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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'>I tried <o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:black'>GET /channels/{channelid}/variable?variable=CHANNEL(pjsip,call-id)<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'>But it responds with<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:black'>"message": "Channel not in Stasis application"<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'>Since I want to get the call-id for a channel not in stasis I guess that won’t work. Similarly, I can’t force the channel through my own code in the dialplan, so the PJSIP_HEADER function won’t work. So it looks like I’ll have to upgrade my Asterisk test system to get the Call-ID from the ARI event. It looks like it was added in Ast 16.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'>Out of curiosity, I see that call-id is returned in the “protocol_id” field of channel data structure. However, since all channels in the same call must have the same Call-ID, how can this data be associated with a channel? Wouldn’t it have to be associated with a bridge? The Call-ID should not be available until two legs are bridged (I think).<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'>Brian<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif;color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><b><span style='font-size:11.0pt;font-family:"Calibri",sans-serif'>From:</span></b><span style='font-size:11.0pt;font-family:"Calibri",sans-serif'> asterisk-users [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Joshua C. Colp<br><b>Sent:</b> Saturday, June 17, 2023 2:25 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><br><b>Subject:</b> Re: [asterisk-users] Get SIP Call-ID from ARI<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><div><div><p class=MsoNormal>On Sat, Jun 17, 2023 at 2:55 PM TTT <<a href="mailto:lists@telium.io">lists@telium.io</a>> wrote:<o:p></o:p></p></div><div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in'><div><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>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<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>GET /channels/{channelId}/pjsip_header?key=Call-Id<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>But that doesn’t work.<o:p></o:p></p></div></div></div></blockquote><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>'pjsip_header' is not a valid route. All possible routes are documented on the wiki, if it's not there then it doesn't exist.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Instead you would use variable[1] to execute the PJSIP_HEADER dialplan function[2] or a better way would be the CHANNEL dialplan function[3] such as:<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>GET /channels/{channelid}/variable?variable=CHANNEL(pjsip,call-id)<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Though I haven't tested that.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Newer versions also include the protocol identifier (Call-ID) in the channel ARI structure[4] which would be in events, or explicitly retrieved[5].<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>[1] <a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Channels+REST+API#Asterisk20ChannelsRESTAPI-getChannelVar">https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Channels+REST+API#Asterisk20ChannelsRESTAPI-getChannelVar</a><o:p></o:p></p></div><div><p class=MsoNormal>[2] <a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_PJSIP_HEADER">https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_PJSIP_HEADER</a><o:p></o:p></p></div><div><p class=MsoNormal>[3] <a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL">https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL</a><o:p></o:p></p></div><div><p class=MsoNormal>[4] <a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+REST+Data+Models#Asterisk20RESTDataModels-Channel">https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+REST+Data+Models#Asterisk20RESTDataModels-Channel</a><o:p></o:p></p></div><div><p class=MsoNormal>[5] <a href="https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Channels+REST+API#Asterisk20ChannelsRESTAPI-get">https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Channels+REST+API#Asterisk20ChannelsRESTAPI-get</a><o:p></o:p></p></div></div><div><p class=MsoNormal><o:p> </o:p></p></div><p class=MsoNormal><span class=gmailsignatureprefix>-- </span><o:p></o:p></p><div><div><div><div><div><div><div><div><div><div><p class=MsoNormal><span style='font-family:"Tahoma",sans-serif;color:#073763'>Joshua C. Colp</span><span style='font-family:"Tahoma",sans-serif'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-family:"Tahoma",sans-serif;color:#073763'>Asterisk Project Lead</span><span style='font-family:"Tahoma",sans-serif'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-family:"Tahoma",sans-serif;color:#073763'>Sangoma Technologies</span><span style='font-family:"Tahoma",sans-serif'><o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-family:"Tahoma",sans-serif;color:#073763'>Check us out at <a href="http://www.sangoma.com" target="_blank">www.sangoma.com</a> and <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a></span><span style='font-family:"Tahoma",sans-serif'><o:p></o:p></span></p></div></div></div></div></div></div></div></div></div></div></div></div></body></html>