T.38 Issues

rcraig114 · December 26, 2012, 1:11am

I’ve enabled T.38 passthru in the Asterisk Sip Settings and enabled it on the inbound route. Faxes are failing in both directions. Outgoing dials out, switches to T.38 as indicated by SIP logs, the distant end picks up, but I never hear the CNG tones from the distant end. My fax machine is sitting next to me. I can hear the tones being sent out by my end, but I never hear the distant end tones. I would imagine that it’s the same problem for receiving faxes. I’ve gone back and forth with NAT settings thinking it was between the adapter and FreePBX (all sitting in the same closet locally) with no change. I’ve pasted the debugs below. If anyone see’s anything, please let me know as I’m pulling my hair out on this one.

Caller: 15203509672
Called: 14792460547

=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2012.12.25 16:15:53 =~=~=~=~=~=~=~=~=~=~=~=
udptl set debug on

localhost*CLI>
e[0KUDPTL Debugging Enabled

e[Klocalhost*CLI> sip set debug on

localhost*CLI>
e[0KSIP Debugging enabled

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.3.108:5060 —>

<------------->

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.1.63:5062 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-580aeb17
From: “15203509672” sip:[email protected];tag=75b5948ecef18203o0
To: sip:[email protected]
Remote-Party-ID: “15203509672” sip:[email protected];screen=yes;party=calling
Call-ID: [email protected]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: “15203509672” sip:[email protected]:5062
Expires: 240
User-Agent: Cisco/SPA122-1.0.2(006)

e[Klocalhost*CLI>
e[0KContent-Length: 277
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp

v=0
o=- 9865914 9865914 IN IP4 192.168.1.63
s=-
c=IN IP4 192.168.1.63
t=0 0
m=audio 16474 RTP/AVP 0 18 2 8 100
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729a/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=ptime:30
a=sendrecv
<------------->
— (15 headers 14 lines) —
Sending to 192.168.1.63:5062 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘12502’ for ‘12502’ from 192.168.1.63:5062

<— Reliably Transmitting (NAT) to 192.168.1.63:5062 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-580aeb17;received=192.168.1.63;rport=5062

From: “15203509672” sip:[email protected];tag=75b5948ecef18203o0

To: sip:[email protected];tag=as571cf21d

Call-ID: [email protected]

CSeq: 101 INVITE

Server: FPBX-2.10.1(1.8.18.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“495b805a”

Content-Length: 0

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

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.1.63:5062 —>
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-580aeb17
From: “15203509672” sip:[email protected];tag=75b5948ecef18203o0
To: sip:[email protected];tag=as571cf21d
Call-ID: [email protected]
CSeq: 101 ACK
Max-Forwards: 70
Contact: “15203509672” sip:[email protected]:5062
User-Agent: Cisco/SPA122-1.0.2(006)
Content-Length: 0

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

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.1.63:5062 —>
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-3e240b0b
From: “15203509672” sip:[email protected];tag=75b5948ecef18203o0
To: sip:[email protected]
Remote-Party-ID: “15203509672” sip:[email protected];screen=yes;party=calling
Call-ID: [email protected]
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username=“12502”,realm=“asterisk”,nonce=“495b805a”,uri="sip:[email protected]",algorithm=MD5,response="5e6a6f88cf5b9f53576c7aac43e64045"
Contact: “15203509672” sip:[email protected]:5062
Expires: 240
User-Agent: Cisco/SPA122-1.0.2(006)
Content-Length: 277
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp

v=0
o=- 9865914 9865914 IN IP4 192.168.1.63
s=-
c=IN IP4 192.168.1.63
t=0 0
m=audio 16474 RTP/AVP 0 18 2 8 100
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729a/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=ptime:30
a=sendrecv
<------------->
— (16 headers 14 lines) —
Sending to 192.168.1.63:5062 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘12502’ for ‘12502’ from 192.168.1.63:5062
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 8
Found RTP audio format 100
Found audio description format PCMU for ID 0
Found audio description format G729a for ID 18
Found audio description format G726-32 for ID 2
Found audio description format PCMA for ID 8
Found unknown media description format NSE for ID 100
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x90c (ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.1.63:16474

e[Klocalhost*CLI>
e[0KLooking for 14792460547 in from-internal (domain 192.168.4.10)

e[Klocalhost*CLI>
e[0Klist_route: hop: sip:[email protected]:5062

e[Klocalhost*CLI>
e[0K
<— Transmitting (NAT) to 192.168.1.63:5062 —>
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-3e240b0b;received=192.168.1.63;rport=5062

From: “15203509672” sip:[email protected];tag=75b5948ecef18203o0

To: sip:[email protected]

Call-ID: [email protected]

CSeq: 102 INVITE

Server: FPBX-2.10.1(1.8.18.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Contact: sip:[email protected]:5060

Content-Length: 0

<------------>

e[Klocalhost*CLI>
e[0K – Executing [14792460547@from-internal:1] e[1;36mMacroe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35muser-callerid,LIMIT,e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:1] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mAMPUSER=12502e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:2] e[1;36mGotoIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?reporte[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:3] e[1;36mExecIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m1?Set(REALCALLERIDNUM=12502)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:4] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mAMPUSER=12502e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:5] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mAMPUSERCIDNAME=Craig-Fax-ATAe[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:6] e[1;36mGotoIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?reporte[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:7] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mAMPUSERCID=12502e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:8] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mCALLERID(all)=“Craig-Fax-ATA” <12502>e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:9] e[1;36mGotoIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?limite[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:10] e[1;36mExecIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m1?Set(GROUP(concurrency_limit)=12502)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:11] e[1;36mExecIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?Set(CHANNEL(language)=)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:12] e[1;36mGosubIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m7?sub-ccss,s,1(from-internal,14792460547)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@sub-ccss:1] e[1;36mExecIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?Return()e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@sub-ccss:2] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mCCSS_SETUP=TRUEe[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@sub-ccss:3] e[1;36mGosubIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?monitor_config,1(from-internal,14792460547):monitor_default,1(from-internal,14792460547)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [monitor_default@sub-ccss:1] e[1;36mGotoIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?is_extene[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [monitor_default@sub-ccss:2] e[1;36mStackPope[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35me[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [monitor_default@sub-ccss:3] e[1;36mReturne[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mFALSEe[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:13] e[1;36mGotoIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m1?continuee[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Goto (macro-user-callerid,s,26)

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:26] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mCALLERID(number)=12502e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:27] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mCALLERID(name)=Craig-Fax-ATAe[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-user-callerid:28] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mCHANNEL(language)=ene[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [14792460547@from-internal:2] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mMOHCLASS=defaulte[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [14792460547@from-internal:3] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m_NODEST=e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [14792460547@from-internal:4] e[1;36mGosube[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35msub-record-check,s,1(out,14792460547,)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@sub-record-check:1] e[1;36mGotoIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m1?checke[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Goto (sub-record-check,s,6)

e[Klocalhost*CLI>
e[0K – Executing [s@sub-record-check:6] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m__MON_FMT=wave[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@sub-record-check:7] e[1;36mGotoIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m1?nexte[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Goto (sub-record-check,s,10)

e[Klocalhost*CLI>
e[0K – Executing [s@sub-record-check:10] e[1;36mExecIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?Return()e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@sub-record-check:11] e[1;36mGotoIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?out,1e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@sub-record-check:12] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m__REC_STATUS=INITIALIZEDe[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@sub-record-check:13] e[1;36mExecIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?Set(__REC_POLICY_MODE=)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@sub-record-check:14] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mNOW=1356477376e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@sub-record-check:15] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m__DAY=25e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@sub-record-check:16] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m__MONTH=12e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@sub-record-check:17] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m__YEAR=2012e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@sub-record-check:18] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m__TIMESTR=20121225-161616e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@sub-record-check:19] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m__FROMEXTEN=12502e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@sub-record-check:20] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m__CALLFILENAME=out-14792460547-12502-20121225-161616-1356477376.770e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@sub-record-check:21] e[1;36mGotoe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mout,1e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Goto (sub-record-check,out,1)

e[Klocalhost*CLI>
e[0K – Executing [out@sub-record-check:1] e[1;36mExecIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m1?Set(__REC_POLICY_MODE=always)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [out@sub-record-check:2] e[1;36mGosubIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m1?record,1(exten,14792460547,12502)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [record@sub-record-check:1] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mAUDIOHOOK_INHERIT(MixMonitor)=yese[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [record@sub-record-check:2] e[1;36mMixMonitore[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m2012/12/25/out-14792460547-12502-20121225-161616-1356477376.770.wav,e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [record@sub-record-check:3] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m__REC_STATUS=RECORDINGe[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [record@sub-record-check:4] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mCDR(recordingfile)=out-14792460547-12502-20121225-161616-1356477376.770.wave[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [record@sub-record-check:5] e[1;36mReturne[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35me[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [out@sub-record-check:3] e[1;36mReturne[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35me[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [14792460547@from-internal:5] e[1;36mMacroe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mdialout-trunk,2,14792460547,e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:1] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mDIAL_TRUNK=2e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:2] e[1;36mGosubIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?sub-pincheck,s,1()e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:3] e[1;36mGotoIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?disabletrunk,1e[0m”) in new stack

e[Klocalhost*CLI>
e[0K == Begin MixMonitor Recording SIP/12502-00000302

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:4] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mDIAL_NUMBER=14792460547e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:5] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mDIAL_TRUNK_OPTIONS=tre[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:6] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mOUTBOUND_GROUP=OUT_2e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:7] e[1;36mGotoIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m1?nomaxe[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Goto (macro-dialout-trunk,s,9)

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:9] e[1;36mGotoIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?skipoutcide[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:10] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mDIAL_TRUNK_OPTIONS=e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:11] e[1;36mMacroe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35moutbound-callerid,2e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:1] e[1;36mExecIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?Set(CALLERPRES()=)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:2] e[1;36mExecIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?Set(REALCALLERIDNUM=12502)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:3] e[1;36mGotoIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m1?normcide[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Goto (macro-outbound-callerid,s,6)

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:6] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mUSEROUTCID=15203509672e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:7] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mEMERGENCYCID=e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:8] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mTRUNKOUTCID=e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:9] e[1;36mGotoIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m1?trunkcide[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Goto (macro-outbound-callerid,s,12)

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:12] e[1;36mExecIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?Set(CALLERID(all)=)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:13] e[1;36mExecIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m1?Set(CALLERID(all)=15203509672)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:14] e[1;36mExecIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?Set(CALLERID(all)=)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-outbound-callerid:15] e[1;36mExecIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?Set(CALLERPRES()=prohib_passed_screen)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:12] e[1;36mGosubIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?sub-flp-2,s,1()e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:13] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mOUTNUM=14792460547e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:14] e[1;36mSete[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mcustom=SIP/Flowroutee[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:15] e[1;36mExecIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:16] e[1;36mExecIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?Set(DIAL_TRUNK_OPTIONS=M(confirm))e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:17] e[1;36mMacroe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mdialout-trunk-predial-hook,e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk-predial-hook:1] e[1;36mMacroExite[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35me[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:18] e[1;36mGotoIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?bypass,1e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:19] e[1;36mExecIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m1?Set(CONNECTEDLINE(num,i)=14792460547)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:20] e[1;36mExecIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m1?Set(CONNECTEDLINE(name,i)=CID:15203509672)e[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:21] e[1;36mGotoIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?customtrunke[0m”) in new stack

e[Klocalhost*CLI>
e[0K – Executing [s@macro-dialout-trunk:22] e[1;36mDiale[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mSIP/Flowroute/14792460547,300,e[0m”) in new stack

e[Klocalhost*CLI>
e[0K == Using SIP RTP TOS bits 184

e[Klocalhost*CLI>
e[0K == Using SIP RTP CoS mark 5

e[Klocalhost*CLI>
e[0KAudio is at 16470

e[Klocalhost*CLI>
e[0KAdding codec 0x4 (ulaw) to SDP

e[Klocalhost*CLI>
e[0KAdding codec 0x8 (alaw) to SDP

e[Klocalhost*CLI>
e[0KAdding codec 0x2 (gsm) to SDP

e[Klocalhost*CLI>
e[0KAdding non-codec 0x1 (telephone-event) to SDP

e[Klocalhost*CLI>
e[0KReliably Transmitting (NAT) to 216.115.69.144:5060:
INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK27ea5ca8;rport

Max-Forwards: 70

From: “15203509672” sip:[email protected];tag=as516769f6

To: sip:[email protected]

Contact: sip:[email protected]:5060

Call-ID: [email protected]

CSeq: 102 INVITE

User-Agent: FPBX-2.10.1(1.8.18.0)

Date: Tue, 25 Dec 2012 23:16:16 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 284

v=0

o=root 1676371789 1676371789 IN IP4 192.168.4.10

s=Asterisk PBX 1.8.18.0

c=IN IP4 192.168.4.10

t=0 0

m=audio 16470 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

e[Klocalhost*CLI>
e[0K – Called SIP/Flowroute/14792460547

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:216.115.69.144:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK27ea5ca8;rport=1414
From: “15203509672” sip:[email protected];tag=as516769f6
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 102 INVITE
Content-Length: 0

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

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:216.115.69.144:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK27ea5ca8;rport=1414
From: “15203509672” sip:[email protected];tag=as516769f6
To: sip:[email protected];tag=da160a8e0534d7b0db4afd8620a22cb7.06f8
Call-ID: [email protected]
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm=“sip.flowroute.com”, nonce=“UNo07FDaM8AfB4Lu1acUFHticjdM/9dX”, qop="auth"
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Transmitting (NAT) to 216.115.69.144:5060:
ACK sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK27ea5ca8;rport

Max-Forwards: 70

From: “15203509672” sip:[email protected];tag=as516769f6

To: sip:[email protected];tag=da160a8e0534d7b0db4afd8620a22cb7.06f8

Contact: sip:[email protected]:5060

Call-ID: [email protected]

CSeq: 102 ACK

User-Agent: FPBX-2.10.1(1.8.18.0)

Content-Length: 0

Audio is at 16470
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 216.115.69.144:5060:
INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK2898c9c8;rport

Max-Forwards: 70

From: “15203509672” sip:[email protected];tag=as516769f6

To: sip:[email protected]

Contact: sip:[email protected]:5060

Call-ID: [email protected]

CSeq: 103 INVITE

User-Agent: FPBX-2.10.1(1.8.18.0)

Proxy-Authorization: Digest username=“08922051”, realm=“sip.flowroute.com”, algorithm=MD5, uri="sip:[email protected]", nonce=“UNo07FDaM8AfB4Lu1acUFHticjdM/9dX”, response=“c58c585633db3613b00fe1d3f69d3f23”, qop=auth, cnonce=“3e50900e”, nc=00000001

Date: Tue, 25 Dec 2012 23:16:16 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 284

v=0

o=root 1676371789 1676371790 IN IP4 192.168.4.10

s=Asterisk PBX 1.8.18.0

c=IN IP4 192.168.4.10

t=0 0

m=audio 16470 RTP/AVP 0 8 3 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:3 GSM/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:216.115.69.144:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK2898c9c8;rport=1414
From: “15203509672” sip:[email protected];tag=as516769f6
To: sip:[email protected]
Call-ID: [email protected]
CSeq: 103 INVITE
Content-Length: 0

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

e[Klocalhost*CLI>
e[0KReally destroying SIP dialog ‘[email protected]’ Method: REGISTER

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:216.115.69.144:5060 —>
OPTIONS sip:72.201.112.131:1414 SIP/2.0
Max-Forwards: 10
Record-Route: sip:216.115.69.144;lr
Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bKb716.eaa2166bfa822b4abaec402f03c70906.0
Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0
Route: sip:216.115.69.144;lr;received="sip:72.201.112.131:1414"
From: sip:ping@invalid;tag=bec19526
To: sip:72.201.112.131:1414
Call-ID: [email protected]
CSeq: 1 OPTIONS
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Looking for s in from-sip-external (domain 72.201.112.131)

<— Transmitting (NAT) to 216.115.69.144:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bKb716.eaa2166bfa822b4abaec402f03c70906.0;received=216.115.69.144;rport=5060

Via: SIP/2.0/UDP 216.115.69.131:5060;branch=0

Record-Route: sip:216.115.69.144;lr

From: sip:ping@invalid;tag=bec19526

To: sip:72.201.112.131:1414;tag=as3ce857d4

Call-ID: [email protected]

CSeq: 1 OPTIONS

Server: FPBX-2.10.1(1.8.18.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Contact: sip:192.168.4.10:5060

Accept: application/sdp

Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: OPTIONS)

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:216.115.69.144:5060 —>
SIP/2.0 180 Ringing
From: “15203509672” sip:[email protected];tag=as516769f6
To: sip:[email protected];tag=SDvakj599-gK0defbcb0
Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK2898c9c8;rport=1414
Call-ID: [email protected]
CSeq: 103 INVITE
Record-Route: sip:216.115.69.133;lr
Record-Route: sip:216.115.69.144;lr
Contact: sip:[email protected]:5060;transport=udp
Content-Length: 173
Content-Type: application/sdp

v=0
o=- 12704 25219 IN IP4 67.16.125.60
s=-
c=IN IP4 67.16.125.60
t=0 0
m=audio 53562 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20
<------------->
— (11 headers 9 lines) —
list_route: hop: sip:216.115.69.144;lr
list_route: hop: sip:216.115.69.133;lr
Found RTP audio format 0
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 67.16.125.60:53562

e[Klocalhost*CLI>
e[0K – SIP/Flowroute-00000303 is ringing

<— Transmitting (NAT) to 192.168.1.63:5062 —>
SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-3e240b0b;received=192.168.1.63;rport=5062

From: “15203509672” sip:[email protected];tag=75b5948ecef18203o0

To: sip:[email protected];tag=as1566990a

Call-ID: [email protected]

CSeq: 102 INVITE

Server: FPBX-2.10.1(1.8.18.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Contact: sip:[email protected]:5060

Content-Length: 0

<------------>
– SIP/Flowroute-00000303 is making progress passing it to SIP/12502-00000302
Audio is at 19070
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP

<— Transmitting (NAT) to 192.168.1.63:5062 —>
SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-3e240b0b;received=192.168.1.63;rport=5062

From: “15203509672” sip:[email protected];tag=75b5948ecef18203o0

To: sip:[email protected];tag=as1566990a

Call-ID: [email protected]

CSeq: 102 INVITE

Server: FPBX-2.10.1(1.8.18.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Contact: sip:[email protected]:5060

Content-Type: application/sdp

Content-Length: 252

v=0

o=root 1639512434 1639512434 IN IP4 192.168.4.10

s=Asterisk PBX 1.8.18.0

c=IN IP4 192.168.4.10

t=0 0

m=audio 19070 RTP/AVP 0 8 18

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=ptime:20

a=sendrecv

<------------>

e[Klocalhost*CLI>
e[0KReliably Transmitting (no NAT) to 192.168.0.5:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK29d446ac

Max-Forwards: 70

From: “Unknown” sip:[email protected];tag=as51eaba0e

To: sip:[email protected]:5060

Contact: sip:[email protected]:5060

Call-ID: [email protected]:5060

CSeq: 102 OPTIONS

User-Agent: FPBX-2.10.1(1.8.18.0)

Date: Tue, 25 Dec 2012 23:16:20 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.0.5:50830 —>
REGISTER sip:192.168.4.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK1DE3D4E
From: sip:[email protected];tag=1F5C5134-156C
To: sip:[email protected]
Date: Tue, 25 Dec 2012 23:16:20 GMT
Call-ID: 5AFE03B2-4D3311E2-85A1B464-AFA4AC95
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1356477380
CSeq: 3313 REGISTER
Contact: sip:[email protected]:5060
Expires: 60
Supported: path
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to 192.168.0.5:50830 (NAT)

<— Transmitting (no NAT) to 192.168.0.5:5060 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK1DE3D4E;received=192.168.0.5

From: sip:[email protected];tag=1F5C5134-156C

To: sip:[email protected];tag=as2e343792

Call-ID: 5AFE03B2-4D3311E2-85A1B464-AFA4AC95

CSeq: 3313 REGISTER

Server: FPBX-2.10.1(1.8.18.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“1a3de9cf”

Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘5AFE03B2-4D3311E2-85A1B464-AFA4AC95’ in 32000 ms (Method: REGISTER)

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.0.5:50830 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK29d446ac
From: “Unknown” sip:[email protected];tag=as51eaba0e
To: sip:[email protected]:5060;tag=1F5C5130-2122
Date: Tue, 25 Dec 2012 23:16:20 GMT
Call-ID: [email protected]:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 447

v=0
o=CiscoSystemsSIP-GW-UserAgent 696 9759 IN IP4 192.168.0.5
s=SIP Call
c=IN IP4 192.168.0.1
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 192.168.0.1
m=image 0 udptl t38
c=IN IP4 192.168.0.1
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
— (14 headers 18 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.0.5:50830 —>
REGISTER sip:192.168.4.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK1DE418BB
From: sip:[email protected];tag=1F5C5134-156C
To: sip:[email protected]
Date: Tue, 25 Dec 2012 23:16:20 GMT
Call-ID: 5AFE03B2-4D3311E2-85A1B464-AFA4AC95
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1356477380
CSeq: 3314 REGISTER
Contact: sip:[email protected]:5060
Expires: 60
Authorization: Digest username=“12501”,realm=“asterisk”,uri=“sip:192.168.4.10:5060”,response=“6596cc0860b607d84ff181383f6a9f1d”,nonce=“1a3de9cf”,algorithm=MD5
Content-Length: 0

<------------->

e[Klocalhost*CLI>
e[0K— (14 headers 0 lines) —

e[Klocalhost*CLI>
e[0KSending to 192.168.0.5:5060 (no NAT)

e[Klocalhost*CLI>
e[0KReliably Transmitting (no NAT) to 192.168.0.5:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK2861e3a7

Max-Forwards: 70

From: “Unknown” sip:[email protected];tag=as5e684d60

To: sip:[email protected]:5060

Contact: sip:[email protected]:5060

Call-ID: [email protected]:5060

CSeq: 102 OPTIONS

User-Agent: FPBX-2.10.1(1.8.18.0)

Date: Tue, 25 Dec 2012 23:16:20 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0

e[Klocalhost*CLI>
e[0K
<— Transmitting (no NAT) to 192.168.0.5:5060 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK1DE418BB;received=192.168.0.5

From: sip:[email protected];tag=1F5C5134-156C

To: sip:[email protected];tag=as2e343792

Call-ID: 5AFE03B2-4D3311E2-85A1B464-AFA4AC95

CSeq: 3314 REGISTER

Server: FPBX-2.10.1(1.8.18.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Expires: 60

Contact: sip:[email protected]:5060;expires=60

Date: Tue, 25 Dec 2012 23:16:20 GMT

Content-Length: 0

<------------>

e[Klocalhost*CLI>
e[0KScheduling destruction of SIP dialog ‘5AFE03B2-4D3311E2-85A1B464-AFA4AC95’ in 32000 ms (Method: REGISTER)

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.0.5:50830 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK2861e3a7
From: “Unknown” sip:[email protected];tag=as5e684d60
To: sip:[email protected]:5060;tag=1F5C5150-20A5
Date: Tue, 25 Dec 2012 23:16:20 GMT
Call-ID: [email protected]:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 448

v=0
o=CiscoSystemsSIP-GW-UserAgent 4645 3000 IN IP4 192.168.0.5
s=SIP Call
c=IN IP4 192.168.0.1
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15 3
c=IN IP4 192.168.0.1
m=image 0 udptl t38
c=IN IP4 192.168.0.1
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy
<------------->

e[Klocalhost*CLI>
e[0K— (14 headers 18 lines) —

e[Klocalhost*CLI>
e[0KReally destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

e[Klocalhost*CLI>
e[0KReliably Transmitting (no NAT) to 192.168.3.108:5060:
OPTIONS sip:[email protected]:5060;rinstance=ec0c42b493a8fe80 SIP/2.0

Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK5294bd29

Max-Forwards: 70

From: “Unknown” sip:[email protected];tag=as59bad90b

To: sip:[email protected]:5060;rinstance=ec0c42b493a8fe80

Contact: sip:[email protected]:5060

Call-ID: [email protected]:5060

CSeq: 102 OPTIONS

User-Agent: FPBX-2.10.1(1.8.18.0)

Date: Tue, 25 Dec 2012 23:16:26 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.3.108:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK5294bd29
Contact: sip:192.168.3.108:5060
To: sip:[email protected]:5060;rinstance=ec0c42b493a8fe80;tag=b9d11064
From: "Unknown"sip:[email protected];tag=as59bad90b
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces
User-Agent: X-Lite release 5.0.0 stamp 67284
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

e[Klocalhost*CLI>
e[0KReally destroying SIP dialog ‘6762004e62d4cde13af866ca36e69df0@[::1]’ Method: REGISTER

e[Klocalhost*CLI>
e[0KReally destroying SIP dialog ‘[email protected]’ Method: OPTIONS

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:216.115.69.144:5060 —>
SIP/2.0 200 OK
From: “15203509672” sip:[email protected];tag=as516769f6
To: sip:[email protected];tag=SDvakj599-gK0defbcb0
Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK2898c9c8;rport=1414
Call-ID: [email protected]
CSeq: 103 INVITE
Record-Route: sip:216.115.69.133;lr
Record-Route: sip:216.115.69.144;lr
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: sip:[email protected]:5060;transport=udp
Session-Expires: 10800;refresher=uas
Content-Length: 173
Content-Type: application/sdp

v=0
o=- 12704 25219 IN IP4 67.16.125.60
s=-
c=IN IP4 67.16.125.60
t=0 0
m=audio 53562 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:20
<------------->
— (13 headers 9 lines) —
list_route: hop: sip:216.115.69.144;lr
list_route: hop: sip:216.115.69.133;lr
set_destination: Parsing sip:216.115.69.144;lr for address/port to send to
set_destination: set destination to 216.115.69.144:5060
Transmitting (NAT) to 216.115.69.144:5060:
ACK sip:[email protected]:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK4c20e532;rport

Route: sip:216.115.69.144;lr,sip:216.115.69.133;lr

Max-Forwards: 70

From: “15203509672” sip:[email protected];tag=as516769f6

To: sip:[email protected];tag=SDvakj599-gK0defbcb0

Contact: sip:[email protected]:5060

Call-ID: [email protected]

CSeq: 103 ACK

User-Agent: FPBX-2.10.1(1.8.18.0)

Content-Length: 0

e[Klocalhost*CLI>
e[0K – SIP/Flowroute-00000303 answered SIP/12502-00000302

e[Klocalhost*CLI>
e[0KAudio is at 19070

e[Klocalhost*CLI>
e[0KAdding codec 0x4 (ulaw) to SDP

e[Klocalhost*CLI>
e[0KAdding codec 0x8 (alaw) to SDP

e[Klocalhost*CLI>
e[0KAdding codec 0x100 (g729) to SDP

e[Klocalhost*CLI>
e[0K
<— Reliably Transmitting (NAT) to 192.168.1.63:5062 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-3e240b0b;received=192.168.1.63;rport=5062

From: “15203509672” sip:[email protected];tag=75b5948ecef18203o0

To: sip:[email protected];tag=as1566990a

Call-ID: [email protected]

CSeq: 102 INVITE

Server: FPBX-2.10.1(1.8.18.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Contact: sip:[email protected]:5060

Content-Type: application/sdp

Content-Length: 252

v=0

o=root 1639512434 1639512435 IN IP4 192.168.4.10

s=Asterisk PBX 1.8.18.0

c=IN IP4 192.168.4.10

t=0 0

m=audio 19070 RTP/AVP 0 8 18

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=ptime:20

a=sendrecv

<------------>

e[Klocalhost*CLI>
e[0KRetransmitting #1 (NAT) to 192.168.1.63:5062:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-3e240b0b;received=192.168.1.63;rport=5062

From: “15203509672” sip:[email protected];tag=75b5948ecef18203o0

To: sip:[email protected];tag=as1566990a

Call-ID: [email protected]

CSeq: 102 INVITE

Server: FPBX-2.10.1(1.8.18.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Contact: sip:[email protected]:5060

Content-Type: application/sdp

Content-Length: 252

v=0

o=root 1639512434 1639512435 IN IP4 192.168.4.10

s=Asterisk PBX 1.8.18.0

c=IN IP4 192.168.4.10

t=0 0

m=audio 19070 RTP/AVP 0 8 18

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=ptime:20

a=sendrecv

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.1.63:5062 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-542ebb3b
From: “15203509672” sip:[email protected];tag=75b5948ecef18203o0
To: sip:[email protected];tag=as1566990a
Call-ID: [email protected]
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username=“12502”,realm=“asterisk”,nonce=“495b805a”,uri="sip:[email protected]",algorithm=MD5,response="5e6a6f88cf5b9f53576c7aac43e64045"
Contact: “15203509672” sip:[email protected]:5062
User-Agent: Cisco/SPA122-1.0.2(006)
Content-Length: 0

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

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.1.63:5062 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-542ebb3b
From: “15203509672” sip:[email protected];tag=75b5948ecef18203o0
To: sip:[email protected];tag=as1566990a
Call-ID: [email protected]
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username=“12502”,realm=“asterisk”,nonce=“495b805a”,uri="sip:[email protected]",algorithm=MD5,response="5e6a6f88cf5b9f53576c7aac43e64045"
Contact: “15203509672” sip:[email protected]:5062
User-Agent: Cisco/SPA122-1.0.2(006)
Content-Length: 0

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

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.1.63:5062 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-ce5048df
From: “15203509672” sip:[email protected];tag=75b5948ecef18203o0
To: sip:[email protected];tag=as1566990a
Remote-Party-ID: “15203509672” sip:[email protected];screen=yes;party=calling
Call-ID: [email protected]
CSeq: 103 INVITE
Max-Forwards: 70
Authorization: Digest username=“12502”,realm=“asterisk”,nonce=“495b805a”,uri=“sip:[email protected]:5060”,algorithm=MD5,response="056f5644287185c1ff1c797ef5262d67"
Contact: “15203509672” sip:[email protected]:5062
Expires: 30
User-Agent: Cisco/SPA122-1.0.2(006)
Content-Length: 267
Content-Type: application/sdp

v=0
o=- 9867359 9867359 IN IP4 192.168.1.63
s=-
c=IN IP4 192.168.1.63
t=0 0
m=image 16474 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
— (14 headers 12 lines) —
Sending to 192.168.1.63:5062 (NAT)
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Got T.38 offer in SDP in dialog [email protected]
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.

<— Transmitting (NAT) to 192.168.1.63:5062 —>
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-ce5048df;received=192.168.1.63;rport=5062

From: “15203509672” sip:[email protected];tag=75b5948ecef18203o0

To: sip:[email protected];tag=as1566990a

Call-ID: [email protected]

CSeq: 103 INVITE

Server: FPBX-2.10.1(1.8.18.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Contact: sip:[email protected]:5060

Content-Length: 0

<------------>

e[Klocalhost*CLI>
e[0K == Using UDPTL TOS bits 184

e[Klocalhost*CLI>
e[0K == Using UDPTL CoS mark 5

e[Klocalhost*CLI>
e[0Kset_destination: Parsing sip:216.115.69.144;lr for address/port to send to

e[Klocalhost*CLI>
e[0Kset_destination: set destination to 216.115.69.144:5060

e[Klocalhost*CLI>
e[0KReliably Transmitting (NAT) to 216.115.69.144:5060:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK23f052ca;rport

Route: sip:216.115.69.144;lr,sip:216.115.69.133;lr

Max-Forwards: 70

From: “15203509672” sip:[email protected];tag=as516769f6

To: sip:[email protected];tag=SDvakj599-gK0defbcb0

Contact: sip:[email protected]:5060

Call-ID: [email protected]

CSeq: 104 INVITE

User-Agent: FPBX-2.10.1(1.8.18.0)

Session-Expires: 10800;refresher=uas

Min-SE: 90

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

X-asterisk-Info: SIP re-invite (External RTP bridge)

Content-Type: application/sdp

Content-Length: 265

v=0

o=root 1676371789 1676371791 IN IP4 192.168.4.10

s=Asterisk PBX 1.8.18.0

c=IN IP4 192.168.4.10

t=0 0

m=image 4556 udptl t38

a=T38FaxVersion:0

a=T38MaxBitRate:14400

a=T38FaxRateManagement:transferredTCF

a=T38FaxMaxDatagram:200

a=T38FaxUdpEC:t38UDPFEC

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:216.115.69.144:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK23f052ca;rport=1414
From: “15203509672” sip:[email protected];tag=as516769f6
To: sip:[email protected];tag=SDvakj599-gK0defbcb0
Call-ID: [email protected]
CSeq: 104 INVITE
Content-Length: 0

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

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:216.115.69.144:5060 —>
SIP/2.0 200 OK
From: “15203509672” sip:[email protected];tag=as516769f6
To: sip:[email protected];tag=SDvakj599-gK0defbcb0
Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK23f052ca;rport=1414
Call-ID: [email protected]
CSeq: 104 INVITE
Record-Route: sip:216.115.69.133;lr
Record-Route: sip:216.115.69.144;lr
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Contact: sip:[email protected]:5060;transport=udp
Session-Expires: 10800;refresher=uas
Content-Length: 173
Content-Type: application/sdp

v=0
o=- 12704 25220 IN IP4 67.16.125.60
s=-
c=IN IP4 67.16.125.60
t=0 0
m=image 53562 udptl t38
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
— (13 headers 8 lines) —
Got T.38 offer in SDP in dialog [email protected]
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
set_destination: Parsing sip:216.115.69.144;lr for address/port to send to
set_destination: set destination to 216.115.69.144:5060
Transmitting (NAT) to 216.115.69.144:5060:
ACK sip:[email protected]:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK69909498;rport

Route: sip:216.115.69.144;lr,sip:216.115.69.133;lr

Max-Forwards: 70

From: “15203509672” sip:[email protected];tag=as516769f6

To: sip:[email protected];tag=SDvakj599-gK0defbcb0

Contact: sip:[email protected]:5060

Call-ID: [email protected]

CSeq: 104 ACK

User-Agent: FPBX-2.10.1(1.8.18.0)

Content-Length: 0

e[Klocalhost*CLI>
e[0K
<— Reliably Transmitting (NAT) to 192.168.1.63:5062 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-ce5048df;received=192.168.1.63;rport=5062

From: “15203509672” sip:[email protected];tag=75b5948ecef18203o0

To: sip:[email protected];tag=as1566990a

Call-ID: [email protected]

CSeq: 103 INVITE

Server: FPBX-2.10.1(1.8.18.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Contact: sip:[email protected]:5060

Content-Type: application/sdp

Content-Length: 271

v=0

o=root 1639512434 1639512436 IN IP4 192.168.4.10

s=Asterisk PBX 1.8.18.0

c=IN IP4 192.168.4.10

t=0 0

m=image 4535 udptl t38

a=T38FaxVersion:0

a=T38MaxBitRate:2400

a=T38FaxRateManagement:transferredTCF

a=T38FaxMaxDatagram:397

a=T38FaxUdpEC:t38UDPRedundancy

<------------>

e[Klocalhost*CLI>
e[0K[2012-12-25 16:16:31] e[1;31mWARNINGe[0m[22440]: e[1;37mres_rtp_asterisk.ce[0m:e[1;37m2156e[0m e[1;37mast_rtp_reade[0m: RTP Read too short

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.1.63:5062 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-f41779f
From: “15203509672” sip:[email protected];tag=75b5948ecef18203o0
To: sip:[email protected];tag=as1566990a
Call-ID: [email protected]
CSeq: 103 ACK
Max-Forwards: 70
Authorization: Digest username=“12502”,realm=“asterisk”,nonce=“495b805a”,uri=“sip:[email protected]:5060”,algorithm=MD5,response="056f5644287185c1ff1c797ef5262d67"
Contact: “15203509672” sip:[email protected]:5062
User-Agent: Cisco/SPA122-1.0.2(006)
Content-Length: 0

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

e[Klocalhost*CLI>
e[0K[2012-12-25 16:16:35] e[1;31mWARNINGe[0m[22440]: e[1;37mres_rtp_asterisk.ce[0m:e[1;37m2156e[0m e[1;37mast_rtp_reade[0m: RTP Read too short

e[Klocalhost*CLI>
e[0K[2012-12-25 16:16:37] e[1;31mWARNINGe[0m[22440]: e[1;37mres_rtp_asterisk.ce[0m:e[1;37m2156e[0m e[1;37mast_rtp_reade[0m: RTP Read too short

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.3.108:5060 —>

<------------->

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.1.63:5062 —>
REGISTER sip:192.168.4.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-71041ca9
From: “15203509672” sip:[email protected];tag=75c27852e4dbd13fo0
To: “15203509672” sip:[email protected]
Call-ID: [email protected]
CSeq: 12508 REGISTER
Max-Forwards: 70
Authorization: Digest username=“12502”,realm=“asterisk”,nonce=“46af368a”,uri=“sip:192.168.4.10”,algorithm=MD5,response="6b6aa1865873757b94b2138424dc2565"
Contact: “15203509672” sip:[email protected]:5062;expires=60
User-Agent: Cisco/SPA122-1.0.2(006)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces

<------------->
— (13 headers 0 lines) —
Sending to 192.168.1.63:5062 (NAT)

<— Transmitting (NAT) to 192.168.1.63:5062 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-71041ca9;received=192.168.1.63;rport=5062

From: “15203509672” sip:[email protected];tag=75c27852e4dbd13fo0

To: “15203509672” sip:[email protected];tag=as512e72ee

Call-ID: [email protected]

CSeq: 12508 REGISTER

Server: FPBX-2.10.1(1.8.18.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“192d265e”

Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.1.63:5062 —>
REGISTER sip:192.168.4.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-43049e96
From: “15203509672” sip:[email protected];tag=75c27852e4dbd13fo0
To: “15203509672” sip:[email protected]
Call-ID: [email protected]
CSeq: 12509 REGISTER
Max-Forwards: 70
Authorization: Digest username=“12502”,realm=“asterisk”,nonce=“192d265e”,uri=“sip:192.168.4.10”,algorithm=MD5,response="d1fdc6c4d92b20811a5cb7c7e0a6e6e2"
Contact: “15203509672” sip:[email protected]:5062;expires=60
User-Agent: Cisco/SPA122-1.0.2(006)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces

<------------->
— (13 headers 0 lines) —
Sending to 192.168.1.63:5062 (NAT)
Reliably Transmitting (NAT) to 192.168.1.63:5062:
OPTIONS sip:[email protected]:5062 SIP/2.0

Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK773caa4a;rport

Max-Forwards: 70

From: “Unknown” sip:[email protected];tag=as71035b10

To: sip:[email protected]:5062

Contact: sip:[email protected]:5060

Call-ID: [email protected]:5060

CSeq: 102 OPTIONS

User-Agent: FPBX-2.10.1(1.8.18.0)

Date: Tue, 25 Dec 2012 23:16:43 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0

<— Transmitting (NAT) to 192.168.1.63:5062 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-43049e96;received=192.168.1.63;rport=5062

F
e[Klocalhost*CLI>
e[0Krom: “15203509672” sip:[email protected];tag=75c27852e4dbd13fo0

To: “15203509672” sip:[email protected];tag=as512e72ee

Call-ID: [email protected]

CSeq: 12509 REGISTER

Server: FPBX-2.10.1(1.8.18.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Expires: 60

Contact: sip:[email protected]:5062;expires=60

Date: Tue, 25 Dec 2012 23:16:43 GMT

Content-Length: 0

<------------>

e[Klocalhost*CLI>
e[0KScheduling destruction of SIP dialog ‘[email protected]:5060’ in 6400 ms (Method: NOTIFY)

e[Klocalhost*CLI>
e[0KReliably Transmitting (NAT) to 192.168.1.63:5062:
NOTIFY sip:[email protected]:5062 SIP/2.0

Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK0709400d;rport

Max-Forwards: 70

From: “Unknown” sip:[email protected];tag=as08c6aa90

To: sip:[email protected]:5062

Contact: sip:[email protected]:5060

Call-ID: [email protected]:5060

CSeq: 102 NOTIFY

User-Agent: FPBX-2.10.1(1.8.18.0)

Event: message-summary

Content-Type: application/simple-message-summary

Content-Length: 87

Messages-Waiting: no

Message-Account: sip:*[email protected]

Voice-Message: 0/0 (0/0)

e[Klocalhost*CLI>
e[0KScheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: REGISTER)

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.1.63:5062 —>
SIP/2.0 486 Busy Here
To: sip:[email protected]:5062;tag=b11a580eb4d114b3i0
From: “Unknown” sip:[email protected];tag=as71035b10
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK773caa4a
Server: Cisco/SPA122-1.0.2(006)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces

<------------->

e[Klocalhost*CLI>
e[0K— (10 headers 0 lines) —

e[Klocalhost*CLI>
e[0KReally destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.1.63:5062 —>
SIP/2.0 200 OK
To: sip:[email protected]:5062;tag=b11a580eb4d114b3i0
From: “Unknown” sip:[email protected];tag=as08c6aa90
Call-ID: [email protected]:5060
CSeq: 102 NOTIFY
Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK0709400d
Server: Cisco/SPA122-1.0.2(006)
Content-Length: 0

<------------->

e[Klocalhost*CLI>
e[0K— (8 headers 0 lines) —

e[Klocalhost*CLI>
e[0KReally destroying SIP dialog ‘[email protected]:5060’ Method: NOTIFY

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:192.168.1.63:5062 —>
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-1b3acadf
From: “15203509672” sip:[email protected];tag=75b5948ecef18203o0
To: sip:[email protected];tag=as1566990a
Call-ID: [email protected]
CSeq: 104 BYE
Max-Forwards: 70
Authorization: Digest username=“12502”,realm=“asterisk”,nonce=“495b805a”,uri=“sip:[email protected]:5060”,algorithm=MD5,response="b8d9fa508ed05207fc32247159b9014d"
User-Agent: Cisco/SPA122-1.0.2(006)
P-RTP-Stat: PS=0,OS=0,PR=0,OR=0,PL=0,JI=0,LA=0,DU=14,EN=G711u,DE=G711u
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 192.168.1.63:5062 (NAT)
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: BYE)

<— Transmitting (NAT) to 192.168.1.63:5062 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.63:5062;branch=z9hG4bK-1b3acadf;received=192.168.1.63;rport=5062

From: “15203509672” sip:[email protected];tag=75b5948ecef18203o0

To: sip:[email protected];tag=as1566990a

Call-ID: [email protected]

CSeq: 104 BYE

Server: FPBX-2.10.1(1.8.18.0)

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces, timer

Content-Length: 0

<------------>

e[Klocalhost*CLI>
e[0K – Executing [h@macro-dialout-trunk:1] e[1;36mMacroe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35mhangupcall,e[0m”) in new stack
– Executing [s@macro-hangupcall:1] e[1;36mGotoIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m1?theende[0m”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] e[1;36mExecIfe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35m0?Set(CDR(recordingfile)=)e[0m”) in new stack
– Executing [s@macro-hangupcall:4] e[1;36mHangupe[0m(“e[1;35mSIP/12502-00000302e[0m”, “e[1;35me[0m”) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/12502-00000302’ in macro ‘hangupcall’
== Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on ‘SIP/12502-00000302’

e[Klocalhost*CLI>
e[0KScheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)
set_destination: Parsing sip:216.115.69.144;lr for address/port to send to
set_destination: set destination to 216.115.69.144:5060
Reliably Transmitting (NAT) to 216.115.69.144:5060:
BYE sip:[email protected]:5060;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK5b3dcc90;rport

Route: sip:216.115.69.144;lr,sip:216.115.69.133;lr

Max-Forwards: 70

From: “15203509672” sip:[email protected];tag=as516769f6

To: sip:[email protected];tag=SDvakj599-gK0defbcb0

Call-ID: [email protected]

CSeq: 105 BYE

User-Agent: FPBX-2.10.1(1.8.18.0)

Proxy-Authorization: Digest username=“08922051”, realm=“sip.flowroute.com”, algorithm=MD5, uri=“sip:[email protected]:5060”, nonce=“UNo07FDaM8AfB4Lu1acUFHticjdM/9dX”, response=“7e4659b7acec3ed001e21c23fd3057c5”, qop=auth, cnonce=“4830cd99”, nc=00000002

X-Asterisk-HangupCause: Normal Clearing

X-Asterisk-HangupCauseCode: 16

Content-Length: 0

e[Klocalhost*CLI>
e[0K == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on ‘SIP/12502-00000302’ in macro ‘dialout-trunk’

e[Klocalhost*CLI>
e[0K == Spawn extension (from-internal, 14792460547, 5) exited non-zero on ‘SIP/12502-00000302’

e[Klocalhost*CLI>
e[0K == MixMonitor close filestream

e[Klocalhost*CLI>
e[0K == End MixMonitor Recording SIP/12502-00000302

e[Klocalhost*CLI>
e[0K
<— SIP read from UDP:216.115.69.144:5060 —>
SIP/2.0 200 OK
From: “15203509672” sip:[email protected];tag=as516769f6
To: sip:[email protected];tag=SDvakj599-gK0defbcb0
Via: SIP/2.0/UDP 192.168.4.10:5060;branch=z9hG4bK5b3dcc90;rport=1414
Call-ID: [email protected]
CSeq: 105 BYE
Record-Route: sip:216.115.69.133;lr
Record-Route: sip:216.115.69.144;lr
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]’ Method: INVITE

e

rcraig114 · December 26, 2012, 4:49pm

Does anyone have an idea? I’ve been at this all night. I still can’t get past the error below and my innability to hear the distant end fax tones.

[2012-12-26 09:37:03] WARNING[10235] res_rtp_asterisk.c: RTP Read too short

The above error appears after the call is answered in either direction and the fax fails. I also tried adjusting the RTP packet size in the ATA from .030 to .020, no luck.

tonyclewis · December 26, 2012, 5:28pm

Based on a quick glance the person you are trying to fax to does not support T38

rcraig114 · December 26, 2012, 5:29pm

Thats what I thought, but I confirmed with Flowroute (provider) that they do support T.38. When I had a CME registered to Flowroute and the same ATA behind the CME, T.38 worked perfectly. It’s when I changed the mix to Flowroute-FreePBX-ATA that T.38 is failing.

rcraig114 · December 28, 2012, 3:39am

I’m still having issues with T.38. Does anyone know what the below error means? It only happens when I turn on ReInvite on the ATA.

[2012-12-26 09:37:03] WARNING[10235] res_rtp_asterisk.c: RTP Read too short

petersonz · February 25, 2013, 10:23pm

I have both Flowroute and AlcazarNetworks and they both claim T.38 support with the usual disclaimer of “good luck”.

My FreePBX is 2.11 with Asterisk 11 and hosted in the cloud with a public IP (no NAT).

My ATA is a Cisco SPA 112 with the brand new firmware and will be behind a NAT firewall. Voice calls already work perfectly with it.

Ill let you know if t.38 passthrough works for me.

I wouldnt think you would want fax detection on the inbound route in FreePBX since that means Asterisk will try and detect the tones and answer for itself which would prevent it from passing through to your ATA… It should just point to the extension matching your ATA port.

The one thing Im not sure about is the allow or disallow codec choices… If you disallow “ALL” and allow just “ulaw”, does that prevent t.38 passthrough? Trunk and extension?

SkykingOH · February 26, 2013, 12:47am

What you are describing is t.38 gateway mode, pass through is if the call is already t.38.

petersonz · February 26, 2013, 1:41am

Here is the setup Im trying

SIP Trunk coming from T38 provider going to ATA on FreePBX with T38 Passthru enabled.

PSTN–T38 SIP Provider–Internet–FreePBX–Internet–NAT Firewall–T38 ATA–Fax Machine

Is that not a passthrough since either the SIP provider starts the T.38 or the ATA starts the T.38 ?

SkykingOH · February 26, 2013, 5:29am

Yes but just because the provider supports t.38 don’t assume they have a t.38 gateway. This really trips people up. You need to check each and every exchange to see what the capabilities are.

petersonz · February 26, 2013, 5:45am

That could be a source of a lot of FreePBX Asterisk/T.38 grief.

If the SIP provider only supports t.38 transitioning on the SIP side of their service, without the PSTN/SIP t.38 gateway, that would be the source of a lot of confusion and finger pointing.

I guess its easy to assume that when they mention t.38, they gotta be talking about the gateway

In any case, I can also test with an FXO ATA plugged into my home pots line and an FXS ATA plugged into my fax machine, both through Asterisk just to see if it passes through like it should and doesnt have to fall back to g.711u. Once established as working from ATA to ATA, then I could switch it too the trunk of the SIP providers to test.

There should probably be a whole thread dedicated to testing SIP provider’s t.38 gateway capabilities