search for: function_pjsip_header

...ers setted in this case ? > > As documented on the wiki[1] the PJSIP_HEADER dialplan function has to be executed on the PJSIP channel itself, not the calling channel. You need to use a pre-dial handler and invoke it there. > > [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Function_PJSIP_HEADER > Thanks Joshua, that worked. As you see above I want to have the value of headers when call is ended. Problem is that on h extension the channel already gone. Is there a solution to archieve this ? -- Daniel

Hi everyone I’m building an Asterisk 16/PJSIP server and my dialplan uses SIPAddHeader & SIPRemoveHeader but the apps don’t appear to be installed in v16. Can anyone tell me where they went and how to get them installed please? Thanks Mark. Mark Farmer Senior UC Systems Architect Intercity Technology Limited HQ 101-114 Holloway Head, Birmingham, B1 1QP Tel: 0330 332 7933 / 07872542107 /

Hi list, to manage an external queue agent the only solution I found is to connect a local account and redirect calls to this account using forward features from the phone (SNOM). The problem I face is that before calling the agent I would like to set extra header. Dialplan to call external agent is this one with (Gosub): [TOOTAiAudio] ; ; Call our gateway exten =

...de the protocol identifier (Call-ID) in the channel ARI structure[4] which would be in events, or explicitly retrieved[5]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Channels+REST+API#Asterisk20ChannelsRESTAPI-getChannelVar [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_PJSIP_HEADER [3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL [4] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+REST+Data+Models#Asterisk20RESTDataModels-Channel [5] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Channels+REST+API#Asterisk20ChannelsRESTAPI-get -- Jo...

...] instead. For instance the following returned the call-id for me: PJSIP_HEADER(read,Call-ID) How are you attempting to use the channel function [2]? For instance the following returned the call-id for me as well: CHANNEL(pjsip,call-id) [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_PJSIP_HEADER [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_CHANNEL -- Kevin Harwell Senior Software Developer Sangoma Technologies Check us out at: https://sangoma.com & https://asterisk.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://li...

Hello, We are moving from an older asterisk/SIP to a newer (18+) asterisk/PJSIP and trying to figure out how to add [identity] header when originating a call from AMI/PAMI. In the older version we would just set a variable like this: $action = new OriginateAction("SIP/...."); $action->setVariable('__SIPADDHEADER51',"Identity: $identity"); // $identity

Based on postings it should be possible to get the SIP Call-ID header value from the ARI. At what point is this value available ? As well, how do I retrieve that value - something like GET /channels/{channelId}/pjsip_header?key=Call-Id But that doesn't work. -------------- next part -------------- An HTML attachment was scrubbed... URL:

...he protocol identifier (Call-ID) in the channel ARI structure[4] which would be in events, or explicitly retrieved[5]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Channels+REST+API#Asterisk20ChannelsRESTAPI-getChannelVar [2] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_PJSIP_HEADER [3] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL [4] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+REST+Data+Models#Asterisk20RESTDataModels-Channel [5] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Channels+REST+API#Asterisk20ChannelsRESTAPI-get...

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

...erisk 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 chann...