FreePBX Version 14.0.3.20
I am trying to set the ringtone on inbound calls to a Sangoma phone by setting the Alert Info field to “(Sangoma) Ring 1”. I can see in the asterisk CLI that the Alert-Info is added to the sip header. When I look at the actual sip invite message, the Alert-Info is not being added to the SIP Header. I have tried using the phone configured as both sip and pjsip extensions. No Luck.
PJSIP/305-000006ca Internal Gosub(func-apply-sipheaders,s,1) start
– Executing [s@func-apply-sipheaders:1] NoOp(“PJSIP/305-000006ca”, “Applying SIP Headers to channel”) in new stack
– Executing [s@func-apply-sipheaders:2] Set(“PJSIP/305-000006ca”, “SIPHEADERKEYS=Alert-Info”) in new stack
– Executing [s@func-apply-sipheaders:3] ExecIf(“PJSIP/305-000006ca”, “0?Set(Rheader=1)”) in new stack
– Executing [s@func-apply-sipheaders:4] While(“PJSIP/305-000006ca”, “1”) in new stack
– Executing [s@func-apply-sipheaders:5] Set(“PJSIP/305-000006ca”, “sipheader=ring1”) in new stack
– Executing [s@func-apply-sipheaders:6] ExecIf(“PJSIP/305-000006ca”, “0?Set(Addheader=1)”) in new stack
– Executing [s@func-apply-sipheaders:7] ExecIf(“PJSIP/305-000006ca”, “0?SIPAddHeader(Alert-Info:ring1)”) in new stack
– Executing [s@func-apply-sipheaders:8] ExecIf(“PJSIP/305-000006ca”, “0?Set(PJSIP_HEADER(add,Alert-Info)=ring1)”) in new stack
– Executing [s@func-apply-sipheaders:9] EndWhile(“PJSIP/305-000006ca”, “”) in new stack
– Executing [s@func-apply-sipheaders:4] While(“PJSIP/305-000006ca”, “0”) in new stack
– Executing [s@func-apply-sipheaders:10] ExecIf(“PJSIP/305-000006ca”, “0?SIPRemoveHeader(Alert-Info:)”) in new stack
– Executing [s@func-apply-sipheaders:11] ExecIf(“PJSIP/305-000006ca”, “0?Set(PJSIP_HEADER(remove,Alert-Info)=)”) in new stack
– Executing [s@func-apply-sipheaders:12] Return(“PJSIP/305-000006ca”, “”) in new stack
== Spawn extension (from-internal, 305, 1) exited non-zero on ‘PJSIP/305-000006ca’
– PJSIP/305-000006ca Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
– Called PJSIP/305/sip:[email protected]:51009
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio TOS bits 184 in TCLASS field.
== Using SIP RTP Audio CoS mark 5
– Connected line update to PJSIP/Bandwidth_B-000006c9 prevented.
– PJSIP/305-000006ca is ringing
– PJSIP/305-000006ca is ringing