search for: psjip

I have an Asterisk 13 that only processes app Transfer with PJSIP, to the tune of 60 per second. No voice calls. After like 2 hours, I can no longer get into Asterisk. This command, asterisk -r, fails, and also "asterisk -rx core show channels", etc. I am returned to the bash prompt. I checked the handles and lsof | grep asterisk |wc -l 7098126 I think there is a kind of handle leak

I am trying to set add a SIP Header to a call before adding it to the Queue. The dial plan sends the call to my macro to perform the work. When I use chan_sip, everything works as expected. When I use PSJIP support, it's not adding the SIP header. Looking at the output, I see the macro is called in both cases. In the PJSIP case, the added sip header never is showing up in the asterisk logs (verbose 999). In the SIP case, I see it. Does the function Set(PJSIP_HEADER(add, ..... not transfer over...

...as you can see another side is very old cisco router, so I can't change anything there. I don't see any examples here https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip#Migratingfromchan_siptores_pjsip-ExampleSIPTrunkConfiguration on insecure=port,invite? psjip? equivalent. Thank you!