Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] Add SIP Header for 1 SIP peer when calling a group of SIP peers


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
jonas.kellens at telen...
Guest





PostPosted: Thu Nov 14, 2013 11:35 am    Post subject: [asterisk-users] Add SIP Header for 1 SIP peer when calling Reply with quote

Hello,

when calling a group of SIP peers like this :

Dial( "SIP/inno0&SIP/inno4&SIP/inno6,30")

is it possible to have a SIP header added for just 1 of these SIP peers, like only for SIP/inno0 but not for SIP/inno4 and SIP/inno6 ??


I know the function SipAddHeader(), but when I use this in the dialplan before the Dial()-command, then the header is added for all the SIP peers that are being called.


So when calling a group of SIP peers, how can I add an extra SIP header for just one of the SIP peers ?



Kind regards,
Jonas.
Back to top
johnkiniston at gmail.com
Guest





PostPosted: Thu Nov 14, 2013 11:38 am    Post subject: [asterisk-users] Add SIP Header for 1 SIP peer when calling Reply with quote

Use a LOCAL Channel and redirect that one peer through some dialplan


Something like this:


Dial(LOCAL/inno0@addheader&SIP/inno4&SIP/inno6,30)


[addheader]

exten => inno0,1,SipAddHeader(foo)
exten => inno0,n,Dial(SIP/inno0)

exten => inno0,n,Hangup





On Thu, Nov 14, 2013 at 9:35 AM, Jonas Kellens <jonas.kellens@telenet.be (jonas.kellens@telenet.be)> wrote:
Quote:
Hello,

when calling a group of SIP peers like this :

Dial( "SIP/inno0&SIP/inno4&SIP/inno6,30")

is it possible to have a SIP header added for just 1 of these SIP peers, like only for SIP/inno0 but not for SIP/inno4 and SIP/inno6 ??


I know the function SipAddHeader(), but when I use this in the dialplan before the Dial()-command, then the header is added for all the SIP peers that are being called.


So when calling a group of SIP peers, how can I add an extra SIP header for just one of the SIP peers ?



Kind regards,
Jonas.




--
_____________________________________________________________________
-- 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



--
A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects.
---Heinlein
Back to top
barryf-lists at flanag...
Guest





PostPosted: Thu Nov 14, 2013 11:45 am    Post subject: [asterisk-users] Add SIP Header for 1 SIP peer when calling Reply with quote

On 14 November 2013 16:35, Jonas Kellens <jonas.kellens@telenet.be (jonas.kellens@telenet.be)> wrote:

Quote:
Hello,

when calling a group of SIP peers like this :

Dial( "SIP/inno0&SIP/inno4&SIP/inno6,30")

is it possible to have a SIP header added for just 1 of these SIP peers, like only for SIP/inno0 but not for SIP/inno4 and SIP/inno6 ??


I know the function SipAddHeader(), but when I use this in the dialplan before the Dial()-command, then the header is added for all the SIP peers that are being called.


So when calling a group of SIP peers, how can I add an extra SIP header for just one of the SIP peers ?




Hi 


You should be able to do this by using a Local channel for the peer you want to add the header to:


exten => _XXXX,1,Dial(Local/inno0&SIP/inno4&SIP/inno6,30)


exten => inno0,1,SipAddHeader("X-YourHeader")
exten => inno0,2,Dial(SIP/inno0)





Hope this helps.


-Barry Flanagan


Quote:


Kind regards,
Jonas.




--
_____________________________________________________________________
-- 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
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services