search for: originatecallid

...er, this is never being passed through the PJSIP INVITE header Action: Originate ActionID: S598 Channel: PJSIP/133 at 1002 Exten: createcall Context: MyContext Priority: 1 Timeout: 60000 CallerID: CustomerName <########## > Variable: CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=5,OriginateCallId=396 Async: true Is there a setting that's required on the PJSIP endpoint to allow overwriting the INVITE packet's Contact header? Is there something else I am missing to perform this? Have a great day! Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: &l...

...Dan Cropp<291> Here is an example of how we do this with AMI successfully. Action: Originate ActionID: S40 Channel: PJSIP/1003 at 1003 Exten: createcall Context: IS Priority: 1 Timeout: 60000 CallerID: Dan Cropp <291> Variable: CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2 Async: true Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200807/a56aec37/attachment.html>

...ion: Originate > > ActionID: S40 > > Channel: PJSIP/1003 at 1003 > > Exten: createcall > > Context: IS > > Priority: 1 > > Timeout: 60000 > > CallerID: Dan Cropp <291> > > Variable: > CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2 > > Async: true > > > > Dan > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/...

...ion: Originate > > ActionID: S40 > > Channel: PJSIP/1003 at 1003 > > Exten: createcall > > Context: IS > > Priority: 1 > > Timeout: 60000 > > CallerID: Dan Cropp <291> > > Variable: > CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2 > > Async: true > > > > Dan > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk...

...Dan Cropp<291> Here is an example of how we do this with AMI successfully. Action: Originate ActionID: S40 Channel: PJSIP/1003 at 1003 Exten: createcall Context: IS Priority: 1 Timeout: 60000 CallerID: Dan Cropp <291> Variable: CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2 Async: true Dan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20200807/b2ed1087/attachment.html>

...d, len 715 If I answer before the timeout, it connects to the dialplan, answers, plays, and hangs up as I expected. Am I sending something wrong? Action: Originate ActionID: 100 Channel: SIP/1000 Exten: 1 Context: createcall Priority: 1 Timeout: 3 CallerID: SIP/1000 Variable: OriginateCallId=100 Async: true Is there a configuration setting I am missing? I've tried calling a Linksys SIP phone and I've also tried it with PhonerLite SIP Client, both are doing the same thing. Have a great day! -------------- next part -------------- An HTML attachment was scrubbed......

...Dan Cropp<291> Here is an example of how we do this with AMI successfully. Action: Originate ActionID: S40 Channel: PJSIP/1003 at 1003 Exten: createcall Context: IS Priority: 1 Timeout: 60000 CallerID: Dan Cropp <291> Variable: CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2 Async: true Dan -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk...

...nnel: PJSIP/1003 at 1003 >> >> Exten: createcall >> >> Context: IS >> >> Priority: 1 >> >> Timeout: 60000 >> >> CallerID: Dan Cropp <291> >> >> Variable: >> CALLERID(num-pres)=allowed_passed_screen,TrunkAllocateId=1,OriginateCallId=2 >> >> Async: true >> >> >> >> Dan >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk com...