Hello, I’m new in ASterisk and FreePBX and wanteted to have a Server for our company. I succesfully set up a working Server, where I can call out- & inside of my office with some benefits like conference and listening, whispering. Its awsome whats possible.
After I installed a Zabbix-Agent and reconfigured my FreePBX firewall (currently disabled), I wasn’t able to call outside and I cann’t figure out why. It would be really nice if someone could help me out.
Here are some informations about my Server:
XXXXXXXXXX is my Sipgate.de ID
YYYYYYYYYY is my Sipgate.de Secret
Users:
freepbx*CLI> sip show users
Username Secret Accountcode Def.Context ACL Forcerport
1000 1000 from-internal Yes No
1001 1001 from-internal Yes No
1003 1003 from-internal Yes No
Peers:
freepbx*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
1000 (Unspecified) D No No A 0 UNKNOWN
1001/1001 192.168.1.32 D No No A 5060 OK (26 ms)
1003/1003 192.168.1.31 D No No A 5060 OK (41 ms)
sipgate.de/XXXXXXXXXX 217.10.79.9 No No 5060 OK (29 ms)
Registry:
freepbx*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
sipgate.de:5060 N XXXXXXXXXX 105 Registered Thu, 26 Oct 2017 16:03:10
1 SIP registrations.
sip_additional.conf:
[1000]
deny=0.0.0.0/0.0.0.0
secret=1000
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
defaultuser=
trustrpid=yes
sendrpid=pai
type=friend
session-timers=accept
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
force_avp=no
icesupport=no
rtcp_mux=no
encryption=no
namedcallgroup=
namedpickupgroup=
dial=SIP/1000
permit=0.0.0.0/0.0.0.0
callerid=Admin <1000>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
[1001]
deny=0.0.0.0/0.0.0.0
secret=1001
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
defaultuser=
trustrpid=yes
sendrpid=pai
type=friend
session-timers=accept
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
force_avp=no
icesupport=no
rtcp_mux=no
encryption=no
namedcallgroup=
namedpickupgroup=
dial=SIP/1001
permit=0.0.0.0/0.0.0.0
callerid=Asterisk 1 <1001>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
[1003]
deny=0.0.0.0/0.0.0.0
secret=1003
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
defaultuser=
trustrpid=yes
sendrpid=pai
type=friend
session-timers=accept
nat=no
port=5060
qualify=yes
qualifyfreq=60
transport=udp
avpf=no
force_avp=no
icesupport=no
rtcp_mux=no
encryption=no
namedcallgroup=
namedpickupgroup=
dial=SIP/1003
permit=0.0.0.0/0.0.0.0
callerid=Asterisk 3 <1003>
callcounter=yes
faxdetect=no
cc_monitor_policy=generic
[sipgate.de]
disallow=all
username=XXXXXXXXXX
type=peer
secret=YYYYYYYYYYYY
qualify=yes
nat=no
insecure=port,invite
host=sipgate.de
outboundproxy=sipgate.de
port=5060
fromuser=XXXXXXXXXX
fromdomain=sipgate.de
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
registertimeout=600
allow=gsm
allow=ulaw
allow=alaw
Outbound Routes in FreePBX GUI:
Route CID = 10002
Route Name = external
Trunk Sequence for Matched Routes = Sipgate
With Pattern = X.
My Incomming Registry String:
[SIP-ID]:[SIP-SECRET]@[SIP-HOST]/[SIP-ID]
Log of a outside call in ‘-rvvvvvvv’ mode:
<--- SIP read from UDP:192.168.1.32:5060 --->
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.32:5060;branch=z9hG4bK6a54304fce0e6d60d3ba5cb53dd749bb;rport
From: "Asterisk 1" <sip:[email protected]>;tag=1579942557
To: <sip:[email protected];user=phone>
Call-ID: 1430505037@192_168_1_32
CSeq: 2 INVITE
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
User-Agent: S850A GO/42.243.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile, talk, check-sync
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 381
v=0
o=1001 5016 15 IN IP4 192.168.1.32
s=Mapping
c=IN IP4 192.168.1.32
t=0 0
m=audio 5016 RTP/AVP 9 8 0 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (14 headers 17 lines) ---
Sending to 192.168.1.32:5060 (no NAT)
Sending to 192.168.1.32:5060 (no NAT)
Using INVITE request as basis request - 1430505037@192_168_1_32
Found peer '1001' for '1001' from 192.168.1.32:5060
<--- Reliably Transmitting (no NAT) to 192.168.1.32:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.32:5060;branch=z9hG4bK6a54304fce0e6d60d3ba5cb53dd749bb;received=192.168.1.32;rport=5060
From: "Asterisk 1" <sip:[email protected]>;tag=1579942557
To: <sip:[email protected];user=phone>;tag=as39a6862f
Call-ID: 1430505037@192_168_1_32
CSeq: 2 INVITE
Server: FPBX-14.0.1.20(13.17.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4edef15d"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1430505037@192_168_1_32' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:192.168.1.32:5060 --->
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.32:5060;branch=z9hG4bK6a54304fce0e6d60d3ba5cb53dd749bb;rport
From: "Asterisk 1" <sip:[email protected]>;tag=1579942557
To: <sip:[email protected];user=phone>;tag=as39a6862f
Call-ID: 1430505037@192_168_1_32
CSeq: 2 ACK
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
User-Agent: S850A GO/42.243.00.000.000
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:192.168.1.32:5060 --->
INVITE sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.32:5060;branch=z9hG4bKc5aa4eca59453b3492bd34365545350d;rport
From: "Asterisk 1" <sip:[email protected]>;tag=1579942557
To: <sip:[email protected];user=phone>
Call-ID: 1430505037@192_168_1_32
CSeq: 3 INVITE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:[email protected];user=phone", nonce="4edef15d", response="dd27d34df9c4236fa91031b5efdfe303"
Max-Forwards: 70
User-Agent: S850A GO/42.243.00.000.000
Supported: replaces
Allow-Events: message-summary, refer, ua-profile, talk, check-sync
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 381
v=0
o=1001 5016 15 IN IP4 192.168.1.32
s=Mapping
c=IN IP4 192.168.1.32
t=0 0
m=audio 5016 RTP/AVP 9 8 0 96 97 2 18 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 G726-32/8000
a=rtpmap:97 AAL2-G726-32/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (15 headers 17 lines) ---
Sending to 192.168.1.32:5060 (no NAT)
Using INVITE request as basis request - 1430505037@192_168_1_32
Found peer '1001' for '1001' from 192.168.1.32:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 2
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 96
Found audio description format AAL2-G726-32 for ID 97
Found audio description format G726-32 for ID 2
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726|g722), peer - audio=(ulaw|g726|alaw|g722|g729|g726aal2)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g726|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0x7fe00002bdd0 -- Strict RTP learning after remote address set to: 192.168.1.32:5016
Peer audio RTP is at port 192.168.1.32:5016
Looking for 01520ZZZZZZZ in from-internal (domain 192.168.1.29)
sip_route_dump: route/path hop: <sip:[email protected]:5060>
<--- Transmitting (no NAT) to 192.168.1.32:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.32:5060;branch=z9hG4bKc5aa4eca59453b3492bd34365545350d;received=192.168.1.32;rport=5060
From: "Asterisk 1" <sip:[email protected]>;tag=1579942557
To: <sip:[email protected];user=phone>
Call-ID: 1430505037@192_168_1_32
CSeq: 3 INVITE
Server: FPBX-14.0.1.20(13.17.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
-- Executing [01520ZZZZZZZ@from-internal:1] Macro("SIP/1001-00000085", "user-callerid,LIMIT,EXTERNAL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/1001-00000085", "TOUCH_MONITOR=1509030681.133") in new stack
-- Executing [s@macro-user-callerid:2] Set("SIP/1001-00000085", "AMPUSER=1001") in new stack
-- Executing [s@macro-user-callerid:3] GotoIf("SIP/1001-00000085", "0?report") in new stack
-- Executing [s@macro-user-callerid:4] ExecIf("SIP/1001-00000085", "1?Set(REALCALLERIDNUM=1001)") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/1001-00000085", "AMPUSER=1001") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/1001-00000085", "0?limit") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/1001-00000085", "AMPUSERCIDNAME=Asterisk 1") in new stack
-- Executing [s@macro-user-callerid:8] ExecIf("SIP/1001-00000085", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
-- Executing [s@macro-user-callerid:9] GotoIf("SIP/1001-00000085", "0?report") in new stack
-- Executing [s@macro-user-callerid:10] Set("SIP/1001-00000085", "AMPUSERCID=1001") in new stack
-- Executing [s@macro-user-callerid:11] Set("SIP/1001-00000085", "__DIAL_OPTIONS=") in new stack
-- Executing [s@macro-user-callerid:12] Set("SIP/1001-00000085", "CALLERID(all)="Asterisk 1" <1001>") in new stack
-- Executing [s@macro-user-callerid:13] GotoIf("SIP/1001-00000085", "0?limit") in new stack
-- Executing [s@macro-user-callerid:14] ExecIf("SIP/1001-00000085", "1?Set(GROUP(concurrency_limit)=1001)") in new stack
-- Executing [s@macro-user-callerid:15] ExecIf("SIP/1001-00000085", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:16] NoOp("SIP/1001-00000085", "Macro Depth is 1") in new stack
-- Executing [s@macro-user-callerid:17] GotoIf("SIP/1001-00000085", "1?report2:macroerror") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] GotoIf("SIP/1001-00000085", "1?continue") in new stack
-- Goto (macro-user-callerid,s,37)
-- Executing [s@macro-user-callerid:37] Set("SIP/1001-00000085", "CALLERID(number)=1001") in new stack
-- Executing [s@macro-user-callerid:38] Set("SIP/1001-00000085", "CALLERID(name)=Asterisk 1") in new stack
-- Executing [s@macro-user-callerid:39] GotoIf("SIP/1001-00000085", "0?cnum") in new stack
-- Executing [s@macro-user-callerid:40] Set("SIP/1001-00000085", "CDR(cnam)=Asterisk 1") in new stack
-- Executing [s@macro-user-callerid:41] Set("SIP/1001-00000085", "CDR(cnum)=1001") in new stack
-- Executing [s@macro-user-callerid:42] Set("SIP/1001-00000085", "CHANNEL(language)=en_GB") in new stack
-- Executing [01520ZZZZZZZ@from-internal:2] Gosub("SIP/1001-00000085", "sub-record-check,s,1(out,01520ZZZZZZZ,no)") in new stack
-- Executing [s@sub-record-check:1] GotoIf("SIP/1001-00000085", "0?initialized") in new stack
-- Executing [s@sub-record-check:2] Set("SIP/1001-00000085", "__REC_STATUS=INITIALIZED") in new stack
-- Executing [s@sub-record-check:3] Set("SIP/1001-00000085", "NOW=1509030681") in new stack
-- Executing [s@sub-record-check:4] Set("SIP/1001-00000085", "__DAY=26") in new stack
-- Executing [s@sub-record-check:5] Set("SIP/1001-00000085", "__MONTH=10") in new stack
-- Executing [s@sub-record-check:6] Set("SIP/1001-00000085", "__YEAR=2017") in new stack
-- Executing [s@sub-record-check:7] Set("SIP/1001-00000085", "__TIMESTR=20171026-151121") in new stack
-- Executing [s@sub-record-check:8] Set("SIP/1001-00000085", "__FROMEXTEN=1001") in new stack
-- Executing [s@sub-record-check:9] Set("SIP/1001-00000085", "__MON_FMT=ogg") in new stack
-- Executing [s@sub-record-check:10] NoOp("SIP/1001-00000085", "Recordings initialized") in new stack
-- Executing [s@sub-record-check:11] ExecIf("SIP/1001-00000085", "0?Set(ARG3=dontcare)") in new stack
-- Executing [s@sub-record-check:12] Set("SIP/1001-00000085", "REC_POLICY_MODE_SAVE=") in new stack
-- Executing [s@sub-record-check:13] ExecIf("SIP/1001-00000085", "0?Set(REC_STATUS=NO)") in new stack
-- Executing [s@sub-record-check:14] GotoIf("SIP/1001-00000085", "3?checkaction") in new stack
-- Goto (sub-record-check,s,17)
-- Executing [s@sub-record-check:17] GotoIf("SIP/1001-00000085", "1?sub-record-check,out,1") in new stack
-- Goto (sub-record-check,out,1)
-- Executing [out@sub-record-check:1] NoOp("SIP/1001-00000085", "Outbound Recording Check from 1001 to 01520ZZZZZZZ") in new stack
-- Executing [out@sub-record-check:2] Set("SIP/1001-00000085", "RECMODE=dontcare") in new stack
-- Executing [out@sub-record-check:3] ExecIf("SIP/1001-00000085", "1?Goto(routewins)") in new stack
-- Goto (sub-record-check,out,7)
-- Executing [out@sub-record-check:7] Gosub("SIP/1001-00000085", "recordcheck,1(no,out,01520ZZZZZZZ)") in new stack
-- Executing [recordcheck@sub-record-check:1] NoOp("SIP/1001-00000085", "Starting recording check against no") in new stack
-- Executing [recordcheck@sub-record-check:2] Goto("SIP/1001-00000085", "no") in new stack
-- Goto (sub-record-check,recordcheck,12)
-- Executing [recordcheck@sub-record-check:12] Set("SIP/1001-00000085", "__REC_POLICY_MODE=NO") in new stack
-- Executing [recordcheck@sub-record-check:13] Return("SIP/1001-00000085", "") in new stack
-- Executing [out@sub-record-check:8] Return("SIP/1001-00000085", "") in new stack
-- Executing [01520ZZZZZZZ@from-internal:3] ExecIf("SIP/1001-00000085", "0 ?Set(CDR(accountcode)=)") in new stack
-- Executing [01520ZZZZZZZ@from-internal:4] Set("SIP/1001-00000085", "ROUTE_CIDSAVE="Asterisk 1" <1001>") in new stack
-- Executing [01520ZZZZZZZ@from-internal:5] Set("SIP/1001-00000085", "MOHCLASS=default") in new stack
-- Executing [01520ZZZZZZZ@from-internal:6] ExecIf("SIP/1001-00000085", "0?Set(TRUNKCIDOVERRIDE=10002)") in new stack
-- Executing [01520ZZZZZZZ@from-internal:7] Set("SIP/1001-00000085", "_NODEST=") in new stack
-- Executing [01520ZZZZZZZ@from-internal:8] Macro("SIP/1001-00000085", "dialout-trunk,1,01520ZZZZZZZ,,off") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/1001-00000085", "DIAL_TRUNK=1") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1001-00000085", "0?sub-pincheck,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:3] ExecIf("SIP/1001-00000085", "0?Set(CALLERID(num)=1001)") in new stack
-- Executing [s@macro-dialout-trunk:4] GotoIf("SIP/1001-00000085", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/1001-00000085", "DIAL_NUMBER=01520ZZZZZZZ") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/1001-00000085", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:7] Set("SIP/1001-00000085", "OUTBOUND_GROUP=OUT_1") in new stack
-- Executing [s@macro-dialout-trunk:8] Set("SIP/1001-00000085", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1001-00000085", "1?nomax") in new stack
-- Goto (macro-dialout-trunk,s,11)
-- Executing [s@macro-dialout-trunk:11] GotoIf("SIP/1001-00000085", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:12] Macro("SIP/1001-00000085", "outbound-callerid,1") in new stack
-- Executing [s@macro-outbound-callerid:1] NoOp("SIP/1001-00000085", "1001") in new stack
-- Executing [s@macro-outbound-callerid:2] NoOp("SIP/1001-00000085", "") in new stack
-- Executing [s@macro-outbound-callerid:3] NoOp("SIP/1001-00000085", "") in new stack
-- Executing [s@macro-outbound-callerid:4] ExecIf("SIP/1001-00000085", "0?Set(CALLERPRES(name-pres)=)") in new stack
-- Executing [s@macro-outbound-callerid:5] ExecIf("SIP/1001-00000085", "0?Set(CALLERPRES(num-pres)=)") in new stack
-- Executing [s@macro-outbound-callerid:6] ExecIf("SIP/1001-00000085", "0?Set(REALCALLERIDNUM=1001)") in new stack
-- Executing [s@macro-outbound-callerid:7] GotoIf("SIP/1001-00000085", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,11)
-- Executing [s@macro-outbound-callerid:11] Set("SIP/1001-00000085", "USEROUTCID=1001") in new stack
-- Executing [s@macro-outbound-callerid:12] Set("SIP/1001-00000085", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:13] Set("SIP/1001-00000085", "TRUNKOUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:14] GotoIf("SIP/1001-00000085", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,19)
-- Executing [s@macro-outbound-callerid:19] ExecIf("SIP/1001-00000085", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:20] ExecIf("SIP/1001-00000085", "1?Set(CALLERID(all)=1001)") in new stack
-- Executing [s@macro-outbound-callerid:21] ExecIf("SIP/1001-00000085", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:22] ExecIf("SIP/1001-00000085", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:23] ExecIf("SIP/1001-00000085", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
-- Executing [s@macro-outbound-callerid:24] Set("SIP/1001-00000085", "CDR(outbound_cnum)=1001") in new stack
-- Executing [s@macro-outbound-callerid:25] Set("SIP/1001-00000085", "CDR(outbound_cnam)=") in new stack
-- Executing [s@macro-dialout-trunk:13] GosubIf("SIP/1001-00000085", "0?sub-flp-1,s,1()") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/1001-00000085", "OUTNUM=01520ZZZZZZZ") in new stack
-- Executing [s@macro-dialout-trunk:15] Set("SIP/1001-00000085", "custom=") in new stack
-- Executing [s@macro-dialout-trunk:16] ExecIf("SIP/1001-00000085", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
-- Executing [s@macro-dialout-trunk:17] ExecIf("SIP/1001-00000085", "0?Set(DIAL_TRUNK_OPTIONS=M(confirm))") in new stack
-- Executing [s@macro-dialout-trunk:18] Macro("SIP/1001-00000085", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1001-00000085", "") in new stack
-- Executing [s@macro-dialout-trunk:19] GotoIf("SIP/1001-00000085", "0?skipcrm") in new stack
-- Executing [s@macro-dialout-trunk:20] Set("SIP/1001-00000085", "__CRM_DIRECTION=OUTBOUND") in new stack
-- Executing [s@macro-dialout-trunk:21] Set("SIP/1001-00000085", "__CRM_DESTINATION=01520ZZZZZZZ") in new stack
-- Executing [s@macro-dialout-trunk:22] Set("SIP/1001-00000085", "__CRM_SOURCE=1001") in new stack
-- Executing [s@macro-dialout-trunk:23] AGI("SIP/1001-00000085", "sangomacrm.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
-- <SIP/1001-00000085>AGI Script sangomacrm.agi completed, returning 0
-- Executing [s@macro-dialout-trunk:24] Set("SIP/1001-00000085", "CHANNEL(hangup_handler_push)=crm-hangup,s,1") in new stack
-- Executing [s@macro-dialout-trunk:25] NoOp("SIP/1001-00000085", "CRM Finished") in new stack
-- Executing [s@macro-dialout-trunk:26] GotoIf("SIP/1001-00000085", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:27] ExecIf("SIP/1001-00000085", "1?Set(CONNECTEDLINE(num,i)=01520ZZZZZZZ)") in new stack
-- Executing [s@macro-dialout-trunk:28] ExecIf("SIP/1001-00000085", "1?Set(CONNECTEDLINE(name,i)=CID:1001)") in new stack
-- Executing [s@macro-dialout-trunk:29] ExecIf("SIP/1001-00000085", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)1001)") in new stack
-- Executing [s@macro-dialout-trunk:30] GotoIf("SIP/1001-00000085", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:31] Dial("SIP/1001-00000085", "/01520ZZZZZZZ,,") in new stack
[2017-10-26 15:11:21] WARNING[7759][C-0000003f]: channel.c:6262 ast_request: No channel type registered for ''
[2017-10-26 15:11:21] WARNING[7759][C-0000003f]: app_dial.c:2525 dial_exec_full: Unable to create channel of type '' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:32] NoOp("SIP/1001-00000085", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 66") in new stack
-- Executing [s@macro-dialout-trunk:33] GotoIf("SIP/1001-00000085", "0?continue,1:s-CHANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/1001-00000085", "RC=66") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/1001-00000085", "66,1") in new stack
-- Goto (macro-dialout-trunk,66,1)
-- Executing [66@macro-dialout-trunk:1] Goto("SIP/1001-00000085", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/1001-00000085", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 66 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:2] ExecIf("SIP/1001-00000085", "1?Set(CALLERID(number)=1001)") in new stack
-- Executing [01520ZZZZZZZ@from-internal:9] Set("SIP/1001-00000085", "CALLERID(all)="Asterisk 1" <1001>") in new stack
-- Executing [01520ZZZZZZZ@from-internal:10] Set("SIP/1001-00000085", "_KEEPCID=TRUE") in new stack
-- Executing [01520ZZZZZZZ@from-internal:11] Goto("SIP/1001-00000085", "ext-trunk,1,1") in new stack
-- Goto (ext-trunk,1,1)
-- Executing [1@ext-trunk:1] Set("SIP/1001-00000085", "TDIAL_STRING=SIP/sipgate.de") in new stack
-- Executing [1@ext-trunk:2] Set("SIP/1001-00000085", "DIAL_TRUNK=1") in new stack
-- Executing [1@ext-trunk:3] Goto("SIP/1001-00000085", "ext-trunk,tdial,1") in new stack
-- Goto (ext-trunk,tdial,1)
-- Executing [tdial@ext-trunk:1] Set("SIP/1001-00000085", "OUTBOUND_GROUP=OUT_1") in new stack
-- Executing [tdial@ext-trunk:2] GotoIf("SIP/1001-00000085", "1?nomax") in new stack
-- Goto (ext-trunk,tdial,4)
-- Executing [tdial@ext-trunk:4] ExecIf("SIP/1001-00000085", "0?Set(CALLERPRES(name-pres)=)") in new stack
-- Executing [tdial@ext-trunk:5] ExecIf("SIP/1001-00000085", "0?Set(CALLERPRES(num-pres)=)") in new stack
-- Executing [tdial@ext-trunk:6] Set("SIP/1001-00000085", "DIAL_NUMBER=") in new stack
-- Executing [tdial@ext-trunk:7] GosubIf("SIP/1001-00000085", "0?sub-flp-1,s,1()") in new stack
-- Executing [tdial@ext-trunk:8] Set("SIP/1001-00000085", "OUTNUM=") in new stack
-- Executing [tdial@ext-trunk:9] Set("SIP/1001-00000085", "DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [tdial@ext-trunk:10] Dial("SIP/1001-00000085", "SIP/sipgate.de/,,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 17330
Adding codec ulaw to SDP
Adding codec gsm to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 217.10.79.9:5060:
INVITE sip:sipgate.de:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK30379ab0
Max-Forwards: 70
From: "Asterisk 1" <sip:[email protected]>;tag=as288e4e96
To: <sip:sipgate.de:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FPBX-14.0.1.20(13.17.2)
Date: Thu, 26 Oct 2017 15:11:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 297
v=0
o=root 1684954234 1684954234 IN IP4 192.168.1.1
s=Asterisk PBX 13.17.2
c=IN IP4 192.168.1.1
t=0 0
m=audio 17330 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
-- Called SIP/sipgate.de/
<--- SIP read from UDP:217.10.79.9:5060 --->
SIP/2.0 404 Not found (no match)
Via: SIP/2.0/UDP 192.168.1.1:5060;rport=61714;received=91.58.239.60;branch=z9hG4bK30379ab0
From: "Asterisk 1" <sip:[email protected]>;tag=as288e4e96
To: <sip:sipgate.de:5060>;tag=86e53dd608d1c001e0b8060625977563.cad7
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Transmitting (no NAT) to 217.10.79.9:5060:
ACK sip:sipgate.de:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK30379ab0
Max-Forwards: 70
From: "Asterisk 1" <sip:[email protected]>;tag=as288e4e96
To: <sip:sipgate.de:5060>;tag=86e53dd608d1c001e0b8060625977563.cad7
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: FPBX-14.0.1.20(13.17.2)
Content-Length: 0
---
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [tdial@ext-trunk:11] Set("SIP/1001-00000085", "CALLERID(number)=1001") in new stack
-- Executing [tdial@ext-trunk:12] Set("SIP/1001-00000085", "CALLERID(name)=Asterisk 1") in new stack
-- Executing [tdial@ext-trunk:13] Hangup("SIP/1001-00000085", "") in new stack
== Spawn extension (ext-trunk, tdial, 13) exited non-zero on 'SIP/1001-00000085'
-- SIP/1001-00000085 Internal Gosub(crm-hangup,s,1) start
-- Executing [s@crm-hangup:1] NoOp("SIP/1001-00000085", "Sending Hangup to CRM") in new stack
-- Executing [s@crm-hangup:2] NoOp("SIP/1001-00000085", "HANGUP CAUSE: 1") in new stack
-- Executing [s@crm-hangup:3] ExecIf("SIP/1001-00000085", "0?Set(__CRM_VOICEMAIL=)") in new stack
-- Executing [s@crm-hangup:4] NoOp("SIP/1001-00000085", "MASTER CHANNEL: 1509030681.133 = 1509030681.133") in new stack
-- Executing [s@crm-hangup:5] GotoIf("SIP/1001-00000085", "0?return") in new stack
-- Executing [s@crm-hangup:6] Set("SIP/1001-00000085", "__CRM_HANGUP=1") in new stack
-- Executing [s@crm-hangup:7] AGI("SIP/1001-00000085", "sangomacrm.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
-- <SIP/1001-00000085>AGI Script sangomacrm.agi completed, returning 0
-- Executing [s@crm-hangup:8] Return("SIP/1001-00000085", "") in new stack
== Spawn extension (ext-trunk, tdial, 13) exited non-zero on 'SIP/1001-00000085'
-- SIP/1001-00000085 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
Scheduling destruction of SIP dialog '1430505037@192_168_1_32' in 6400 ms (Method: INVITE)
<--- Reliably Transmitting (no NAT) to 192.168.1.32:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.32:5060;branch=z9hG4bKc5aa4eca59453b3492bd34365545350d;received=192.168.1.32;rport=5060
From: "Asterisk 1" <sip:[email protected]>;tag=1579942557
To: <sip:[email protected];user=phone>;tag=as461f27e6
Call-ID: 1430505037@192_168_1_32
CSeq: 3 INVITE
Server: FPBX-14.0.1.20(13.17.2)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
<--- SIP read from UDP:192.168.1.32:5060 --->
ACK sip:[email protected];user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.32:5060;branch=z9hG4bKc5aa4eca59453b3492bd34365545350d;rport
From: "Asterisk 1" <sip:[email protected]>;tag=1579942557
To: <sip:[email protected];user=phone>;tag=as461f27e6
Call-ID: 1430505037@192_168_1_32
CSeq: 3 ACK
Contact: <sip:[email protected]:5060>
Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:[email protected];user=phone", nonce="4edef15d", response="dd27d34df9c4236fa91031b5efdfe303"
Max-Forwards: 70
User-Agent: S850A GO/42.243.00.000.000
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---