[SOLVED]Callout disconnect 6400ms - sip-retransmission

Ezra5447 · January 26, 2018, 9:20am

Hi Guys, having weird issue on callout from asterisk when incoming call is working fine.
tried to follow with the sip transmission web link but not so sure about the solutions as the asterisk running behind fortigate, and the users must connect to asterisk by using the VPN.

the call flow is
Softphone + Forticlient VPN > asterisk > trunks.

Hopefully you guys can help me, thanks in advance.

will provide the sip debug at below

Ezra5447 · January 26, 2018, 12:42pm

this is my sip set debug,

<— SIP read from UDP:192.168.8.1:65069 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.8.1:65069;branch=z9hG4bK001acebce300e811ba5eb0a0553453f5;rport
From: “PhonerLite” sip:[email protected];tag=4209725802
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 7 INVITE
Contact: sip:[email protected]:65069
Content-Type: application/sdp
Mime-Version: 1.0
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
P-Early-Media: supported
User-Agent: SIPPER for PhonerLite
Session-Expires: 1800
Supported: 100rel, replaces, from-change, timer
P-Preferred-Identity: sip:[email protected]
Content-Length: 491

v=0
o=- 230988866 1 IN IP4 192.168.8.1
s=SIPPER for PhonerLite
c=IN IP4 192.168.8.1
t=0 0
m=audio 10922 RTP/AVP 107 8 0 2 3 97 110 111 9 18 101
a=rtpmap:107 opus/48000/2
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:4265427681
a=sendrecv
<------------->
— (17 headers 21 lines) —
Sending to 192.168.8.1:65069 (NAT)
Sending to 192.168.8.1:65069 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘110002’ for ‘110002’ from 192.168.8.1:65069

<— Reliably Transmitting (no NAT) to 192.168.8.1:65069 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.8.1:65069;branch=z9hG4bK001acebce300e811ba5eb0a0553453f5;received=192.168.8.1;rport=65069
From: “PhonerLite” sip:[email protected];tag=4209725802
To: sip:[email protected];tag=as413a6756
Call-ID: [email protected]
CSeq: 7 INVITE
Server: FPBX-13.0.192.19(11.23.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="6da12b64"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:192.168.8.1:65069 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.8.1:65069;branch=z9hG4bK001acebce300e811ba5eb0a0553453f5;rport
From: “PhonerLite” sip:[email protected];tag=4209725802
To: sip:[email protected];tag=as413a6756
Call-ID: [email protected]
CSeq: 7 ACK
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:192.168.8.1:65069 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.8.1:65069;branch=z9hG4bK001acebce300e811ba5fb0a0553453f5;rport
From: “PhonerLite” sip:[email protected];tag=4209725802
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 8 INVITE
Contact: sip:[email protected]:65069
Authorization: Digest username=“110002”, realm=“asterisk”, nonce=“6da12b64”, uri="sip:[email protected]", response=“3e76cba967c51fd2bedf5dd718a04cd8”, algorithm=MD5
Content-Type: application/sdp
Mime-Version: 1.0
Allow: INVITE, ACK, BYE, CANCEL, INFO, MESSAGE, NOTIFY, OPTIONS, REFER, UPDATE, PRACK
Max-Forwards: 70
P-Early-Media: supported
User-Agent: SIPPER for PhonerLite
Session-Expires: 1800
Supported: 100rel, replaces, from-change, timer
P-Preferred-Identity: sip:[email protected]
Content-Length: 491

v=0
o=- 230988866 1 IN IP4 192.168.8.1
s=SIPPER for PhonerLite
c=IN IP4 192.168.8.1
t=0 0
m=audio 10922 RTP/AVP 107 8 0 2 3 97 110 111 9 18 101
a=rtpmap:107 opus/48000/2
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:4265427681
a=sendrecv
<------------->
— (18 headers 21 lines) —
Sending to 192.168.8.1:65069 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘110002’ for ‘110002’ from 192.168.8.1:65069
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 107
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 9
Found RTP audio format 18
Found RTP audio format 101
Found unknown media description format opus for ID 107
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G726-32 for ID 2
Found audio description format GSM for ID 3
Found audio description format iLBC for ID 97
Found audio description format speex for ID 110
Found audio description format speex for ID 111
Found audio description format G722 for ID 9
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw), peer - audio=(gsm|ulaw|alaw|g726|g729|speex|speex16|ilbc|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.8.1:10922
Looking for 0060146349060 in from-internal (domain 110.74.163.106)
list_route: hop: sip:[email protected]:65069

<— Transmitting (no NAT) to 192.168.8.1:65069 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.8.1:65069;branch=z9hG4bK001acebce300e811ba5fb0a0553453f5;received=192.168.8.1;rport=65069
From: “PhonerLite” sip:[email protected];tag=4209725802
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 8 INVITE
Server: FPBX-13.0.192.19(11.23.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Length: 0

<------------>
– Executing [0060146349060@from-internal:1] Macro(“SIP/110002-00001c7f”, “user-callerid,LIMIT,EXTERNAL,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/110002-00001c7f”, “TOUCH_MONITOR=1516956676.7297”) in new stack
– Executing [s@macro-user-callerid:2] Set(“SIP/110002-00001c7f”, “AMPUSER=110002”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“SIP/110002-00001c7f”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] ExecIf(“SIP/110002-00001c7f”, “1?Set(REALCALLERIDNUM=110002)”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/110002-00001c7f”, “AMPUSER=110002”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/110002-00001c7f”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/110002-00001c7f”, “AMPUSERCIDNAME=110002”) in new stack
– Executing [s@macro-user-callerid:8] GotoIf(“SIP/110002-00001c7f”, “0?report”) in new stack
– Executing [s@macro-user-callerid:9] Set(“SIP/110002-00001c7f”, “AMPUSERCID=110002”) in new stack
– Executing [s@macro-user-callerid:10] Set(“SIP/110002-00001c7f”, “__DIAL_OPTIONS=Ttr”) in new stack
– Executing [s@macro-user-callerid:11] Set(“SIP/110002-00001c7f”, “CALLERID(all)=“110002” <110002>”) in new stack
– Executing [s@macro-user-callerid:12] GotoIf(“SIP/110002-00001c7f”, “0?limit”) in new stack
– Executing [s@macro-user-callerid:13] ExecIf(“SIP/110002-00001c7f”, “1?Set(GROUP(concurrency_limit)=110002)”) in new stack
– Executing [s@macro-user-callerid:14] ExecIf(“SIP/110002-00001c7f”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:15] NoOp(“SIP/110002-00001c7f”, “Macro Depth is 1”) in new stack
– Executing [s@macro-user-callerid:16] GotoIf(“SIP/110002-00001c7f”, “1?report2:macroerror”) in new stack
– Goto (macro-user-callerid,s,18)
– Executing [s@macro-user-callerid:18] GotoIf(“SIP/110002-00001c7f”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,36)
– Executing [s@macro-user-callerid:36] Set(“SIP/110002-00001c7f”, “CALLERID(number)=110002”) in new stack
– Executing [s@macro-user-callerid:37] Set(“SIP/110002-00001c7f”, “CALLERID(name)=110002”) in new stack
– Executing [s@macro-user-callerid:38] GotoIf(“SIP/110002-00001c7f”, “0?cnum”) in new stack
– Executing [s@macro-user-callerid:39] Set(“SIP/110002-00001c7f”, “CDR(cnam)=110002”) in new stack
– Executing [s@macro-user-callerid:40] Set(“SIP/110002-00001c7f”, “CDR(cnum)=110002”) in new stack
– Executing [s@macro-user-callerid:41] Set(“SIP/110002-00001c7f”, “CHANNEL(language)=en”) in new stack
– Executing [0060146349060@from-internal:2] Gosub(“SIP/110002-00001c7f”, “sub-record-check,s,1(out,0060146349060,dontcare)”) in new stack
– Executing [s@sub-record-check:1] GotoIf(“SIP/110002-00001c7f”, “0?initialized”) in new stack
– Executing [s@sub-record-check:2] Set(“SIP/110002-00001c7f”, “__REC_STATUS=INITIALIZED”) in new stack
– Executing [s@sub-record-check:3] Set(“SIP/110002-00001c7f”, “NOW=1516956676”) in new stack
– Executing [s@sub-record-check:4] Set(“SIP/110002-00001c7f”, “__DAY=26”) in new stack
– Executing [s@sub-record-check:5] Set(“SIP/110002-00001c7f”, “__MONTH=01”) in new stack
– Executing [s@sub-record-check:6] Set(“SIP/110002-00001c7f”, “__YEAR=2018”) in new stack
– Executing [s@sub-record-check:7] Set(“SIP/110002-00001c7f”, “__TIMESTR=20180126-165116”) in new stack
– Executing [s@sub-record-check:8] Set(“SIP/110002-00001c7f”, “__FROMEXTEN=110002”) in new stack
– Executing [s@sub-record-check:9] Set(“SIP/110002-00001c7f”, “__MON_FMT=wav”) in new stack
– Executing [s@sub-record-check:10] NoOp(“SIP/110002-00001c7f”, “Recordings initialized”) in new stack
– Executing [s@sub-record-check:11] ExecIf(“SIP/110002-00001c7f”, “0?Set(ARG3=dontcare)”) in new stack
– Executing [s@sub-record-check:12] Set(“SIP/110002-00001c7f”, “REC_POLICY_MODE_SAVE=”) in new stack
– Executing [s@sub-record-check:13] ExecIf(“SIP/110002-00001c7f”, “0?Set(REC_STATUS=NO)”) in new stack
– Executing [s@sub-record-check:14] GotoIf(“SIP/110002-00001c7f”, “3?checkaction”) in new stack
– Goto (sub-record-check,s,17)
– Executing [s@sub-record-check:17] GotoIf(“SIP/110002-00001c7f”, “1?sub-record-check,out,1”) in new stack
– Goto (sub-record-check,out,1)
– Executing [out@sub-record-check:1] NoOp(“SIP/110002-00001c7f”, “Outbound Recording Check from 110002 to 0060146349060”) in new stack
– Executing [out@sub-record-check:2] Set(“SIP/110002-00001c7f”, “RECMODE=yes”) in new stack
– Executing [out@sub-record-check:3] ExecIf(“SIP/110002-00001c7f”, “0?Goto(routewins)”) in new stack
– Executing [out@sub-record-check:4] ExecIf(“SIP/110002-00001c7f”, “0?Goto(routewins)”) in new stack
– Executing [out@sub-record-check:5] Gosub(“SIP/110002-00001c7f”, “recordcheck,1(yes,out,0060146349060)”) in new stack
– Executing [recordcheck@sub-record-check:1] NoOp(“SIP/110002-00001c7f”, “Starting recording check against yes”) in new stack
– Executing [recordcheck@sub-record-check:2] Goto(“SIP/110002-00001c7f”, “yes”) in new stack
– Goto (sub-record-check,recordcheck,9)
– Executing [recordcheck@sub-record-check:9] ExecIf(“SIP/110002-00001c7f”, “0?Return()”) in new stack
– Executing [recordcheck@sub-record-check:10] Set(“SIP/110002-00001c7f”, “__REC_POLICY_MODE=YES”) in new stack
– Executing [recordcheck@sub-record-check:11] Goto(“SIP/110002-00001c7f”, “startrec”) in new stack
– Goto (sub-record-check,recordcheck,16)
– Executing [recordcheck@sub-record-check:16] NoOp(“SIP/110002-00001c7f”, “Starting recording: out, 0060146349060”) in new stack
– Executing [recordcheck@sub-record-check:17] Set(“SIP/110002-00001c7f”, “AUDIOHOOK_INHERIT(MixMonitor)=yes”) in new stack
– Executing [recordcheck@sub-record-check:18] Set(“SIP/110002-00001c7f”, “__CALLFILENAME=out-0060146349060-110002-20180126-165116-1516956676.7297”) in new stack
– Executing [recordcheck@sub-record-check:19] MixMonitor(“SIP/110002-00001c7f”, “2018/01/26/out-0060146349060-110002-20180126-165116-1516956676.7297.wav,abi(LOCAL_MIXMON_ID),”) in new stack
== Begin MixMonitor Recording SIP/110002-00001c7f
– Executing [recordcheck@sub-record-check:20] Set(“SIP/110002-00001c7f”, “__MIXMON_ID=0xb5d0c3a0”) in new stack
– Executing [recordcheck@sub-record-check:21] Set(“SIP/110002-00001c7f”, “__RECORD_ID=SIP/110002-00001c7f”) in new stack
– Executing [recordcheck@sub-record-check:22] Set(“SIP/110002-00001c7f”, “__REC_STATUS=RECORDING”) in new stack
– Executing [recordcheck@sub-record-check:23] Set(“SIP/110002-00001c7f”, “CDR(recordingfile)=out-0060146349060-110002-20180126-165116-1516956676.7297.wav”) in new stack
– Executing [recordcheck@sub-record-check:24] Return(“SIP/110002-00001c7f”, “”) in new stack
– Executing [out@sub-record-check:6] Return(“SIP/110002-00001c7f”, “”) in new stack
– Executing [0060146349060@from-internal:3] ExecIf(“SIP/110002-00001c7f”, “0 ?Set(CDR(accountcode)=)”) in new stack
– Executing [0060146349060@from-internal:4] Set(“SIP/110002-00001c7f”, “MOHCLASS=default”) in new stack
– Executing [0060146349060@from-internal:5] Set(“SIP/110002-00001c7f”, “_NODEST=”) in new stack
– Executing [0060146349060@from-internal:6] Macro(“SIP/110002-00001c7f”, “dialout-trunk,7,9990160146349060,on”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/110002-00001c7f”, “DIAL_TRUNK=7”) in new stack
– Executing [s@macro-dialout-trunk:2] GosubIf(“SIP/110002-00001c7f”, “0?sub-pincheck,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:3] GotoIf(“SIP/110002-00001c7f”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:4] Set(“SIP/110002-00001c7f”, “DIAL_NUMBER=9990160146349060”) in new stack
– Executing [s@macro-dialout-trunk:5] Set(“SIP/110002-00001c7f”, “DIAL_TRUNK_OPTIONS=Ttr”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/110002-00001c7f”, “OUTBOUND_GROUP=OUT_7”) in new stack
– Executing [s@macro-dialout-trunk:7] Set(“SIP/110002-00001c7f”, “DIAL_TRUNK_OPTIONS=Ttr”) in new stack
– Executing [s@macro-dialout-trunk:8] GotoIf(“SIP/110002-00001c7f”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,10)
– Executing [s@macro-dialout-trunk:10] GotoIf(“SIP/110002-00001c7f”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:11] Macro(“SIP/110002-00001c7f”, “outbound-callerid,7”) in new stack
– Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/110002-00001c7f”, “0?Set(CALLERPRES(name-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:2] ExecIf(“SIP/110002-00001c7f”, “0?Set(CALLERPRES(num-pres)=)”) in new stack
– Executing [s@macro-outbound-callerid:3] ExecIf(“SIP/110002-00001c7f”, “0?Set(REALCALLERIDNUM=110002)”) in new stack
– Executing [s@macro-outbound-callerid:4] GotoIf(“SIP/110002-00001c7f”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,7)
– Executing [s@macro-outbound-callerid:7] Set(“SIP/110002-00001c7f”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:8] Set(“SIP/110002-00001c7f”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:9] Set(“SIP/110002-00001c7f”, “TRUNKOUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:10] GotoIf(“SIP/110002-00001c7f”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,15)
– Executing [s@macro-outbound-callerid:15] ExecIf(“SIP/110002-00001c7f”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:16] ExecIf(“SIP/110002-00001c7f”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:17] ExecIf(“SIP/110002-00001c7f”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:18] ExecIf(“SIP/110002-00001c7f”, “0?Set(CALLERPRES(name-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:19] ExecIf(“SIP/110002-00001c7f”, “0?Set(CALLERPRES(num-pres)=prohib_passed_screen)”) in new stack
– Executing [s@macro-outbound-callerid:20] Set(“SIP/110002-00001c7f”, “CDR(outbound_cnum)=110002”) in new stack
– Executing [s@macro-outbound-callerid:21] Set(“SIP/110002-00001c7f”, “CDR(outbound_cnam)=110002”) in new stack
– Executing [s@macro-dialout-trunk:12] GosubIf(“SIP/110002-00001c7f”, “0?sub-flp-7,s,1()”) in new stack
– Executing [s@macro-dialout-trunk:13] Set(“SIP/110002-00001c7f”, “OUTNUM=9990160146349060”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“SIP/110002-00001c7f”, “custom=SIP/Superunity_Server”) in new stack
– Executing [s@macro-dialout-trunk:15] ExecIf(“SIP/110002-00001c7f”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Ttr)”) in new stack
– Executing [s@macro-dialout-trunk:16] ExecIf(“SIP/110002-00001c7f”, “0?Set(DIAL_TRUNK_OPTIONS=TtrM(confirm))”) in new stack
– Executing [s@macro-dialout-trunk:17] Macro(“SIP/110002-00001c7f”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/110002-00001c7f”, “”) in new stack
– Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/110002-00001c7f”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:19] ExecIf(“SIP/110002-00001c7f”, “1?Set(CONNECTEDLINE(num,i)=9990160146349060)”) in new stack
– Executing [s@macro-dialout-trunk:20] ExecIf(“SIP/110002-00001c7f”, “1?Set(CONNECTEDLINE(name,i)=CID:110002)”) in new stack
– Executing [s@macro-dialout-trunk:21] ExecIf(“SIP/110002-00001c7f”, “0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)110002)”) in new stack
– Executing [s@macro-dialout-trunk:22] GotoIf(“SIP/110002-00001c7f”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:23] Dial(“SIP/110002-00001c7f”, “SIP/Superunity_Server/9990160146349060,300,Ttr”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/Superunity_Server/9990160146349060

<— Transmitting (no NAT) to 192.168.8.1:65069 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.8.1:65069;branch=z9hG4bK001acebce300e811ba5fb0a0553453f5;received=192.168.8.1;rport=65069
From: “PhonerLite” sip:[email protected];tag=4209725802
To: sip:[email protected];tag=as54308621
Call-ID: [email protected]
CSeq: 8 INVITE
Server: FPBX-13.0.192.19(11.23.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Length: 0

<------------>
– SIP/Superunity_Server-00001c80 is ringing

<— Transmitting (no NAT) to 192.168.8.1:65069 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.8.1:65069;branch=z9hG4bK001acebce300e811ba5fb0a0553453f5;received=192.168.8.1;rport=65069
From: “PhonerLite” sip:[email protected];tag=4209725802
To: sip:[email protected];tag=as54308621
Call-ID: [email protected]
CSeq: 8 INVITE
Server: FPBX-13.0.192.19(11.23.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
Content-Length: 0

<------------>
– SIP/Superunity_Server-00001c80 answered SIP/110002-00001c7f
Audio is at 21924
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 192.168.8.1:65069 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.1:65069;branch=z9hG4bK001acebce300e811ba5fb0a0553453f5;received=192.168.8.1;rport=65069
From: “PhonerLite” sip:[email protected];tag=4209725802
To: sip:[email protected];tag=as54308621
Call-ID: [email protected]
CSeq: 8 INVITE
Server: FPBX-13.0.192.19(11.23.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: sip:[email protected]:5060
P-Asserted-Identity: “CID:110002” sip:[email protected]
Content-Type: application/sdp
Require: timer
Content-Length: 260

v=0
o=root 1823788812 1823788812 IN IP4 192.168.8.15
s=Asterisk PBX 11.23.1
c=IN IP4 192.168.8.15
t=0 0
m=audio 21924 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

cynjut · January 26, 2018, 3:07pm

A couple of things to look for:

Using PJ-SIP and VPN connections where the connections are in different subnets can be problematic. I doubt that’s it, but it is the one that was in my head last.
There are router settings, session timers specifically, that can fade out if you don’t have the timeouts in the extensions/trunk set correctly. Off the top of my head, I’m not sure which ones to start with, but I think this is a more likely rabbit-hole to dive down.

Ezra5447 · January 29, 2018, 1:07am

Hi cynjut, Thanks for your feedback, will try based on your given points.

Ezra5447 · February 12, 2018, 2:11am

Hi, finally this issue has been solved;

Solution: added a new policy in Firewall - internal to VPN-ssl , previously only added the VPN user to asterisk server.

but weird thing is previously we was using Welltech instead of asterisk and we only added 1 policy(VPN user to asterisk server.) and its working fine.

Anyways, Thanks.

system · February 12, 2019, 2:12am

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