXTEN}) to send the call to the agent.


On Thu, Aug 27, 2015 at 1:40 PM, Dan Cropp <dan at amtelco.com<mailto:dan
at amtelco.com>> wrote:
I have a call coming in.
I need to add a SIP Header to the channel.
Then, I need to send the call to the Queue so it is sent to the Agent.

The SIP header I added, I need to have appear in the INVITE sent to the Agent.

It works in chan_sip.  I send the call to a macro which does?
n,SIPAddHeader(My-Header-Name:${MY-HEADER-VALUE})
n,Queue(${ARG2})


In PJSIP , this doesn?t seem to work.  Is there any way to add custom PJSIP
headers to be sent as part of the INVITE to the Agent?
When I look at the code, it seems as though the INVITE doesn?t look for any
custom headers to be included with the INVITE packet.  Is this correct?

Have a great day!
Dan

--
_____________________________________________________________________
-- 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



--
[Digium logo]
Scott Griepentrog
Digium, Inc ? Software Developer
445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US
direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090
Check us out at: http://digium.com ? http://asterisk.org

--
_____________________________________________________________________
-- 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



--
[Digium logo]
Scott Griepentrog
Digium, Inc ? Software Developer
445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US
direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090
Check us out at: http://digium.com ? http://asterisk.org
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Scott Griepentrog

2015-Aug-27 21:57 UTC

head link

[asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?

Are you using this method of setting headers on PJSIP?

https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_PJSIP_HEADER


On Thu, Aug 27, 2015 at 4:54 PM, Dan Cropp <dan at amtelco.com> wrote:
> Thanks Scott.
>
>
>
> I was able to get the basic concept to run.
>
> However, it seems PJSIP INVITE for the Dial also does not support added
> headers.
>
>
>
> The Local channel dial plan did have the channel variable values.  I added
> them as SIP headers, then Dial(PJSIP/Agent).
>
> The INVITE for the Dial on PJSIP continues to not include the SIP Headers
> I added.
>
>
>
> For chan_sip, I have no problem with this.  Even the original Queue code I
> had includes the added SIP headers with it?s INVITE to the Agent.
>
>
>
>
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Scott
Griepentrog
> *Sent:* Thursday, August 27, 2015 4:28 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Is it possible to perform PJSIP Add
> Header prior to calling Queue and have it part of the INVITE packet?
>
>
>
> Local channels:
>
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeeperDialplan_id324598.html
>
>
>
> This explains adding members to queues, although it doesn't
specifically
> provide an example of using local channels in a queue:
>
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id289508.html
>
>
>
> Basically, read that book, and if you get stuck ask for help.
>
>
>
>
>
> On Thu, Aug 27, 2015 at 3:08 PM, Dan Cropp <dan at amtelco.com>
wrote:
>
> Thanks Scott.
>
>
>
> I?m taking over for someone else?s code, so I must admit I?m still
> learning the Agent and Queue concepts.  Local channels are something I have
> not used either.  Would local channels essentially be an internal bridge?
>
>
>
> How would I
>
> ?Register Local/number at agent in the queue on behalf of the agent
(replace
> number with the agent's extension number)?
>
>
>
>
>
>
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Scott
Griepentrog
> *Sent:* Thursday, August 27, 2015 1:57 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Is it possible to perform PJSIP Add
> Header prior to calling Queue and have it part of the INVITE packet?
>
>
>
> To add a header to the call leg that goes to the agent, try using a local
> channel to activate dialplan on the outbound call:
>
>
>
> Register Local/number at agent in the queue on behalf of the agent (replace
> number with the agent's extension number)
>
>
>
> In dialplan [agent], wild card match the number, add the header, and then
> Dial(PJSIP/{$EXTEN}) to send the call to the agent.
>
>
>
>
>
> On Thu, Aug 27, 2015 at 1:40 PM, Dan Cropp <dan at amtelco.com>
wrote:
>
> I have a call coming in.
>
> I need to add a SIP Header to the channel.
>
> Then, I need to send the call to the Queue so it is sent to the Agent.
>
>
>
> The SIP header I added, I need to have appear in the INVITE sent to the
> Agent.
>
>
>
> It works in chan_sip.  I send the call to a macro which does?
>
> n,SIPAddHeader(My-Header-Name:${MY-HEADER-VALUE})
>
> n,Queue(${ARG2})
>
>
>
>
>
> In PJSIP , this doesn?t seem to work.  Is there any way to add custom
> PJSIP headers to be sent as part of the INVITE to the Agent?
>
> When I look at the code, it seems as though the INVITE doesn?t look for
> any custom headers to be included with the INVITE packet.  Is this correct?
>
>
>
> Have a great day!
>
> Dan
>
>
> --
> _____________________________________________________________________
> -- 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
>
>
>
>
>
> --
>
> [image: Digium logo]
>
> Scott Griepentrog
> Digium, Inc ? Software Developer
> 445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US
> direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090
> Check us out at: http://digium.com ? http://asterisk.org
>
>
> --
> _____________________________________________________________________
> -- 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
>
>
>
>
>
> --
>
> [image: Digium logo]
>
> Scott Griepentrog
> Digium, Inc ? Software Developer
> 445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US
> direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090
> Check us out at: http://digium.com ? http://asterisk.org
>
> --
> _____________________________________________________________________
> -- 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
>


-- 
[image: Digium logo]
Scott Griepentrog
Digium, Inc ? Software Developer
445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US
direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090
Check us out at: http://digium.com ? http://asterisk.org
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Dan Cropp

2015-Aug-27 22:07 UTC

head link

[asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?

I have both the PJSIP add and the chan_sip way of adding SIP headers in there. 
The Verbose is showing the variable value is there.
The INVITE to PJSIP/Agent1 does not include either X-My-DNID or X-My-DNID2
headers.

exten => 1234,1,Verbose(X-My-DNID:${MY_DNID})
same => n,Set(X-My-DNID=${MY_DNID})
same => n,Set(PJSIP_HEADER(add,X-My-DNID2)=${MY_DNID})
same => n,Dial(PJSIP/Agent1)


From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of Scott Griepentrog
Sent: Thursday, August 27, 2015 4:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior
to calling Queue and have it part of the INVITE packet?

Are you using this method of setting headers on PJSIP?

https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_PJSIP_HEADER


On Thu, Aug 27, 2015 at 4:54 PM, Dan Cropp <dan at amtelco.com<mailto:dan
at amtelco.com>> wrote:
Thanks Scott.

I was able to get the basic concept to run.
However, it seems PJSIP INVITE for the Dial also does not support added headers.

The Local channel dial plan did have the channel variable values.  I added them
as SIP headers, then Dial(PJSIP/Agent).
The INVITE for the Dial on PJSIP continues to not include the SIP Headers I
added.

For chan_sip, I have no problem with this.  Even the original Queue code I had
includes the added SIP headers with it?s INVITE to the Agent.


From: asterisk-users-bounces at
lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>
[mailto:asterisk-users-bounces at
lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>] On
Behalf Of Scott Griepentrog
Sent: Thursday, August 27, 2015 4:28 PM

To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior
to calling Queue and have it part of the INVITE packet?

Local channels:
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeeperDialplan_id324598.html

This explains adding members to queues, although it doesn't specifically
provide an example of using local channels in a queue:
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id289508.html

Basically, read that book, and if you get stuck ask for help.


On Thu, Aug 27, 2015 at 3:08 PM, Dan Cropp <dan at amtelco.com<mailto:dan
at amtelco.com>> wrote:
Thanks Scott.

I?m taking over for someone else?s code, so I must admit I?m still learning the
Agent and Queue concepts.  Local channels are something I have not used either. 
Would local channels essentially be an internal bridge?

How would I
?Register Local/number at agent in the queue on behalf of the agent (replace
number with the agent's extension number)?



From: asterisk-users-bounces at
lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>
[mailto:asterisk-users-bounces at
lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>] On
Behalf Of Scott Griepentrog
Sent: Thursday, August 27, 2015 1:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior
to calling Queue and have it part of the INVITE packet?

To add a header to the call leg that goes to the agent, try using a local
channel to activate dialplan on the outbound call:

Register Local/number at agent in the queue on behalf of the agent (replace
number with the agent's extension number)

In dialplan [agent], wild card match the number, add the header, and then
Dial(PJSIP/{$EXTEN}) to send the call to the agent.


On Thu, Aug 27, 2015 at 1:40 PM, Dan Cropp <dan at amtelco.com<mailto:dan
at amtelco.com>> wrote:
I have a call coming in.
I need to add a SIP Header to the channel.
Then, I need to send the call to the Queue so it is sent to the Agent.

The SIP header I added, I need to have appear in the INVITE sent to the Agent.

It works in chan_sip.  I send the call to a macro which does?
n,SIPAddHeader(My-Header-Name:${MY-HEADER-VALUE})
n,Queue(${ARG2})


In PJSIP , this doesn?t seem to work.  Is there any way to add custom PJSIP
headers to be sent as part of the INVITE to the Agent?
When I look at the code, it seems as though the INVITE doesn?t look for any
custom headers to be included with the INVITE packet.  Is this correct?

Have a great day!
Dan

--
_____________________________________________________________________
-- 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



--
[Digium logo]
Scott Griepentrog
Digium, Inc ? Software Developer
445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US
direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090
Check us out at: http://digium.com ? http://asterisk.org

--
_____________________________________________________________________
-- 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



--
[Digium logo]
Scott Griepentrog
Digium, Inc ? Software Developer
445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US
direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090
Check us out at: http://digium.com ? http://asterisk.org

--
_____________________________________________________________________
-- 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



--
[Digium logo]
Scott Griepentrog
Digium, Inc ? Software Developer
445 Jan Davis Drive NW ? Huntsville, AL 35806 ? US
direct/fax: +1 256 428 6239 ? mobile: +1 256 580 6090
Check us out at: http://digium.com ? http://asterisk.org
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