Well, we had it working on a previous outdated version of freepbx, so i am wondering why this one does not work the way that the old one did.
Dial Pattern Problems
Internal call answered externaly
Hello again,
regarding to my problem I logged what asterisk does when a call from the FritzBox arrives,
seems that eyerything is working as expected until it comes to ending the call, maybe one of you can see what is the problem (WAN-IP changed ;-)) the last BYE request is repeatet 5 times:
thanks in advance
best regards
marco
<--- Received SIP request (1084 bytes) from UDP:192.168.189.1:5060 --->
INVITE sip:test@raspbx.fritz.box SIP/2.0
Via: SIP/2.0/UDP 78.83.22.287:61877;rport;branch=z9hG4bKD070C3BB80E2C5FA
From: <sip:anonymous@fritz.box>;tag=BE86F76F88149A82
To: <sip:test@raspbx.fritz.box>
Call-ID: 60903BA57D463210@78.83.22.287
CSeq: 190 INVITE
Contact: <sip:anonymous@78.83.22.287:61877;uniq=D61805C4A256BE2CFFF00B2AF766D>
Max-Forwards: 70
Expires: 120
User-Agent: AVM FRITZ!Box 7490 113.07.12 (Jul 3 2019)
Supported: 100rel,replaces
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,ME SSAGE,PUBLISH
Content-Type: application/sdp
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 359
v=0
o=user 766034 766034 IN IP4 78.83.22.287
s=call
c=IN IP4 78.83.22.287
t=0 0
m=audio 61878 RTP/AVP 8 0 2 102 100 99 97 101
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:61879
== Setting global variable 'SIPDOMAIN' to 'raspbx.fritz.box'
<--- Transmitting SIP response (327 bytes) to UDP:192.168.189.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 78.83.22.287:61877;rport=5060;received=192.168.189.1;branch=z9 hG4bKD070C3BB80E2C5FA
Call-ID: 60903BA57D463210@78.83.22.287
From: <sip:anonymous@fritz.box>;tag=BE86F76F88149A82
To: <sip:test@raspbx.fritz.box>
CSeq: 190 INVITE
Server: FPBX-14.0.5.9(13.27.0)
Content-Length: 0
-- Executing [test@from-sip-external:1] NoOp("PJSIP/anonymous-0000002b", "Received incoming SIP connection from unknown peer to test") in new stack
-- Executing [test@from-sip-external:2] Set("PJSIP/anonymous-0000002b", "DID=test") in new stack
-- Executing [test@from-sip-external:3] Goto("PJSIP/anonymous-0000002b", "s,1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing [s@from-sip-external:1] GotoIf("PJSIP/anonymous-0000002b", "1?s etlanguage:checkanon") in new stack
-- Goto (from-sip-external,s,2)
-- Executing [s@from-sip-external:2] Set("PJSIP/anonymous-0000002b", "CHANNE L(language)=en") in new stack
-- Executing [s@from-sip-external:3] GotoIf("PJSIP/anonymous-0000002b", "0?n oanonymous") in new stack
-- Executing [s@from-sip-external:4] Goto("PJSIP/anonymous-0000002b", "from- trunk,test,1") in new stack
-- Goto (from-trunk,test,1)
-- Executing [test@from-trunk:1] Goto("PJSIP/anonymous-0000002b", "from- internal,754,1") in new stack
-- Goto (from-internal,754,1)
-- Executing [754@from-internal:1] GotoIf("PJSIP/anonymous-0000002b", "0?ext -local,754,1:followme-check,754,1") in new stack
-- Goto (followme-check,754,1)
-- Executing [754@followme-check:1] Gosub("PJSIP/anonymous-0000002b", "follo wme-sub,754,1()") in new stack
-- Executing [754@followme-sub:1] Macro("PJSIP/anonymous-0000002b", "user-ca llerid,") in new stack
-- Executing [s@macro-user-callerid:1] Set("PJSIP/anonymous-0000002b", "TOUC H_MONITOR=1567074031.43") in new stack
-- Executing [s@macro-user-callerid:2] Set("PJSIP/anonymous-0000002b", "AMPU SER=anonymous") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("PJSIP/anonymous-0000002b", "0 ?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("PJSIP/anonymous-0000002b", "1 ?Set(REALCALLERIDNUM=anonymous)") in new stack
-- Executing [s@macro-user-callerid:5] Set("PJSIP/anonymous-0000002b", "AMPU SER=") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("PJSIP/anonymous-0000002b", "0 ?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("PJSIP/anonymous-0000002b", "AMPU SERCIDNAME=") in new stack
-- Executing [s@macro-user-callerid:8] ExecIf("PJSIP/anonymous-0000002b", "0 ?Set(__CIDMASQUERADING=TRUE)") in new stack
-- Executing [s@macro-user-callerid:9] GotoIf("PJSIP/anonymous-0000002b", "1 ?report") in new stack
-- Goto (macro-user-callerid,s,15)
-- Executing [s@macro-user-callerid:15] NoOp("PJSIP/anonymous-0000002b", "Ma cro Depth is 1") in new stack
-- Executing [s@macro-user-callerid:16] GotoIf("PJSIP/anonymous-0000002b", " 1?report2:macroerror") in new stack
-- Goto (macro-user-callerid,s,17)
-- Executing [s@macro-user-callerid:17] GotoIf("PJSIP/anonymous-0000002b", " 0?continue") in new stack
-- Executing [s@macro-user-callerid:18] Set("PJSIP/anonymous-0000002b", "__T TL=64") in new stack
-- Executing [s@macro-user-callerid:19] GotoIf("PJSIP/anonymous-0000002b", " 1?continue") in new stack
-- Goto (macro-user-callerid,s,35)
-- Executing [s@macro-user-callerid:35] Set("PJSIP/anonymous-0000002b", "CAL LERID(number)=anonymous") in new stack
-- Executing [s@macro-user-callerid:36] Set("PJSIP/anonymous-0000002b", "CAL LERID(name)=") in new stack
-- Executing [s@macro-user-callerid:37] GotoIf("PJSIP/anonymous-0000002b", " 1?cnum") in new stack
-- Goto (macro-user-callerid,s,39)
-- Executing [s@macro-user-callerid:39] Set("PJSIP/anonymous-0000002b", "CDR (cnum)=anonymous") in new stack
-- Executing [s@macro-user-callerid:40] Set("PJSIP/anonymous-0000002b", "CHA NNEL(language)=en") in new stack
-- Executing [754@followme-sub:2] Set("PJSIP/anonymous-0000002b", "DIAL_OPTI ONS=TtrI") in new stack
-- Executing [754@followme-sub:3] Set("PJSIP/anonymous-0000002b", "CONNECTED LINE(num,i)=754") in new stack
-- Executing [754@followme-sub:4] Gosub("PJSIP/anonymous-0000002b", "sub-pre sencestate-display,s,1(754)") in new stack
-- Executing [s@sub-presencestate-display:1] Goto("PJSIP/anonymous-0000002b" , "state-not_set,1") in new stack
-- Goto (sub-presencestate-display,state-not_set,1)
-- Executing [state-not_set@sub-presencestate-display:1] Set("PJSIP/anonymou s-0000002b", "PRESENCESTATE_DISPLAY=") in new stack
-- Executing [state-not_set@sub-presencestate-display:2] Return("PJSIP/anony mous-0000002b", "") in new stack
-- Executing [754@followme-sub:5] Set("PJSIP/anonymous-0000002b", "CONNECTED LINE(name)=test") in new stack
-- Executing [754@followme-sub:6] Set("PJSIP/anonymous-0000002b", "FM_DIALST ATUS=UNAVAILABLE") in new stack
-- Executing [754@followme-sub:7] Set("PJSIP/anonymous-0000002b", "__EXTTOCA LL=754") in new stack
-- Executing [754@followme-sub:8] Set("PJSIP/anonymous-0000002b", "__PICKUPM ARK=754") in new stack
-- Executing [754@followme-sub:9] Macro("PJSIP/anonymous-0000002b", "blkvm-s etifempty,") in new stack
-- Executing [s@macro-blkvm-setifempty:1] GotoIf("PJSIP/anonymous-0000002b", "1?init") in new stack
-- Goto (macro-blkvm-setifempty,s,4)
-- Executing [s@macro-blkvm-setifempty:4] Set("PJSIP/anonymous-0000002b", "_ _BLKVM_CHANNEL=PJSIP/anonymous-0000002b") in new stack
-- Executing [s@macro-blkvm-setifempty:5] Set("PJSIP/anonymous-0000002b", "S HARED(BLKVM,PJSIP/anonymous-0000002b)=TRUE") in new stack
-- Executing [s@macro-blkvm-setifempty:6] Set("PJSIP/anonymous-0000002b", "G OSUB_RETVAL=TRUE") in new stack
-- Executing [s@macro-blkvm-setifempty:7] MacroExit("PJSIP/anonymous-0000002 b", "") in new stack
-- Executing [754@followme-sub:10] GotoIf("PJSIP/anonymous-0000002b", "1?ski pov") in new stack
-- Goto (followme-sub,754,13)
-- Executing [754@followme-sub:13] Set("PJSIP/anonymous-0000002b", "RRNODEST =") in new stack
-- Executing [754@followme-sub:14] Set("PJSIP/anonymous-0000002b", "__NODEST =754") in new stack
-- Executing [754@followme-sub:15] GosubIf("PJSIP/anonymous-0000002b", "0?su b-fmsetcid,s,1()") in new stack
-- Executing [754@followme-sub:16] GotoIf("PJSIP/anonymous-0000002b", "1?ski pprepend") in new stack
-- Goto (followme-sub,754,18)
-- Executing [754@followme-sub:18] Set("PJSIP/anonymous-0000002b", "RecordMe thod=Group") in new stack
-- Executing [754@followme-sub:19] Gosub("PJSIP/anonymous-0000002b", "sub-re cord-check,s,1(exten,754,)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("PJSIP/anonymous-0000002b", "0?in itialized") in new stack
-- Executing [s@sub-record-check:2] Set("PJSIP/anonymous-0000002b", "__REC_S TATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:3] Set("PJSIP/anonymous-0000002b", "NOW=156 7074031") in new stack
-- Executing [s@sub-record-check:4] Set("PJSIP/anonymous-0000002b", "__DAY=2 9") in new stack
-- Executing [s@sub-record-check:5] Set("PJSIP/anonymous-0000002b", "__MONTH =08") in new stack
-- Executing [s@sub-record-check:6] Set("PJSIP/anonymous-0000002b", "__YEAR= 2019") in new stack
-- Executing [s@sub-record-check:7] Set("PJSIP/anonymous-0000002b", "__TIMES TR=20190829-102031") in new stack
-- Executing [s@sub-record-check:8] Set("PJSIP/anonymous-0000002b", "__FROME XTEN=anonymous") in new stack
-- Executing [s@sub-record-check:9] Set("PJSIP/anonymous-0000002b", "__MON_F MT=wav") in new stack
-- Executing [s@sub-record-check:10] NoOp("PJSIP/anonymous-0000002b", "Recor dings initialized") in new stack
-- Executing [s@sub-record-check:11] ExecIf("PJSIP/anonymous-0000002b", "1?S et(ARG3=dontcare)") in new stack
-- Executing [s@sub-record-check:12] Set("PJSIP/anonymous-0000002b", "REC_PO LICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:13] ExecIf("PJSIP/anonymous-0000002b", "0?S et(REC_STATUS=NO)") in new stack
-- Executing [exten@sub-record-check:14] ExecIf("PJSIP/anonymous-0000002b", "1?Set(RECMODE=dontcare)") in new stack
-- Executing [exten@sub-record-check:15] ExecIf("PJSIP/anonymous-0000002b", "1?Set(RECMODE=dontcare)") in new stack
-- Executing [exten@sub-record-check:16] Gosub("PJSIP/anonymous-0000002b", " recordcheck,1(dontcare,internal,754)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/anonymous-0000002b ", "Starting recording check against dontcare") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("PJSIP/anonymous-0000002b ", "dontcare") in new stack
-- Goto (sub-record-check,recordcheck,3)
-- Executing [recordcheck@sub-record-check:3] Return("PJSIP/anonymous-000000 2b", "") in new stack
-- Executing [exten@sub-record-check:17] Return("PJSIP/anonymous-0000002b", "") in new stack
-- Executing [754@followme-sub:20] GotoIf("PJSIP/anonymous-0000002b", "1?ski pdring") in new stack
-- Goto (followme-sub,754,23)
-- Executing [754@followme-sub:23] Set("PJSIP/anonymous-0000002b", "STRATEGY =ringallv2-prim") in new stack
-- Executing [754@followme-sub:24] Set("PJSIP/anonymous-0000002b", "__RVOL=" ) in new stack
-- Executing [754@followme-sub:25] GotoIf("PJSIP/anonymous-0000002b", "1?ski psimple") in new stack
-- Goto (followme-sub,754,28)
-- Executing [754@followme-sub:28] Set("PJSIP/anonymous-0000002b", "RingGrou pMethod=ringallv2-prim") in new stack
-- Executing [754@followme-sub:29] Set("PJSIP/anonymous-0000002b", "_FMGRP=7 54") in new stack
-- Executing [754@followme-sub:30] GotoIf("PJSIP/anonymous-0000002b", "1?DIA LGRP") in new stack
-- Goto (followme-sub,754,34)
-- Executing [754@followme-sub:34] ExecIf("PJSIP/anonymous-0000002b", "1?Set (DOPTS=TtrI):Set(DOPTS=m(Ring)TtI)") in new stack
-- Executing [754@followme-sub:35] Set("PJSIP/anonymous-0000002b", "__ALT_CO NFIRM_MSG=") in new stack
-- Executing [754@followme-sub:36] GotoIf("PJSIP/anonymous-0000002b", "0?doc onfirm") in new stack
-- Executing [754@followme-sub:37] GotoIf("PJSIP/anonymous-0000002b", "1?rin gallv21") in new stack
-- Goto (followme-sub,754,40)
-- Executing [754@followme-sub:40] Macro("PJSIP/anonymous-0000002b", "dial,3 ,TtrI,754") in new stack
-- Executing [s@macro-dial:1] NoOp("PJSIP/anonymous-0000002b", "Blind Transf er: , Attended Transfer: , User: , Alert Info: ") in new stack
-- Executing [s@macro-dial:2] ExecIf("PJSIP/anonymous-0000002b", "0?Set(ALER T_INFO=)") in new stack
-- Executing [s@macro-dial:3] ExecIf("PJSIP/anonymous-0000002b", "0?Set(ALER T_INFO=)") in new stack
-- Executing [s@macro-dial:4] ExecIf("PJSIP/anonymous-0000002b", "0?Set(ALER T_INFO=)") in new stack
-- Executing [s@macro-dial:5] ExecIf("PJSIP/anonymous-0000002b", "0?Set(CHAN NEL(musicclass)=)") in new stack
-- Executing [s@macro-dial:6] AGI("PJSIP/anonymous-0000002b", "dialparties.a gi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
dialparties.agi: Caller ID name is 'unknown' number is 'anonymous'
dialparties.agi: CW Ignore is:
dialparties.agi: CF Ignore is:
dialparties.agi: CW IN_USE/BUSY is: 1
> dialparties.agi: USE_CONFIRMATION: 'FALSE'
> dialparties.agi: RINGGROUP_INDEX: ''
dialparties.agi: Methodology of ring is 'ringallv2-prim'
-- dialparties.agi: Added extension 754 to extension map
> dialparties.agi: got fmgrp_prering: 2, fmgrp_grptime: 2
> dialparties.agi: fmgrp_totalprering: 4
> dialparties.agi: found extension in pre-ring and array
> dialparties.agi: ringallv2 ring times: REALPRERING: 4, PRERING: 2
-- dialparties.agi: Extension 754 cf is disabled
-- dialparties.agi: Extension 754 do not disturb is disabled
> dialparties.agi: extnum 754 has: cw: 1; hascfb: 0 [] hascfu: 0 []
dialparties.agi: EXTENSION_STATE: 4 (UNAVAILABLE)
dialparties.agi: Extension 754 has ExtensionState: 4
== dialparties.agi: Discovered PJSIP Endpoint PJSIP/754
-- dialparties.agi: Ended up with no PJSIP contacts
-- dialparties.agi: DbDel CALLTRACE/754 - Caller ID is not defined
-- dialparties.agi: Filtered ARG3: 754
-- dialparties.agi: RING ALL V2 :
> dialparties.agi: NODEST: 754 adding M(auto-blkvm) to dialopts: TtrIM(au to-blkvm)
> dialparties.agi: NODEST: 754 blkvm enabled macro already in dialopts: T trIM(auto-blkvm)
dialparties.agi: Setting default NOANSWER DIALSTATUS since no extensions availa ble
dialparties.agi: RVOL_MODE ''
dialparties.agi: RVOL is:
dialparties.agi: RVOLPARENT is:
-- <PJSIP/anonymous-0000002b>AGI Script dialparties.agi completed, returning 0
-- Executing [s@macro-dial:7] NoOp("PJSIP/anonymous-0000002b", "Returned fro m dialparties with no extensions to call and DIALSTATUS: NOANSWER") in new stack
-- Executing [s@macro-dial:8] MacroExit("PJSIP/anonymous-0000002b", "") in n ew stack
-- Executing [754@followme-sub:41] Goto("PJSIP/anonymous-0000002b", "nextste p") in new stack
-- Goto (followme-sub,754,46)
-- Executing [754@followme-sub:46] Set("PJSIP/anonymous-0000002b", "RingGrou pMethod=") in new stack
-- Executing [754@followme-sub:47] GotoIf("PJSIP/anonymous-0000002b", "0?nod est") in new stack
-- Executing [754@followme-sub:48] Set("PJSIP/anonymous-0000002b", "__NODEST =") in new stack
-- Executing [754@followme-sub:49] Set("PJSIP/anonymous-0000002b", "__PICKUP MARK=") in new stack
-- Executing [754@followme-sub:50] Macro("PJSIP/anonymous-0000002b", "blkvm- clr,") in new stack
-- Executing [s@macro-blkvm-clr:1] Set("PJSIP/anonymous-0000002b", "SHARED(B LKVM,PJSIP/anonymous-0000002b)=") in new stack
-- Executing [s@macro-blkvm-clr:2] Set("PJSIP/anonymous-0000002b", "GOSUB_RE TVAL=") in new stack
-- Executing [s@macro-blkvm-clr:3] MacroExit("PJSIP/anonymous-0000002b", "") in new stack
-- Executing [754@followme-sub:51] Set("PJSIP/anonymous-0000002b", "DIALSTAT US=CHANUNAVAIL") in new stack
-- Executing [754@followme-sub:52] GotoIf("PJSIP/anonymous-0000002b", "0?doh angup") in new stack
-- Executing [754@followme-sub:53] Goto("PJSIP/anonymous-0000002b", "app-ann ouncement-1,s,1") in new stack
-- Goto (app-announcement-1,s,1)
-- Executing [s@app-announcement-1:1] GotoIf("PJSIP/anonymous-0000002b", "0? begin") in new stack
-- Executing [s@app-announcement-1:2] Answer("PJSIP/anonymous-0000002b", "") in new stack
> 0x73f04470 -- Strict RTP learning after remote address set to: 78.83.22.287 2.187:61878
<--- Transmitting SIP response (903 bytes) to UDP:192.168.189.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 78.83.22.287:61877;rport=5060;received=192.168.189.1;branch=z9 hG4bKD070C3BB80E2C5FA
Call-ID: 60903BA57D463210@78.83.22.287
From: <sip:anonymous@fritz.box>;tag=BE86F76F88149A82
To: <sip:test@raspbx.fritz.box>;tag=0fa20040-638c-4020-a37d-cd92c8c8f26b
CSeq: 190 INVITE
Server: FPBX-14.0.5.9(13.27.0)
Contact: <sip:192.168.189.39:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PR ACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 310
v=0
o=- 766034 766036 IN IP4 192.168.189.39
s=Asterisk
c=IN IP4 192.168.189.39
t=0 0
m=audio 14526 RTP/AVP 0 8 2 97 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<--- Received SIP request (463 bytes) from UDP:192.168.189.1:5060 --->
ACK sip:192.168.189.39:5060 SIP/2.0
Via: SIP/2.0/UDP 78.83.22.287:5060;branch=z9hG4bKA99DE06B7A356731
From: <sip:anonymous@fritz.box>;tag=BE86F76F88149A82
To: <sip:test@raspbx.fritz.box>;tag=0fa20040-638c-4020-a37d-cd92c8c8f26b
Call-ID: 60903BA57D463210@78.83.22.287
CSeq: 190 ACK
Contact: <sip:anonymous@78.83.22.287;uniq=D61805C4A256BE2CFFF00B2AF766D>
Max-Forwards: 70
User-Agent: AVM FRITZ!Box 7490 113.07.12 (Jul 3 2019)
Content-Length: 0
> 0x73f04470 -- Strict RTP qualifying stream type: audio
> 0x73f04470 -- Strict RTP switching source address to 192.168.189.1:7078
-- Executing [s@app-announcement-1:3] Wait("PJSIP/anonymous-0000002b", "1") in new stack
-- Executing [s@app-announcement-1:4] NoOp("PJSIP/anonymous-0000002b", "Play ing announcement test") in new stack
-- Executing [s@app-announcement-1:5] Playback("PJSIP/anonymous-0000002b", " ,noanswer") in new stack
-- Executing [s@app-announcement-1:6] Goto("PJSIP/anonymous-0000002b", "app- blackhole,hangup,1") in new stack
-- Goto (app-blackhole,hangup,1)
-- Executing [hangup@app-blackhole:1] NoOp("PJSIP/anonymous-0000002b", "Blac khole Dest: Hangup") in new stack
-- Executing [hangup@app-blackhole:2] Hangup("PJSIP/anonymous-0000002b", "") in new stack
== Spawn extension (app-blackhole, hangup, 2) exited non-zero on 'PJSIP/anonym ous-0000002b'
<--- Transmitting SIP request (467 bytes) to UDP:78.83.22.287:61877 --->
BYE sip:anonymous@78.83.22.287:61877;uniq=D61805C4A256BE2CFFF00B2AF766D SIP/2.0
Via: SIP/2.0/UDP 192.168.189.39:5060;rport;branch=z9hG4bKPjfd3d195d-84a0-47d8-be 10-0e12006e0f22
From: <sip:test@raspbx.fritz.box>;tag=0fa20040-638c-4020-a37d-cd92c8c8f26b
To: <sip:anonymous@fritz.box>;tag=BE86F76F88149A82
Call-ID: 60903BA57D463210@78.83.22.287
CSeq: 335 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: FPBX-14.0.5.9(13.27.0)
Content-Length: 0
>
Internal call answered externaly
The “rewrite_contact” option needs to be set to “yes” in order to have requests sent to the source IP address/port instead of what the remote side says (which in this case is a public IP address and port). I don’t know how that is done in FreePBX, but that would send the BYE to the right IP address/port.
Internal call answered externaly
This is probably due to the fritz box having some type of SIP NAT helper / ALG
Internal call answered externaly
Hi jcolp, thanks again for your reply, unfortunately this option is allready set for this extension,
but this option help text states adress-port is it the ip, too?
best regards
marco
Internal call answered externaly
it states “source IP address-port”. Did you reload after applying such a thing? Did you confirm it is matching the appropriate entry? It seems to be coming in as anonymous, so if the option is not applied to anonymous it would have no impact.
PJSIP update external IP address automatically?
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Serious Development Question
I know people keep pointing this out like it makes a difference. Since SNG7 (v14) the OS has be derived of RHEL. Distro 6 (v13) and below are CentOS derived. It doesn’t matter that CentOS 8 is coming out, the distro is not derived from it.
It’s currently RHEL7.5 derived.
Serious Development Question
I gave them the benefit of the doubt. RHEL 8 was released on May 7, 2019 and still ships with PHP 7.2
Internal call answered externaly
ah ok, sounds good.
since i’m very new to this voip stuff, could you tell me where one could set this?
is this an otion for the extension? how could this option be named?
thank you very much
best regards
marco
Custom dialplan
do a macro like
[dialout-trunk-POSTcall-hook] exist?
I want to do custom dialplan when a call end
Trunk callerID provider (dialplan)
Can someone tell me more about providers and CallerID ?
I tried to do
exten => s,1,Set(CALLERID(num)=+337XXXXXX)
but the number is still the same.
I don’t know how this work.
Serious Development Question
OK, well that would be the most preferred way. That allows everyone to see the flow and what each person is doing. I don’t want my own techs working on items outside of my own ticketing system because then I have no idea what the hell they are doing and there’s no method of tracking. I mean the bottom line is most projects/support teams work out of tickets for support/bug issues.
Wait, you didn’t know? They already had one someone held the title of “Community Leader” and it was their job to deal with community facing stuff (Forums/IRC). Was here for years. Didn’t you notice?
OK well the ticketing system isn’t Sangoma. It’s JIRA, the same ticketing system I use. The same one Asterisk uses and the same one many of the large development projects/companies I’ve worked with use. So this one is more a you just don’t like any project that uses JIRA for its ticketing/issues tracking system.
Internal call answered externaly
OK this is now the point to where you show us the whole call not just a snippet or one packet of the call like your previous examples. Put it on pastebin so the formatting is correct and give us the link.
We can’t guess what things look like or how they are actually being sent. We need to see actual data.
Make call and play sound from external system
No - quite the opposite. There are at least six ways to do this, not counting the way that is working for you right now. The thing that isn’t available is a way to do this from the FreePBX GUI.
No matter how you do it (CallFile, AMI, ARI, REST, SangomaCRM, Originate() scripts, etc.) you are going to have to write it yourself. There is no single way to do this. All of them have their advantages and disadvantages. None of them is a “service”.
IAX local extensions not regster
OK. Time to fish or cut bait. I’m not going to help with this any more unless I see some output from /var/log/asterisk/full showing the extension getting kicked off the network. Also, “it doesn’t work” isn’t enough “more information” for us to be able to help you.
Since Intrusion Detection watches the log files and locks hosts out when then do specific things, you need to look at the log files and figure out what is happening that is triggering the ID module. Without logs, there’s no way anyone can help you.
Reset feature codes for extension to default
call forwarding, dnd, login into and out of queues.
I’ve noticed that a number of handsets, have issues with basic functions, such as tranferring calls, simple calling from another handset, but they do function within the queue properly.
On more than one occasion I’ve found the end user has accidentally, not understanding entered random function codes that effect that hand set, and no other.
So, when I look at 2 handsets, one working perfectly, the other which won’t accept call transfered to voicemail, or don’t receive calls from queue. I review settings, identical, but when you look at functions codes for the handset, such as DND, you find a feature on. I see it very often with receptionist, who has very long nails, and hits stray keys.
I don’t know that simply setting a handset to factory defaults, then reintroducing it to our pbx would help, if that extension still has some stray function set to on or off.
Dial Pattern Problems
How outdated? I’ve been using FreePBX for pretty much all of the 21st Century, and multi-tenant FreePBX has always been a problem.
This isn’t actually how the pattern matching works in Outbound Routes. It will try all of the outbound routes looking for a pattern match, then failing that, will look for one that has the least restrictions. You MUST match an outbound route pattern, so you’re going to have to show us what you are doing and explain it in a little more detail.
One other thing to keep in mind: the underlying Asterisk implementation could also be jacking you up. One “feature” with Asterisk is that your provider may consolidate all of your inbound routes into the first SIP account that matches their configuration - usually IP address. Because of this, your provider may drop a call into an “account” that isn’t that customer’s account and, if they are messing with Caller ID, screw with the presentation of the account appropriately.
Logs would actually be a good step here. Look at which trunks are being chosen to find what is matching in the outbound routes, then make sure that the provider isn’t “simplifying” your calling efforts in a way that screws you up.
Custom dialplan
Post call survey sounds like a good start.
Trunk callerID provider (dialplan)
Let’s talk about how Caller ID gets set - perhaps you’ll get why that is a hard question:
There are several places where Caller ID can be set for an outbound call:
-
If you are using a Softphone, you can set the Caller ID in the phone itself. Most “hard” phones don’t do this - they rely on Level 2. EIther way, level 2 can override this option.
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Your extension configuration can set the phone’s Caller ID. This is handy for places where the individual phone decides what the output CID should be. For example, if you want “COMPANY BILLING” to show up in your CID, you set that at the Extension level. It could be overridden by Level 3.
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Your Outbound Routes have the option of setting and overriding all calls that match that Outbound Route. This can be handy for a bunch of different things. Like all the rest, it can be overridden by Level 4.
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Your Trunk has the option of setting and overriding all calls through that particular trunk. This isn’t as handy as one would hope, since it messes with FMFM, but other than that, if you need it, you have it. It is kind of nice for “home phone” networks. Like the others, it can be overridden by Level 5.
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Your provider may completely ignore all of that and set the Caller ID for you to whatever they want.
As if that wasn’t enough fun, sometimes your calls will get identified using the CNAME set for your phone number in the CNAME database your provider uses.
Now that you know all of the places you and other people can mess with your Caller ID, the only way to know which is setting the final CID is to look through the logs and to understand that you might not be getting a vote in the first place.