VoIP Mailing List Archives

[dan at amtelco.com]

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

[sgriepentrog at digium...]

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@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@amtelco.com (dan@amtelco.com)> wrote:

[dan at amtelco.com]

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@agent in the queue on behalf of the agent (replace number with the agent's extension number)�



From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@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@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@amtelco.com (dan@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



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_____________________________________________________________________
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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

[sgriepentrog at digium...]

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@amtelco.com (dan@amtelco.com)> wrote:

[dan at amtelco.com]

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@lists.digium.com [mailto:asterisk-users-bounces@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@amtelco.com (dan@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@agent in the queue on behalf of the agent (replace number with the agent's extension number)�



From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@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@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@amtelco.com (dan@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





--

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





--

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

[sgriepentrog at digium...]

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@amtelco.com (dan@amtelco.com)> wrote:

[dan at amtelco.com]

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@lists.digium.com [mailto:asterisk-users-bounces@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@amtelco.com (dan@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@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@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@amtelco.com (dan@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@agent in the queue on behalf of the agent (replace number with the agent's extension number)�



From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@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@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@amtelco.com (dan@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





--

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





--

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





--

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

[niklas at tese.se]

Den 2015-08-28 kl. 00:07, skrev Dan Cropp:

[dan at amtelco.com]

Thank you Niklas

That solved my problem.

From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Niklas Larsson
Sent: Friday, August 28, 2015 1:55 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?



Den 2015-08-28 kl. 00:07, skrev Dan Cropp:


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.

You have to use an pre-dial handler option in the Dial command:

same => n,Dial(PJSIP/Agent1,,b(set_header^s^1))

[set_header]
exten => s,1,Set(PJSIP_HEADER(add,X-My-DNID2)=${MY_DNID})
same => n,Return

/niklas

[dan at amtelco.com]

I’m running into a new problem because of the Local channels.

I add Local/Agent1 using QueueAdd.
When the PJSIP Agent dials a feature code, it adds/removes/pauses/unpauses the appropriate EmergencyAgent for the queue.

When the code executes the Queue, I was planning to have that Local section dial plan Dial the appropriate PJSIP EmergencyAgent endpoint (using the pre-dial).

However, the Queue never sees the local channels because they have no device state.
[Aug 28 18:05:06] DEBUG[17868][C-00000004] app_queue.c: TestApp_EmergencyAgent5 is unavailable because his device state is 'invalid'
[Aug 28 18:05:06] DEBUG[17868][C-00000004] app_queue.c: TestApp_EmergencyAgent3 is unavailable because his device state is 'invalid'
[Aug 28 18:05:06] DEBUG[17868][C-00000004] app_queue.c: TestApp_EmergencyAgent2 is unavailable because his device state is 'invalid'
[Aug 28 18:05:06] DEBUG[17868][C-00000004] app_queue.c: TestApp_EmergencyAgent1 is unavailable because his device state is 'invalid'

It seems I may be stuck using chan_sip since it’s Queue behavior does add any SIPADDHEADERs when it executes the INVITE.

From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dan Cropp
Sent: Friday, August 28, 2015 8:20 AM
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?



Thank you Niklas

That solved my problem.

From: asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com) [mailto:asterisk-users-bounces@lists.digium.com (asterisk-users-bounces@lists.digium.com)] On Behalf Of Niklas Larsson
Sent: Friday, August 28, 2015 1:55 AM
To: asterisk-users@lists.digium.com (asterisk-users@lists.digium.com)
Subject: Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?



Den 2015-08-28 kl. 00:07, skrev Dan Cropp:
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.

You have to use an pre-dial handler option in the Dial command:

same => n,Dial(PJSIP/Agent1,,b(set_header^s^1))

[set_header]
exten => s,1,Set(PJSIP_HEADER(add,X-My-DNID2)=${MY_DNID})
same => n,Return

/niklas