ramarajan222
commented
1 month ago Hi @JoseGoncalves ,
Can you make the dialplan change on your systemV17 system and let us know its working with PJSIP or not?
Change details
[macro-dialout-enum]
exten => s,n,Set(TRYDIAL=${REPLACE(TRYDIAL,/,:)}) ; for replace / with :
exten => s,n,Dial(PJSIP/${FROMEXTEN}/${TRYDIAL},${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS}) ; modified to match pjsip syntaxTo make this change on your system, just copy the below dialplan in to extensions_override_freepbx.conf file.
[macro-dialout-enum]
include => macro-dialout-enum-custom
exten => s,1,GosubIf($[$["${ARG3}" != ""] & $["${DB(AMPUSER/${AMPUSER}/pinless)}" != "NOPASSWD"]]?sub-pincheck,s,1())
exten => s,n,GotoIf($["x${OUTDISABLE_${DIAL_TRUNK}}" = "xon"]?disabletrunk,1)
exten => s,n,Set(DIAL_TRUNK_OPTIONS=${IF($["${DB_EXISTS(TRUNK/${DIAL_TRUNK}/dialopts)}" = "1"]?${DB_RESULT}:${TRUNK_OPTIONS})})
exten => s,n,Set(OUTBOUND_GROUP=OUT_${ARG1})
exten => s,n,GotoIf($["${OUTMAXCHANS_${ARG1}}foo" = "foo"]?nomax)
exten => s,n,GotoIf($[ ${GROUP_COUNT(OUT_${ARG1})} >= ${OUTMAXCHANS_${ARG1}} ]?nochans)
exten => s,n(nomax),Set(DIAL_NUMBER=${ARG2})
exten => s,n,Set(DIAL_TRUNK=${ARG1})
exten => s,n,GotoIf($["${INTRACOMPANYROUTE}" = "YES"]?skipoutcid)
exten => s,n,Set(DIAL_TRUNK_OPTIONS=${DIAL_OPTIONS})
exten => s,n,Gosub(macro-outbound-callerid,s,1(${DIAL_TRUNK}))
exten => s,n(skipoutcid),GosubIf($["${PREFIX_TRUNK_${DIAL_TRUNK}}" != ""]?sub-flp-${DIAL_TRUNK},s,1())
exten => s,n,AGI(agi://127.0.0.1/enumlookup.agi)
exten => s,n,ExecIf($["${DB(AMPUSER/${AMPUSER}/cidname)}" != ""]?Set(CONNECTEDLINE(num,i)=${DIAL_NUMBER}))
exten => s,n,ExecIf($["${DB(AMPUSER/${AMPUSER}/cidname)}" != ""]?Set(CONNECTEDLINE(name,i)=CID:${CALLERID(number)}))
exten => s,n(dialloop),GotoIf($["foo${DIALARR}"="foo"]?s-${DIALSTATUS},1)
exten => s,n,ExecIf($["${MOHCLASS}"!="default" & "${MOHCLASS}"!="" & "${FORCE_CONFIRM}"="" ]?Set(DIAL_TRUNK_OPTIONS=M(setmusic^${MOHCLASS})${DIAL_TRUNK_OPTIONS}))
exten => s,n,ExecIf($["${FORCE_CONFIRM}"!="" ]?Set(DIAL_TRUNK_OPTIONS=M(confirm)${DIAL_TRUNK_OPTIONS}))
exten => s,n,Set(TRYDIAL=${CUT(DIALARR,%,1)})
exten => s,n,Set(DIALARR=${CUT(DIALARR,%,2-)})
exten => s,n,Set(TRYDIAL=${REPLACE(TRYDIAL,/,:)})
exten => s,n,Dial(PJSIP/${FROMEXTEN}/${TRYDIAL},${TRUNK_RING_TIMER},${DIAL_TRUNK_OPTIONS})
exten => s,n,GotoIf($[ $[ "${DIALSTATUS}" = "CHANUNAVAIL" ] | $[ "${DIALSTATUS}" = "CONGESTION" ] ]?dialloop)
exten => s,n,GotoIf($["${ARG4}" = "on"]?continue,1:s-${DIALSTATUS},1)
exten => s,n(nochans),Noop(max channels used up)Run dialplan reload in asterisk cli and then test the calls with ENUM trunk.
JoseGoncalves
FreePBX Version
FreePBX 17
Issue Description
I had a FreePBX 15 system were I used an ENUM trunk to connect to a remote institute. After migrated that system to FreePBX 17 with Asterisk 20 and
chan_sipdisabled, I was unable to communicate with that trunk! Looking at Asterisk logs I see this when I try to make a call on that trunk:Only after enabling
chan_sipI was able to restore the calls with it. So it seems ENUM trunks need to be updated to supportchan_pjsip, or else, they are useless with Asterisk 21 or 22, werechan_sipis unavailable.Operating Environment
Debian 12 Asterisk 20.9.3
Relevant log output