[resolved] intermittently not detecting or ignoring t38

Hi,

Our setup is Sip Trunk <–> Asterisk <–> Fax with inbuild t.38. Running asterisk version 11.15.0

Inbound call where t.38 works successfully:

[code]<— SIP read from UDP:10.1.1.251:63825 —>
INVITE sip:xxx@10.2.0.72;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.251;branch=z9hG4bK5gD4Eohl7eGkl
Max-Forwards: 70
From: sip:029855555@10.1.1.251;user=phone;tag=x’DdEH!r6N
To: “xxx” sip:xxx@10.2.0.72;tag=as79a7cf9b
Call-ID: 41ca7d952d2727d910436c9e628f34f6@10.2.0.72:5060
User-Agent: RicohIAF Version 2.0
CSeq: 1 INVITE
Contact: sip:029855555@10.1.1.251;user=phone
Content-Type: application/sdp
Content-Length: 408

v=0
o=RICOH-SIP-IPFAX 1419282925 1419282925 IN IP4 10.1.1.251
s=Session SDP
t=0 0
m=image 49152 tcp t38
c=IN IP4 10.1.1.251
a=T38FaxVersion:0
a=T38MaxBitRate:144
a=T38FaxRateManagement:localTCF
a=T38VendorInfo:0 0 37
m=image 49152 udptl t38
c=IN IP4 10.1.1.251
a=T38FaxVersion:0
a=T38MaxBitRate:144
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
a=T38VendorInfo:0 0 37
<------------->
[Dec 23 08:17:02] VERBOSE[2165] chan_sip.c: — (11 headers 17 lines) —
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: Sending to 10.1.1.251:5060 (no NAT)
[Dec 23 08:17:02] WARNING[2165][C-00000001] chan_sip.c: Declining image stream due to unsupported transport: image 49152 tcp t38
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] netsock2.c: == Using UDPTL CoS mark 5
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: Got T.38 offer in SDP in dialog 41ca7d952d2727d910436c9e628f34f6@10.2.0.72:5060
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140$
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c:
<— Transmitting (no NAT) to 10.1.1.251:5060 —>
[/code]

Yet when we hangup and call back without changing anything, it fails to acknowledge the t.38 invite:

[code]<— SIP read from UDP:10.1.1.251:63820 —>
INVITE sip:xxx@10.2.0.72;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.251;branch=z9hG4bKw+5DNy_s3FT30
Max-Forwards: 70
From: sip:029855555@10.1.1.251;user=phone;tag=j7bCQq.jbO
To: “xxx” sip:xxx@10.2.0.72;tag=as0e1003da
Call-ID: 664f4ad85f822a433b6fa0e2779b3d36@10.2.0.72:5060
User-Agent: RicohIAF Version 2.0
CSeq: 1 INVITE
Contact: sip:029855555@10.1.1.251;user=phone
Content-Type: application/sdp
Content-Length: 408

v=0
o=RICOH-SIP-IPFAX 1419284055 1419284055 IN IP4 10.1.1.251
s=Session SDP
t=0 0
m=image 49152 tcp t38
c=IN IP4 10.1.1.251
a=T38FaxVersion:0
a=T38MaxBitRate:144
a=T38FaxRateManagement:localTCF
a=T38VendorInfo:0 0 37
m=image 49152 udptl t38
c=IN IP4 10.1.1.251
a=T38FaxVersion:0
a=T38MaxBitRate:144
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
a=T38VendorInfo:0 0 37
<------------->
[Dec 23 08:35:52] VERBOSE[2165] chan_sip.c: — (11 headers 17 lines) —
[Dec 23 08:35:52] VERBOSE[2165][C-00000005] chan_sip.c: Sending to 10.1.1.251:5060 (no NAT)
[Dec 23 08:35:52] VERBOSE[2165][C-00000005] chan_sip.c:
<— Transmitting (no NAT) to 10.1.1.251:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.251;branch=z9hG4bKw+5DNy_s3FT30;received=10.1.1.251
From: sip:029855555@10.1.1.251;user=phone;tag=j7bCQq.jbO
To: “xxx” sip:xxx@10.2.0.72;tag=as0e1003da
Call-ID: 664f4ad85f822a433b6fa0e2779b3d36@10.2.0.72:5060
CSeq: 1 INVITE
Server: Firestick ICT PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:xxx@10.2.0.72:5060
Content-Length: 0
[/code]

As far as I can tell the invite from the fax machine is the same, however the t.38 is ignored a majority of the time and the call promptly terminates.

I don’t think this is a problem with our SIP provider, as if I have asterisk --> iaxmodem --> hylafax receive the fax, it switches to t.38 successfully every time.

The trace is insufficiently complete, but my guess is that, if you provided a more complete trace, the call ID on the second INVITE would be a duplicate of one that you have cut out.

Hi David,

I was worried about publishing the full log, however I’ve done a search / replace on anything that could be considered sensitive (public IP’s / usernames / Caller ID Name / Number), everything else is as original. I’m no expert at reading pcaps, but I’ve tried to start and stop the log where the relevant parts are.

Paul

Call 1 (working):

<------------>
[Dec 23 08:16:59] VERBOSE[2165] chan_sip.c: Scheduling destruction of SIP dialog '747f976d022e97ec-6373@x.y.35.121' in 32000 ms (Method: OPTIONS)
[Dec 23 08:17:01] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:x.25.112.19:5060 --->
INVITE sip:61298555555@1.2.127.65:5060 SIP/2.0
Record-Route: <sip:x.25.112.19;lr;ftag=as0e8e4a75;did=afe.42300e31>
Via: SIP/2.0/UDP x.25.112.19:5060;branch=z9hG4bKacd6.b66c1267.0
Max-Forwards: 69
To: <sip:61298555555@x.25.112.19:5060>
From: "61xxx"<sip:61xxx@x.25.112.19>;tag=as0e8e4a75
Call-ID: 6b9be3d0759011ce051c48484ff4e306@x.25.112.19:5050
Contact: <sip:61xxx@x.25.112.19:5060>
CSeq: 102 INVITE
User-Agent: VoIP Networks
Date: Mon, 22 Dec 2014 21:16:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-voipnow-extension: out-sip
X-voipnow-pbx: e24fee848be09752172d
X-voipnow-infrastructureid: 14c3aabd36a5
X-voipnow-did: 61298555555
Content-Type: application/sdp
Content-Length: 307

v=0
o=root 628248660 628248660 IN IP4 x.25.112.19
s=Asterisk PBX 1.8.9.2
c=IN IP4 x.25.112.19
t=0 0
m=audio 15600 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/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
a=sendrecv
<------------->
[Dec 23 08:17:01] VERBOSE[2165] chan_sip.c: --- (19 headers 14 lines) ---
[Dec 23 08:17:01] VERBOSE[2165] chan_sip.c: Sending to x.25.112.19:5060 (no NAT)
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: Sending to x.25.112.19:5060 (no NAT)
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: Using INVITE request as basis request - 6b9be3d0759011ce051c48484ff4e306@x.25.112.19:5050
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: Found peer 'out-sip' for '61xxx' from x.25.112.19:5060
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] netsock2.c:   == Using SIP RTP CoS mark 5
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: Found RTP audio format 0
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: Found RTP audio format 8
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: Found RTP audio format 18
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: Found RTP audio format 101
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: Found audio description format PCMU for ID 0
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: Found audio description format PCMA for ID 8
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: Found audio description format G729 for ID 18
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 101
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: Peer audio RTP is at port x.25.112.19:15600
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: Looking for 61298555555 in inbound (domain 1.2.127.65)
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: list_route: hop: <sip:x.25.112.19;lr;ftag=as0e8e4a75;did=afe.42300e31>
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: 
<--- Transmitting (no NAT) to x.25.112.19:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP x.25.112.19:5060;branch=z9hG4bKacd6.b66c1267.0;received=x.25.112.19
Record-Route: <sip:x.25.112.19;lr;ftag=as0e8e4a75;did=afe.42300e31>
From: "61xxx"<sip:61xxx@x.25.112.19>;tag=as0e8e4a75
To: <sip:61298555555@x.25.112.19:5060>
Call-ID: 6b9be3d0759011ce051c48484ff4e306@x.25.112.19:5050
CSeq: 102 INVITE
Server: Firestick ICT PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:61298555555@1.2.127.65:5060>
Content-Length: 0


<------------>
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [61298555555@inbound:1] NoOp("SIP/out-sip-00000002", "") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [61298555555@inbound:2] Goto("SIP/out-sip-00000002", "inbound,0298555555,1") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Goto (inbound,0298555555,1)
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@inbound:1] NoOp("SIP/out-sip-00000002", "61xxx 61xxx To 61298555555") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@inbound:2] Set("SIP/out-sip-00000002", "CALLERID(num)=0xxx") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@inbound:3] GotoIf("SIP/out-sip-00000002", "0?anonymous:") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@inbound:4] GotoIf("SIP/out-sip-00000002", "0?anonymous:") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@inbound:5] Set("SIP/out-sip-00000002", "CIDN=xxx") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@inbound:6] Set("SIP/out-sip-00000002", "CALLERID(name)=xxx") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@inbound:7] Set("SIP/out-sip-00000002", "CIDFOUND=found") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@inbound:8] Set("SIP/out-sip-00000002", "DB(cid/staff)=11916") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@inbound:9] Set("SIP/out-sip-00000002", "DB(cid/stafft)=6") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@inbound:10] GotoIf("SIP/out-sip-00000002", "1?found:") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Goto (inbound,0298555555,30)
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@inbound:30] NoOp("SIP/out-sip-00000002", "callerid name xxx") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@inbound:31] Set("SIP/out-sip-00000002", "PCID2=xxx") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@inbound:32] Set("SIP/out-sip-00000002", "CALLERID(name)=xxx") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@inbound:33] Set("SIP/out-sip-00000002", "CALLERID(name)=xxx") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@inbound:34] NoOp("SIP/out-sip-00000002", "xxx") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@inbound:35] NoOp("SIP/out-sip-00000002", "") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@inbound:36] Goto("SIP/out-sip-00000002", "incoming,0298555555,1") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Goto (incoming,0298555555,1)
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@incoming:1] Goto("SIP/out-sip-00000002", "fax-receive,0298555555,1") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Goto (fax-receive,0298555555,1)
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@fax-receive:1] NoOp("SIP/out-sip-00000002", "") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@fax-receive:2] NoOp("SIP/out-sip-00000002", "0") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@fax-receive:3] Set("SIP/out-sip-00000002", "FAXOPT(gateway)=yes") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@fax-receive:4] Set("SIP/out-sip-00000002", "CHANNEL(t38passthrough)=yes") in new stack
[Dec 23 08:17:01] WARNING[2173][C-00000001] func_channel.c: Unknown or unavailable item requested: 't38passthrough'
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@fax-receive:5] NoOp("SIP/out-sip-00000002", "0") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] pbx.c:     -- Executing [0298555555@fax-receive:6] Dial("SIP/out-sip-00000002", "SIP/0298555555") in new stack
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] netsock2.c:   == Using SIP RTP CoS mark 5
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: We think we can do text
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Audio is at 19736
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100001 (g723) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100002 (gsm) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100002 (gsm) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100005 (g726aal2) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100006 (adpcm) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100007 (lpc10) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100008 (g729) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100009 (speex) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100010 (ilbc) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100011 (g726) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100012 (g722) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100013 (siren7) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100014 (siren14) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100015 (g719) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100016 (speex16) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100017 (testlaw) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100017 (testlaw) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100019 (slin) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100020 (slin12) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100021 (slin16) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100022 (slin24) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100023 (slin32) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100024 (slin44) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100025 (slin48) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100026 (slin96) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100027 (slin192) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100028 (speex32) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 10.1.1.251:5060:
INVITE sip:0298555555@10.1.1.251;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.2.0.72:5060;branch=z9hG4bK2843b597
Max-Forwards: 70
From: "xxx" <sip:0xxx@10.2.0.72>;tag=as79a7cf9b
To: <sip:0298555555@10.1.1.251;user=phone>
Contact: <sip:0xxx@10.2.0.72:5060>
Call-ID: 41ca7d952d2727d910436c9e628f34f6@10.2.0.72:5060
CSeq: 102 INVITE
User-Agent: Firestick ICT PBX
Date: Mon, 22 Dec 2014 21:17:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "xxx" <sip:0xxx@10.2.0.72>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 904

v=0
o=root 1670434890 1670434890 IN IP4 10.2.0.72
s=Firestick ICT PBX
c=IN IP4 10.2.0.72
t=0 0
m=audio 19736 RTP/AVP 0 4 3 3 8 8 112 5 7 18 110 97 111 9 102 115 116 117 10 118 119 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:117 speex/16000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:119 speex/32000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] app_dial.c:     -- Called SIP/0298555555
[Dec 23 08:17:01] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:10.1.1.251:63825 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.0.72:5060;branch=z9hG4bK2843b597
From: "xxx" <sip:0xxx@10.2.0.72>;tag=as79a7cf9b
To: <sip:0298555555@10.1.1.251;user=phone>;tag=x'DdEH!r6N
Call-ID: 41ca7d952d2727d910436c9e628f34f6@10.2.0.72:5060
User-Agent: RicohIAF Version 2.0
CSeq: 102 INVITE
Content-Length: 0

<------------->
[Dec 23 08:17:01] VERBOSE[2165] chan_sip.c: --- (8 headers 0 lines) ---
[Dec 23 08:17:01] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:10.1.1.251:63825 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.2.0.72:5060;branch=z9hG4bK2843b597
From: "xxx" <sip:0xxx@10.2.0.72>;tag=as79a7cf9b
To: <sip:0298555555@10.1.1.251;user=phone>;tag=x'DdEH!r6N
Call-ID: 41ca7d952d2727d910436c9e628f34f6@10.2.0.72:5060
User-Agent: RicohIAF Version 2.0
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 153

v=0
o=RICOH-SIP-IPFAX 1419282924 1419282924 IN IP4 10.1.1.251
s=Session SDP
t=0 0
m=audio 5004 RTP/AVP 0
c=IN IP4 10.1.1.251
a=rtpmap:0 PCMU/8000
<------------->
[Dec 23 08:17:01] VERBOSE[2165] chan_sip.c: --- (9 headers 7 lines) ---
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: list_route: no route
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: Found RTP audio format 0
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: Found audio description format PCMU for ID 0
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Dec 23 08:17:01] VERBOSE[2165][C-00000001] chan_sip.c: Peer audio RTP is at port 10.1.1.251:5004
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] app_dial.c:     -- SIP/0298555555-00000003 is ringing
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: 
<--- Transmitting (no NAT) to x.25.112.19:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP x.25.112.19:5060;branch=z9hG4bKacd6.b66c1267.0;received=x.25.112.19
Record-Route: <sip:x.25.112.19;lr;ftag=as0e8e4a75;did=afe.42300e31>
From: "61xxx"<sip:61xxx@x.25.112.19>;tag=as0e8e4a75
To: <sip:61298555555@x.25.112.19:5060>;tag=as34e60dbb
Call-ID: 6b9be3d0759011ce051c48484ff4e306@x.25.112.19:5050
CSeq: 102 INVITE
Server: Firestick ICT PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:61298555555@1.2.127.65:5060>
Content-Length: 0


<------------>
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] app_dial.c:     -- SIP/0298555555-00000003 is making progress passing it to SIP/out-sip-00000002
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Audio is at 16818
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Dec 23 08:17:01] VERBOSE[2173][C-00000001] chan_sip.c: 
<--- Transmitting (no NAT) to x.25.112.19:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP x.25.112.19:5060;branch=z9hG4bKacd6.b66c1267.0;received=x.25.112.19
Record-Route: <sip:x.25.112.19;lr;ftag=as0e8e4a75;did=afe.42300e31>
From: "61xxx"<sip:61xxx@x.25.112.19>;tag=as0e8e4a75
To: <sip:61298555555@x.25.112.19:5060>;tag=as34e60dbb
Call-ID: 6b9be3d0759011ce051c48484ff4e306@x.25.112.19:5050
CSeq: 102 INVITE
Server: Firestick ICT PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:61298555555@1.2.127.65:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 257

v=0
o=root 1479028997 1479028997 IN IP4 1.2.127.65
s=Firestick ICT PBX
c=IN IP4 1.2.127.65
t=0 0
m=audio 16818 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[Dec 23 08:17:02] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:10.1.1.251:63825 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.0.72:5060;branch=z9hG4bK2843b597
From: "xxx" <sip:0xxx@10.2.0.72>;tag=as79a7cf9b
To: <sip:0298555555@10.1.1.251;user=phone>;tag=x'DdEH!r6N
Call-ID: 41ca7d952d2727d910436c9e628f34f6@10.2.0.72:5060
User-Agent: RicohIAF Version 2.0
CSeq: 102 INVITE
Contact: <sip:0298555555@10.1.1.251;user=phone>
Content-Type: application/sdp
Content-Length: 153

v=0
o=RICOH-SIP-IPFAX 1419282924 1419282924 IN IP4 10.1.1.251
s=Session SDP
t=0 0
m=audio 5004 RTP/AVP 0
c=IN IP4 10.1.1.251
a=rtpmap:0 PCMU/8000
<------------->
[Dec 23 08:17:02] VERBOSE[2165] chan_sip.c: --- (10 headers 7 lines) ---
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: list_route: hop: <sip:0298555555@10.1.1.251;user=phone>
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: set_destination: Parsing <sip:0298555555@10.1.1.251;user=phone> for address/port to send to
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: set_destination: set destination to 10.1.1.251:5060
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: Transmitting (no NAT) to 10.1.1.251:5060:
ACK sip:0298555555@10.1.1.251;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.2.0.72:5060;branch=z9hG4bK482df856
Max-Forwards: 70
From: "xxx" <sip:0xxx@10.2.0.72>;tag=as79a7cf9b
To: <sip:0298555555@10.1.1.251;user=phone>;tag=x'DdEH!r6N
Contact: <sip:0xxx@10.2.0.72:5060>
Call-ID: 41ca7d952d2727d910436c9e628f34f6@10.2.0.72:5060
CSeq: 102 ACK
User-Agent: Firestick ICT PBX
Content-Length: 0


---
[Dec 23 08:17:02] VERBOSE[2173][C-00000001] app_dial.c:     -- SIP/0298555555-00000003 answered SIP/out-sip-00000002
[Dec 23 08:17:02] VERBOSE[2173][C-00000001] chan_sip.c: Audio is at 16818
[Dec 23 08:17:02] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Dec 23 08:17:02] VERBOSE[2173][C-00000001] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Dec 23 08:17:02] VERBOSE[2173][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Dec 23 08:17:02] VERBOSE[2173][C-00000001] chan_sip.c: 
<--- Reliably Transmitting (no NAT) to x.25.112.19:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.25.112.19:5060;branch=z9hG4bKacd6.b66c1267.0;received=x.25.112.19
Record-Route: <sip:x.25.112.19;lr;ftag=as0e8e4a75;did=afe.42300e31>
From: "61xxx"<sip:61xxx@x.25.112.19>;tag=as0e8e4a75
To: <sip:61298555555@x.25.112.19:5060>;tag=as34e60dbb
Call-ID: 6b9be3d0759011ce051c48484ff4e306@x.25.112.19:5050
CSeq: 102 INVITE
Server: Firestick ICT PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:61298555555@1.2.127.65:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 257

v=0
o=root 1479028997 1479028997 IN IP4 1.2.127.65
s=Firestick ICT PBX
c=IN IP4 1.2.127.65
t=0 0
m=audio 16818 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[Dec 23 08:17:02] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:x.25.112.19:5060 --->
ACK sip:61298555555@1.2.127.65:5060 SIP/2.0
Via: SIP/2.0/UDP x.25.112.19:5060;branch=z9hG4bKcydzigwkX
Max-Forwards: 69
To: <sip:61298555555@x.25.112.19:5060>;tag=as34e60dbb
From: "61xxx"<sip:61xxx@x.25.112.19>;tag=as0e8e4a75
Call-ID: 6b9be3d0759011ce051c48484ff4e306@x.25.112.19:5050
Contact: <sip:61xxx@x.25.112.19:5060>
CSeq: 102 ACK
User-Agent: VoIP Networks
Content-Length: 0

<------------->
[Dec 23 08:17:02] VERBOSE[2165] chan_sip.c: --- (10 headers 0 lines) ---
[Dec 23 08:17:02] VERBOSE[2173][C-00000001] res_rtp_asterisk.c:        > 0x7f35b403f4e0 -- Probation passed - setting RTP source address to x.25.112.19:15600
[Dec 23 08:17:02] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:10.1.1.251:63825 --->
INVITE sip:0xxx@10.2.0.72;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.251;branch=z9hG4bK5gD4Eohl7eGkl
Max-Forwards: 70
From: <sip:0298555555@10.1.1.251;user=phone>;tag=x'DdEH!r6N
To: "xxx" <sip:0xxx@10.2.0.72>;tag=as79a7cf9b
Call-ID: 41ca7d952d2727d910436c9e628f34f6@10.2.0.72:5060
User-Agent: RicohIAF Version 2.0
CSeq: 1 INVITE
Contact: <sip:0298555555@10.1.1.251;user=phone>
Content-Type: application/sdp
Content-Length: 408

v=0
o=RICOH-SIP-IPFAX 1419282925 1419282925 IN IP4 10.1.1.251
s=Session SDP
t=0 0
m=image 49152 tcp t38
c=IN IP4 10.1.1.251
a=T38FaxVersion:0
a=T38MaxBitRate:144
a=T38FaxRateManagement:localTCF
a=T38VendorInfo:0 0 37
m=image 49152 udptl t38
c=IN IP4 10.1.1.251
a=T38FaxVersion:0
a=T38MaxBitRate:144
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
a=T38VendorInfo:0 0 37
<------------->
[Dec 23 08:17:02] VERBOSE[2165] chan_sip.c: --- (11 headers 17 lines) ---
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: Sending to 10.1.1.251:5060 (no NAT)
[Dec 23 08:17:02] WARNING[2165][C-00000001] chan_sip.c: Declining image stream due to unsupported transport: image 49152 tcp t38
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] netsock2.c:   == Using UDPTL CoS mark 5
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: Got T.38 offer in SDP in dialog 41ca7d952d2727d910436c9e628f34f6@10.2.0.72:5060
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.251:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.251;branch=z9hG4bK5gD4Eohl7eGkl;received=10.1.1.251
From: <sip:0298555555@10.1.1.251;user=phone>;tag=x'DdEH!r6N
To: "xxx" <sip:0xxx@10.2.0.72>;tag=as79a7cf9b
Call-ID: 41ca7d952d2727d910436c9e628f34f6@10.2.0.72:5060
CSeq: 1 INVITE
Server: Firestick ICT PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0xxx@10.2.0.72:5060>
Content-Length: 0


<------------>
[Dec 23 08:17:02] VERBOSE[2173][C-00000001] netsock2.c:   == Using UDPTL CoS mark 5
[Dec 23 08:17:02] VERBOSE[2173][C-00000001] chan_sip.c: set_destination: Parsing <sip:x.25.112.19;lr;ftag=as0e8e4a75;did=afe.42300e31> for address/port to send to
[Dec 23 08:17:02] VERBOSE[2173][C-00000001] chan_sip.c: set_destination: set destination to x.25.112.19:5060
[Dec 23 08:17:02] VERBOSE[2173][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to x.25.112.19:5060:
INVITE sip:61xxx@x.25.112.19:5060 SIP/2.0
Via: SIP/2.0/UDP 1.2.127.65:5060;branch=z9hG4bK0c763be7
Route: <sip:x.25.112.19;lr;ftag=as0e8e4a75;did=afe.42300e31>
Max-Forwards: 70
From: <sip:61298555555@x.25.112.19:5060>;tag=as34e60dbb
To: "61xxx"<sip:61xxx@x.25.112.19>;tag=as0e8e4a75
Contact: <sip:61298555555@1.2.127.65:5060>
Call-ID: 6b9be3d0759011ce051c48484ff4e306@x.25.112.19:5050
CSeq: 102 INVITE
User-Agent: Firestick ICT PBX
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 291

v=0
o=root 1479028997 1479028998 IN IP4 1.2.127.65
s=Firestick ICT PBX
c=IN IP4 1.2.127.65
t=0 0
m=image 4036 udptl t38
c=IN IP4 1.2.127.65
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:397
a=T38FaxUdpEC:t38UDPRedundancy

---
[Dec 23 08:17:02] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:x.25.112.19:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 1.2.127.65:5060;branch=z9hG4bK0c763be7
From: <sip:61298555555@x.25.112.19:5060>;tag=as34e60dbb
To: "61xxx"<sip:61xxx@x.25.112.19>;tag=as0e8e4a75
Call-ID: 6b9be3d0759011ce051c48484ff4e306@x.25.112.19:5050
CSeq: 102 INVITE
Server: VoipNow
Content-Length: 0

<------------->
[Dec 23 08:17:02] VERBOSE[2165] chan_sip.c: --- (8 headers 0 lines) ---
[Dec 23 08:17:02] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:x.25.112.19:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 1.2.127.65:5060;branch=z9hG4bK0c763be7
From: <sip:61298555555@x.25.112.19:5060>;tag=as34e60dbb
To: "61xxx"<sip:61xxx@x.25.112.19>;tag=as0e8e4a75
Call-ID: 6b9be3d0759011ce051c48484ff4e306@x.25.112.19:5050
Contact: <sip:61xxx@x.25.112.19:5060>
CSeq: 102 INVITE
Server: VoIP Networks
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 271

v=0
o=root 628248660 628248661 IN IP4 x.25.112.19
s=Asterisk PBX 1.8.9.2
c=IN IP4 x.25.112.19
t=0 0
m=image 5233 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:397
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
[Dec 23 08:17:02] VERBOSE[2165] chan_sip.c: --- (13 headers 11 lines) ---
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: Got T.38 offer in SDP in dialog 6b9be3d0759011ce051c48484ff4e306@x.25.112.19:5050
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: Capabilities: us - (ulaw|alaw), peer - audio=(nothing)/video=(nothing)/text=(nothing), combined - (nothing)
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: set_destination: Parsing <sip:x.25.112.19;lr;ftag=as0e8e4a75;did=afe.42300e31> for address/port to send to
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: set_destination: set destination to x.25.112.19:5060
[Dec 23 08:17:02] VERBOSE[2165][C-00000001] chan_sip.c: Transmitting (no NAT) to x.25.112.19:5060:
ACK sip:61xxx@x.25.112.19:5060 SIP/2.0
Via: SIP/2.0/UDP 1.2.127.65:5060;branch=z9hG4bK162e4e1b
Route: <sip:x.25.112.19;lr;ftag=as0e8e4a75;did=afe.42300e31>
Max-Forwards: 70
From: <sip:61298555555@x.25.112.19:5060>;tag=as34e60dbb
To: "61xxx"<sip:61xxx@x.25.112.19>;tag=as0e8e4a75
Contact: <sip:61298555555@1.2.127.65:5060>
Call-ID: 6b9be3d0759011ce051c48484ff4e306@x.25.112.19:5050
CSeq: 102 ACK
User-Agent: Firestick ICT PBX
Content-Length: 0


---
[Dec 23 08:17:02] VERBOSE[2173][C-00000001] chan_sip.c: 
<--- Reliably Transmitting (no NAT) to 10.1.1.251:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.251;branch=z9hG4bK5gD4Eohl7eGkl;received=10.1.1.251
From: <sip:0298555555@10.1.1.251;user=phone>;tag=x'DdEH!r6N
To: "xxx" <sip:0xxx@10.2.0.72>;tag=as79a7cf9b
Call-ID: 41ca7d952d2727d910436c9e628f34f6@10.2.0.72:5060
CSeq: 1 INVITE
Server: Firestick ICT PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0xxx@10.2.0.72:5060>
Content-Type: application/sdp
Content-Length: 301

v=0
o=root 1670434890 1670434891 IN IP4 10.2.0.72
s=Firestick ICT PBX
c=IN IP4 10.2.0.72
t=0 0
m=image 0 tcp t38
m=image 4262 udptl t38
c=IN IP4 10.2.0.72
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:397
a=T38FaxUdpEC:t38UDPRedundancy

<------------>
[Dec 23 08:17:02] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:10.1.1.251:63825 --->
ACK sip:0xxx@10.2.0.72;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.251;branch=z9hG4bKTnn3yhpfmjzx~
Max-Forwards: 70
From: <sip:0298555555@10.1.1.251;user=phone>;tag=x'DdEH!r6N
To: "xxx" <sip:0xxx@10.2.0.72>;tag=as79a7cf9b
Call-ID: 41ca7d952d2727d910436c9e628f34f6@10.2.0.72:5060
CSeq: 1 ACK
Content-Length: 0


<------------>
[Dec 23 08:17:10] VERBOSE[2165] chan_sip.c: Scheduling destruction of SIP dialog '3f19c2f838864ae503ad59cb5601d8f7@10.2.0.71' in 32000 ms (Method: REGISTER)
[Dec 23 08:17:10] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:10.2.0.71:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.2.0.72:5060;branch=z9hG4bK7842913a;received=10.2.0.72
From: "asterisk" <sip:vs02trunk@10.2.0.72>;tag=as06e7825f
To: <sip:vs02trunk@10.2.0.71:5060>;tag=as44742d0c
Call-ID: 262e3dfb38c6c2ae77273fc7623ff5a5@10.2.0.72:5060
CSeq: 102 OPTIONS
Server: Firestick ICT PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------->
[Dec 23 08:17:10] VERBOSE[2165] chan_sip.c: --- (11 headers 0 lines) ---
[Dec 23 08:17:10] VERBOSE[2165] chan_sip.c: Really destroying SIP dialog '262e3dfb38c6c2ae77273fc7623ff5a5@10.2.0.72:5060' Method: OPTIONS
[Dec 23 08:17:11] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:x.25.112.19:5060 --->
BYE sip:61298555555@1.2.127.65:5060 SIP/2.0
Record-Route: <sip:x.25.112.19;lr;ftag=as0e8e4a75>
Via: SIP/2.0/UDP x.25.112.19:5060;branch=z9hG4bKbcd6.0cf2af44.0
Max-Forwards: 69
To: <sip:61298555555@x.25.112.19:5060>;tag=as34e60dbb
From: "61xxx"<sip:61xxx@x.25.112.19>;tag=as0e8e4a75
Call-ID: 6b9be3d0759011ce051c48484ff4e306@x.25.112.19:5050
CSeq: 103 BYE
User-Agent: VoIP Networks
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<------------->
[Dec 23 08:17:11] VERBOSE[2165] chan_sip.c: --- (12 headers 0 lines) ---
[Dec 23 08:17:11] VERBOSE[2165][C-00000001] chan_sip.c: Sending to x.25.112.19:5060 (no NAT)
[Dec 23 08:17:11] VERBOSE[2165][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog '6b9be3d0759011ce051c48484ff4e306@x.25.112.19:5050' in 6400 ms (Method: BYE)
[Dec 23 08:17:11] VERBOSE[2165][C-00000001] chan_sip.c: 
<--- Transmitting (no NAT) to x.25.112.19:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.25.112.19:5060;branch=z9hG4bKbcd6.0cf2af44.0;received=x.25.112.19
Record-Route: <sip:x.25.112.19;lr;ftag=as0e8e4a75>
From: "61xxx"<sip:61xxx@x.25.112.19>;tag=as0e8e4a75
To: <sip:61298555555@x.25.112.19:5060>;tag=as34e60dbb
Call-ID: 6b9be3d0759011ce051c48484ff4e306@x.25.112.19:5050
CSeq: 103 BYE
Server: Firestick ICT PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[Dec 23 08:17:11] VERBOSE[2173][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog '41ca7d952d2727d910436c9e628f34f6@10.2.0.72:5060' in 32000 ms (Method: ACK)
[Dec 23 08:17:11] VERBOSE[2173][C-00000001] chan_sip.c: set_destination: Parsing <sip:0298555555@10.1.1.251;user=phone> for address/port to send to
[Dec 23 08:17:11] VERBOSE[2173][C-00000001] chan_sip.c: set_destination: set destination to 10.1.1.251:5060
[Dec 23 08:17:11] VERBOSE[2173][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 10.1.1.251:5060:
BYE sip:0298555555@10.1.1.251;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.2.0.72:5060;branch=z9hG4bK7a9eac76
Max-Forwards: 70
From: "xxx" <sip:0xxx@10.2.0.72>;tag=as79a7cf9b
To: <sip:0298555555@10.1.1.251;user=phone>;tag=x'DdEH!r6N
Call-ID: 41ca7d952d2727d910436c9e628f34f6@10.2.0.72:5060
CSeq: 103 BYE
User-Agent: Firestick ICT PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Dec 23 08:17:11] VERBOSE[2173][C-00000001] pbx.c:   == Spawn extension (fax-receive, 0298555555, 6) exited non-zero on 'SIP/out-sip-00000002'
[Dec 23 08:17:11] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:10.1.1.251:63825 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.0.72:5060;branch=z9hG4bK7a9eac76
From: "xxx" <sip:0xxx@10.2.0.72>;tag=as79a7cf9b
To: <sip:0298555555@10.1.1.251;user=phone>;tag=x'DdEH!r6N;tag=yKfbKe57yU
Call-ID: 41ca7d952d2727d910436c9e628f34f6@10.2.0.72:5060
User-Agent: RicohIAF Version 2.0
CSeq: 103 BYE
Content-Length: 0

<------------->
[Dec 23 08:17:11] VERBOSE[2165] chan_sip.c: --- (8 headers 0 lines) ---
[Dec 23 08:17:11] VERBOSE[2165][C-00000001] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
[Dec 23 08:17:11] VERBOSE[2165] chan_sip.c: Really destroying SIP dialog '41ca7d952d2727d910436c9e628f34f6@10.2.0.72:5060' Method: ACK
[Dec 23 08:17:12] VERBOSE[2165] chan_sip.c: Really destroying SIP dialog '3af72bb3265a3c2f172a178c01789238@10.2.0.72:5060' Method: BYE
[Dec 23 08:17:15] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:10.0.6.7:5062 --->

Was over character limit to respond in one post.

Call 2 (not working):

<------------->
[Dec 23 08:17:19] VERBOSE[2165] chan_sip.c: --- (11 headers 0 lines) ---
[Dec 23 08:17:19] VERBOSE[2165] chan_sip.c: Really destroying SIP dialog '166f6cb141d3999c6f77a0de3d8a44a9@10.2.0.72:5060' Method: OPTIONS
[Dec 23 08:17:19] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:x.25.112.19:5060 --->
INVITE sip:61298555555@1.2.127.65:5060 SIP/2.0
Record-Route: <sip:x.25.112.19;lr;ftag=as0240b44a;did=a1b.e262dcf4>
Via: SIP/2.0/UDP x.25.112.19:5060;branch=z9hG4bK9e92.57eea271.0
Max-Forwards: 69
To: <sip:61298555555@x.25.112.19:5060>
From: "61xxx"<sip:61xxx@x.25.112.19>;tag=as0240b44a
Call-ID: 0e4129be61afae572d7d5f87060e5d6c@x.25.112.19:5050
Contact: <sip:61xxx@x.25.112.19:5060>
CSeq: 102 INVITE
User-Agent: VoIP Networks
Date: Mon, 22 Dec 2014 21:17:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-voipnow-extension: out-sip
X-voipnow-pbx: e24fee848be09752172d
X-voipnow-infrastructureid: 14c3aabd36a5
X-voipnow-did: 61298555555
Content-Type: application/sdp
Content-Length: 309

v=0
o=root 1123896873 1123896873 IN IP4 x.25.112.19
s=Asterisk PBX 1.8.9.2
c=IN IP4 x.25.112.19
t=0 0
m=audio 16348 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/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
a=sendrecv
<------------->
[Dec 23 08:17:19] VERBOSE[2165] chan_sip.c: --- (19 headers 14 lines) ---
[Dec 23 08:17:19] VERBOSE[2165] chan_sip.c: Sending to x.25.112.19:5060 (no NAT)
[Dec 23 08:17:19] VERBOSE[2165][C-00000002] chan_sip.c: Sending to x.25.112.19:5060 (no NAT)
[Dec 23 08:17:19] VERBOSE[2165][C-00000002] chan_sip.c: Using INVITE request as basis request - 0e4129be61afae572d7d5f87060e5d6c@x.25.112.19:5050
[Dec 23 08:17:19] VERBOSE[2165][C-00000002] chan_sip.c: Found peer 'out-sip' for '61xxx' from x.25.112.19:5060
[Dec 23 08:17:19] VERBOSE[2165][C-00000002] netsock2.c:   == Using SIP RTP CoS mark 5
[Dec 23 08:17:19] VERBOSE[2165][C-00000002] chan_sip.c: Found RTP audio format 0
[Dec 23 08:17:19] VERBOSE[2165][C-00000002] chan_sip.c: Found RTP audio format 8
[Dec 23 08:17:19] VERBOSE[2165][C-00000002] chan_sip.c: Found RTP audio format 18
[Dec 23 08:17:19] VERBOSE[2165][C-00000002] chan_sip.c: Found RTP audio format 101
[Dec 23 08:17:19] VERBOSE[2165][C-00000002] chan_sip.c: Found audio description format PCMU for ID 0
[Dec 23 08:17:19] VERBOSE[2165][C-00000002] chan_sip.c: Found audio description format PCMA for ID 8
[Dec 23 08:17:19] VERBOSE[2165][C-00000002] chan_sip.c: Found audio description format G729 for ID 18
[Dec 23 08:17:19] VERBOSE[2165][C-00000002] chan_sip.c: Found audio description format telephone-event for ID 101
[Dec 23 08:17:19] VERBOSE[2165][C-00000002] chan_sip.c: Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[Dec 23 08:17:19] VERBOSE[2165][C-00000002] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Dec 23 08:17:19] VERBOSE[2165][C-00000002] chan_sip.c: Peer audio RTP is at port x.25.112.19:16348
[Dec 23 08:17:19] VERBOSE[2165][C-00000002] chan_sip.c: Looking for 61298555555 in inbound (domain 1.2.127.65)
[Dec 23 08:17:19] VERBOSE[2165][C-00000002] chan_sip.c: list_route: hop: <sip:x.25.112.19;lr;ftag=as0240b44a;did=a1b.e262dcf4>
[Dec 23 08:17:19] VERBOSE[2165][C-00000002] chan_sip.c: 
<--- Transmitting (no NAT) to x.25.112.19:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP x.25.112.19:5060;branch=z9hG4bK9e92.57eea271.0;received=x.25.112.19
Record-Route: <sip:x.25.112.19;lr;ftag=as0240b44a;did=a1b.e262dcf4>
From: "61xxx"<sip:61xxx@x.25.112.19>;tag=as0240b44a
To: <sip:61298555555@x.25.112.19:5060>
Call-ID: 0e4129be61afae572d7d5f87060e5d6c@x.25.112.19:5050
CSeq: 102 INVITE
Server: Firestick ICT PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:61298555555@1.2.127.65:5060>
Content-Length: 0


<------------>
[Dec 23 08:17:19] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [61298555555@inbound:1] NoOp("SIP/out-sip-00000004", "") in new stack
[Dec 23 08:17:19] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [61298555555@inbound:2] Goto("SIP/out-sip-00000004", "inbound,0298555555,1") in new stack
[Dec 23 08:17:19] VERBOSE[2174][C-00000002] pbx.c:     -- Goto (inbound,0298555555,1)
[Dec 23 08:17:19] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@inbound:1] NoOp("SIP/out-sip-00000004", "61xxx 61xxx To 61298555555") in new stack
[Dec 23 08:17:19] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@inbound:2] Set("SIP/out-sip-00000004", "CALLERID(num)=0xxx") in new stack
[Dec 23 08:17:19] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@inbound:3] GotoIf("SIP/out-sip-00000004", "0?anonymous:") in new stack
[Dec 23 08:17:19] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@inbound:4] GotoIf("SIP/out-sip-00000004", "0?anonymous:") in new stack
[Dec 23 08:17:19] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@inbound:5] Set("SIP/out-sip-00000004", "CIDN=xxx") in new stack
[Dec 23 08:17:19] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@inbound:6] Set("SIP/out-sip-00000004", "CALLERID(name)=xxx") in new stack
[Dec 23 08:17:19] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@inbound:7] Set("SIP/out-sip-00000004", "CIDFOUND=found") in new stack
[Dec 23 08:17:19] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@inbound:8] Set("SIP/out-sip-00000004", "DB(cid/staff)=11917") in new stack
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@inbound:9] Set("SIP/out-sip-00000004", "DB(cid/stafft)=7") in new stack
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@inbound:10] GotoIf("SIP/out-sip-00000004", "1?found:") in new stack
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] pbx.c:     -- Goto (inbound,0298555555,30)
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@inbound:30] NoOp("SIP/out-sip-00000004", "callerid name xxx") in new stack
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@inbound:31] Set("SIP/out-sip-00000004", "PCID2=xxx") in new stack
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@inbound:32] Set("SIP/out-sip-00000004", "CALLERID(name)=xxx") in new stack
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@inbound:33] Set("SIP/out-sip-00000004", "CALLERID(name)=xxx") in new stack
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@inbound:34] NoOp("SIP/out-sip-00000004", "xxx") in new stack
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@inbound:35] NoOp("SIP/out-sip-00000004", "") in new stack
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@inbound:36] Goto("SIP/out-sip-00000004", "incoming,0298555555,1") in new stack
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] pbx.c:     -- Goto (incoming,0298555555,1)
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@incoming:1] Goto("SIP/out-sip-00000004", "fax-receive,0298555555,1") in new stack
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] pbx.c:     -- Goto (fax-receive,0298555555,1)
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@fax-receive:1] NoOp("SIP/out-sip-00000004", "") in new stack
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@fax-receive:2] NoOp("SIP/out-sip-00000004", "0") in new stack
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@fax-receive:3] Set("SIP/out-sip-00000004", "FAXOPT(gateway)=yes") in new stack
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@fax-receive:4] Set("SIP/out-sip-00000004", "CHANNEL(t38passthrough)=yes") in new stack
[Dec 23 08:17:20] WARNING[2174][C-00000002] func_channel.c: Unknown or unavailable item requested: 't38passthrough'
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@fax-receive:5] NoOp("SIP/out-sip-00000004", "0") in new stack
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] pbx.c:     -- Executing [0298555555@fax-receive:6] Dial("SIP/out-sip-00000004", "SIP/0298555555") in new stack
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] netsock2.c:   == Using SIP RTP CoS mark 5
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: We think we can do text
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Audio is at 16912
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100001 (g723) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100002 (gsm) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100002 (gsm) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100005 (g726aal2) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100006 (adpcm) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100007 (lpc10) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100008 (g729) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100009 (speex) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100010 (ilbc) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100011 (g726) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100012 (g722) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100013 (siren7) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100014 (siren14) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100015 (g719) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100016 (speex16) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100017 (testlaw) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100017 (testlaw) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100019 (slin) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100020 (slin12) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100021 (slin16) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100022 (slin24) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100023 (slin32) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100024 (slin44) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100025 (slin48) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100026 (slin96) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100027 (slin192) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100028 (speex32) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Reliably Transmitting (no NAT) to 10.1.1.251:5060:
INVITE sip:0298555555@10.1.1.251;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.2.0.72:5060;branch=z9hG4bK6787078e
Max-Forwards: 70
From: "xxx" <sip:0xxx@10.2.0.72>;tag=as713ca014
To: <sip:0298555555@10.1.1.251;user=phone>
Contact: <sip:0xxx@10.2.0.72:5060>
Call-ID: 1d47ec1566ec142936f37d6c55199833@10.2.0.72:5060
CSeq: 102 INVITE
User-Agent: Firestick ICT PBX
Date: Mon, 22 Dec 2014 21:17:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "xxx" <sip:0xxx@10.2.0.72>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 904

v=0
o=root 1522758492 1522758492 IN IP4 10.2.0.72
s=Firestick ICT PBX
c=IN IP4 10.2.0.72
t=0 0
m=audio 16912 RTP/AVP 0 4 3 3 8 8 112 5 7 18 110 97 111 9 102 115 116 117 10 118 119 101
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:102 G7221/16000
a=fmtp:102 bitrate=32000
a=rtpmap:115 G7221/32000
a=fmtp:115 bitrate=48000
a=rtpmap:116 G719/48000
a=fmtp:116 bitrate=64000
a=rtpmap:117 speex/16000
a=rtpmap:10 L16/8000
a=rtpmap:118 L16/16000
a=rtpmap:119 speex/32000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] app_dial.c:     -- Called SIP/0298555555
[Dec 23 08:17:20] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:10.1.1.251:63823 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.2.0.72:5060;branch=z9hG4bK6787078e
From: "xxx" <sip:0xxx@10.2.0.72>;tag=as713ca014
To: <sip:0298555555@10.1.1.251;user=phone>;tag=_'Lis!N70S
Call-ID: 1d47ec1566ec142936f37d6c55199833@10.2.0.72:5060
User-Agent: RicohIAF Version 2.0
CSeq: 102 INVITE
Content-Length: 0

<------------->
[Dec 23 08:17:20] VERBOSE[2165] chan_sip.c: --- (8 headers 0 lines) ---
[Dec 23 08:17:20] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:10.1.1.251:63823 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.2.0.72:5060;branch=z9hG4bK6787078e
From: "xxx" <sip:0xxx@10.2.0.72>;tag=as713ca014
To: <sip:0298555555@10.1.1.251;user=phone>;tag=_'Lis!N70S
Call-ID: 1d47ec1566ec142936f37d6c55199833@10.2.0.72:5060
User-Agent: RicohIAF Version 2.0
CSeq: 102 INVITE
Content-Type: application/sdp
Content-Length: 153

v=0
o=RICOH-SIP-IPFAX 1419282943 1419282943 IN IP4 10.1.1.251
s=Session SDP
t=0 0
m=audio 5004 RTP/AVP 0
c=IN IP4 10.1.1.251
a=rtpmap:0 PCMU/8000
<------------->
[Dec 23 08:17:20] VERBOSE[2165] chan_sip.c: --- (9 headers 7 lines) ---
[Dec 23 08:17:20] VERBOSE[2165][C-00000002] chan_sip.c: list_route: no route
[Dec 23 08:17:20] VERBOSE[2165][C-00000002] chan_sip.c: Found RTP audio format 0
[Dec 23 08:17:20] VERBOSE[2165][C-00000002] chan_sip.c: Found audio description format PCMU for ID 0
[Dec 23 08:17:20] VERBOSE[2165][C-00000002] chan_sip.c: Capabilities: us - (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|speex16|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|red|t140|siren7|siren14|testlaw|g719|speex32|slin12|slin24|slin32|slin44|slin48|slin96|slin192), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
[Dec 23 08:17:20] VERBOSE[2165][C-00000002] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Dec 23 08:17:20] VERBOSE[2165][C-00000002] chan_sip.c: Peer audio RTP is at port 10.1.1.251:5004
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] app_dial.c:     -- SIP/0298555555-00000005 is ringing
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: 
<--- Transmitting (no NAT) to x.25.112.19:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP x.25.112.19:5060;branch=z9hG4bK9e92.57eea271.0;received=x.25.112.19
Record-Route: <sip:x.25.112.19;lr;ftag=as0240b44a;did=a1b.e262dcf4>
From: "61xxx"<sip:61xxx@x.25.112.19>;tag=as0240b44a
To: <sip:61298555555@x.25.112.19:5060>;tag=as35bc6e53
Call-ID: 0e4129be61afae572d7d5f87060e5d6c@x.25.112.19:5050
CSeq: 102 INVITE
Server: Firestick ICT PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:61298555555@1.2.127.65:5060>
Content-Length: 0


<------------>
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] app_dial.c:     -- SIP/0298555555-00000005 is making progress passing it to SIP/out-sip-00000004
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Audio is at 12592
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: 
<--- Transmitting (no NAT) to x.25.112.19:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP x.25.112.19:5060;branch=z9hG4bK9e92.57eea271.0;received=x.25.112.19
Record-Route: <sip:x.25.112.19;lr;ftag=as0240b44a;did=a1b.e262dcf4>
From: "61xxx"<sip:61xxx@x.25.112.19>;tag=as0240b44a
To: <sip:61298555555@x.25.112.19:5060>;tag=as35bc6e53
Call-ID: 0e4129be61afae572d7d5f87060e5d6c@x.25.112.19:5050
CSeq: 102 INVITE
Server: Firestick ICT PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:61298555555@1.2.127.65:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 255

v=0
o=root 183233980 183233980 IN IP4 1.2.127.65
s=Firestick ICT PBX
c=IN IP4 1.2.127.65
t=0 0
m=audio 12592 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[Dec 23 08:17:20] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:10.1.1.251:63823 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.2.0.72:5060;branch=z9hG4bK6787078e
From: "xxx" <sip:0xxx@10.2.0.72>;tag=as713ca014
To: <sip:0298555555@10.1.1.251;user=phone>;tag=_'Lis!N70S
Call-ID: 1d47ec1566ec142936f37d6c55199833@10.2.0.72:5060
User-Agent: RicohIAF Version 2.0
CSeq: 102 INVITE
Contact: <sip:0298555555@10.1.1.251;user=phone>
Content-Type: application/sdp
Content-Length: 153

v=0
o=RICOH-SIP-IPFAX 1419282943 1419282943 IN IP4 10.1.1.251
s=Session SDP
t=0 0
m=audio 5004 RTP/AVP 0
c=IN IP4 10.1.1.251
a=rtpmap:0 PCMU/8000
<------------->
[Dec 23 08:17:20] VERBOSE[2165] chan_sip.c: --- (10 headers 7 lines) ---
[Dec 23 08:17:20] VERBOSE[2165][C-00000002] chan_sip.c: list_route: hop: <sip:0298555555@10.1.1.251;user=phone>
[Dec 23 08:17:20] VERBOSE[2165][C-00000002] chan_sip.c: set_destination: Parsing <sip:0298555555@10.1.1.251;user=phone> for address/port to send to
[Dec 23 08:17:20] VERBOSE[2165][C-00000002] chan_sip.c: set_destination: set destination to 10.1.1.251:5060
[Dec 23 08:17:20] VERBOSE[2165][C-00000002] chan_sip.c: Transmitting (no NAT) to 10.1.1.251:5060:
ACK sip:0298555555@10.1.1.251;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.2.0.72:5060;branch=z9hG4bK5e3c75d2
Max-Forwards: 70
From: "xxx" <sip:0xxx@10.2.0.72>;tag=as713ca014
To: <sip:0298555555@10.1.1.251;user=phone>;tag=_'Lis!N70S
Contact: <sip:0xxx@10.2.0.72:5060>
Call-ID: 1d47ec1566ec142936f37d6c55199833@10.2.0.72:5060
CSeq: 102 ACK
User-Agent: Firestick ICT PBX
Content-Length: 0


---
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] app_dial.c:     -- SIP/0298555555-00000005 answered SIP/out-sip-00000004
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Audio is at 12592
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: 
<--- Reliably Transmitting (no NAT) to x.25.112.19:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP x.25.112.19:5060;branch=z9hG4bK9e92.57eea271.0;received=x.25.112.19
Record-Route: <sip:x.25.112.19;lr;ftag=as0240b44a;did=a1b.e262dcf4>
From: "61xxx"<sip:61xxx@x.25.112.19>;tag=as0240b44a
To: <sip:61298555555@x.25.112.19:5060>;tag=as35bc6e53
Call-ID: 0e4129be61afae572d7d5f87060e5d6c@x.25.112.19:5050
CSeq: 102 INVITE
Server: Firestick ICT PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:61298555555@1.2.127.65:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 255

v=0
o=root 183233980 183233980 IN IP4 1.2.127.65
s=Firestick ICT PBX
c=IN IP4 1.2.127.65
t=0 0
m=audio 12592 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[Dec 23 08:17:20] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:x.25.112.19:5060 --->
ACK sip:61298555555@1.2.127.65:5060 SIP/2.0
Via: SIP/2.0/UDP x.25.112.19:5060;branch=z9hG4bKcydzigwkX
Max-Forwards: 69
To: <sip:61298555555@x.25.112.19:5060>;tag=as35bc6e53
From: "61xxx"<sip:61xxx@x.25.112.19>;tag=as0240b44a
Call-ID: 0e4129be61afae572d7d5f87060e5d6c@x.25.112.19:5050
Contact: <sip:61xxx@x.25.112.19:5060>
CSeq: 102 ACK
User-Agent: VoIP Networks
Content-Length: 0

<------------->
[Dec 23 08:17:20] VERBOSE[2165] chan_sip.c: --- (10 headers 0 lines) ---
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] res_rtp_asterisk.c:        > 0x7f35b4006390 -- Probation passed - setting RTP source address to x.25.112.19:16348
[Dec 23 08:17:20] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:10.1.1.251:63823 --->
INVITE sip:0xxx@10.2.0.72;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.251;branch=z9hG4bKhAfPKs3*~J5WS
Max-Forwards: 70
From: <sip:0298555555@10.1.1.251;user=phone>;tag=_'Lis!N70S
To: "xxx" <sip:0xxx@10.2.0.72>;tag=as713ca014
Call-ID: 1d47ec1566ec142936f37d6c55199833@10.2.0.72:5060
User-Agent: RicohIAF Version 2.0
CSeq: 1 INVITE
Contact: <sip:0298555555@10.1.1.251;user=phone>
Content-Type: application/sdp
Content-Length: 408

v=0
o=RICOH-SIP-IPFAX 1419282943 1419282943 IN IP4 10.1.1.251
s=Session SDP
t=0 0
m=image 49152 tcp t38
c=IN IP4 10.1.1.251
a=T38FaxVersion:0
a=T38MaxBitRate:144
a=T38FaxRateManagement:localTCF
a=T38VendorInfo:0 0 37
m=image 49152 udptl t38
c=IN IP4 10.1.1.251
a=T38FaxVersion:0
a=T38MaxBitRate:144
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
a=T38VendorInfo:0 0 37
<------------->
[Dec 23 08:17:20] VERBOSE[2165] chan_sip.c: --- (11 headers 17 lines) ---
[Dec 23 08:17:20] VERBOSE[2165][C-00000002] chan_sip.c: Sending to 10.1.1.251:5060 (no NAT)
[Dec 23 08:17:20] VERBOSE[2165][C-00000002] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.251:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.251;branch=z9hG4bKhAfPKs3*~J5WS;received=10.1.1.251
From: <sip:0298555555@10.1.1.251;user=phone>;tag=_'Lis!N70S
To: "xxx" <sip:0xxx@10.2.0.72>;tag=as713ca014
Call-ID: 1d47ec1566ec142936f37d6c55199833@10.2.0.72:5060
CSeq: 1 INVITE
Server: Firestick ICT PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0xxx@10.2.0.72:5060>
Content-Length: 0


<------------>
[Dec 23 08:17:20] VERBOSE[2165][C-00000002] chan_sip.c: Audio is at 16912
[Dec 23 08:17:20] VERBOSE[2165][C-00000002] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Dec 23 08:17:20] VERBOSE[2165][C-00000002] chan_sip.c: 
<--- Reliably Transmitting (no NAT) to 10.1.1.251:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.251;branch=z9hG4bKhAfPKs3*~J5WS;received=10.1.1.251
From: <sip:0298555555@10.1.1.251;user=phone>;tag=_'Lis!N70S
To: "xxx" <sip:0xxx@10.2.0.72>;tag=as713ca014
Call-ID: 1d47ec1566ec142936f37d6c55199833@10.2.0.72:5060
CSeq: 1 INVITE
Server: Firestick ICT PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:0xxx@10.2.0.72:5060>
Content-Type: application/sdp
Content-Length: 171

v=0
o=root 1522758492 1522758492 IN IP4 10.2.0.72
s=Firestick ICT PBX
c=IN IP4 10.2.0.72
t=0 0
m=audio 16912 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

<------------>
[Dec 23 08:17:20] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:10.1.1.251:63823 --->
ACK sip:0xxx@10.2.0.72;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.251;branch=z9hG4bKY3JNDa'3A7KiN
Max-Forwards: 70
From: <sip:0298555555@10.1.1.251;user=phone>;tag=_'Lis!N70S
To: "xxx" <sip:0xxx@10.2.0.72>;tag=as713ca014
Call-ID: 1d47ec1566ec142936f37d6c55199833@10.2.0.72:5060
CSeq: 1 ACK
Content-Length: 0

<------------->
[Dec 23 08:17:20] VERBOSE[2165] chan_sip.c: --- (8 headers 0 lines) ---
[Dec 23 08:17:20] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:10.1.1.251:63823 --->
BYE sip:0xxx@10.2.0.72 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.251;branch=z9hG4bKwoqT'x1h~+gs2
Max-Forwards: 70
From: <sip:0298555555@10.1.1.251;user=phone>;tag=_'Lis!N70S
To: "xxx" <sip:0xxx@10.2.0.72>;tag=as713ca014
Call-ID: 1d47ec1566ec142936f37d6c55199833@10.2.0.72:5060
CSeq: 2 BYE
Content-Length: 0

<------------->
[Dec 23 08:17:20] VERBOSE[2165] chan_sip.c: --- (8 headers 0 lines) ---
[Dec 23 08:17:20] VERBOSE[2165][C-00000002] chan_sip.c: Sending to 10.1.1.251:5060 (no NAT)
[Dec 23 08:17:20] VERBOSE[2165][C-00000002] chan_sip.c: Scheduling destruction of SIP dialog '1d47ec1566ec142936f37d6c55199833@10.2.0.72:5060' in 32000 ms (Method: BYE)
[Dec 23 08:17:20] VERBOSE[2165][C-00000002] chan_sip.c: 
<--- Transmitting (no NAT) to 10.1.1.251:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.251;branch=z9hG4bKwoqT'x1h~+gs2;received=10.1.1.251
From: <sip:0298555555@10.1.1.251;user=phone>;tag=_'Lis!N70S
To: "xxx" <sip:0xxx@10.2.0.72>;tag=as713ca014
Call-ID: 1d47ec1566ec142936f37d6c55199833@10.2.0.72:5060
CSeq: 2 BYE
Server: Firestick ICT PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] pbx.c:   == Spawn extension (fax-receive, 0298555555, 6) exited non-zero on 'SIP/out-sip-00000004'
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Scheduling destruction of SIP dialog '0e4129be61afae572d7d5f87060e5d6c@x.25.112.19:5050' in 6400 ms (Method: ACK)
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: set_destination: Parsing <sip:x.25.112.19;lr;ftag=as0240b44a;did=a1b.e262dcf4> for address/port to send to
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: set_destination: set destination to x.25.112.19:5060
[Dec 23 08:17:20] VERBOSE[2174][C-00000002] chan_sip.c: Reliably Transmitting (no NAT) to x.25.112.19:5060:
BYE sip:61xxx@x.25.112.19:5060 SIP/2.0
Via: SIP/2.0/UDP 1.2.127.65:5060;branch=z9hG4bK75ac21d2
Route: <sip:x.25.112.19;lr;ftag=as0240b44a;did=a1b.e262dcf4>
Max-Forwards: 70
From: <sip:61298555555@x.25.112.19:5060>;tag=as35bc6e53
To: "61xxx"<sip:61xxx@x.25.112.19>;tag=as0240b44a
Call-ID: 0e4129be61afae572d7d5f87060e5d6c@x.25.112.19:5050
CSeq: 102 BYE
User-Agent: Firestick ICT PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Dec 23 08:17:20] VERBOSE[2165] chan_sip.c: 
<--- SIP read from UDP:x.25.112.19:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 1.2.127.65:5060;branch=z9hG4bK75ac21d2
From: <sip:61298555555@x.25.112.19:5060>;tag=as35bc6e53
To: "61xxx"<sip:61xxx@x.25.112.19>;tag=as0240b44a
Call-ID: 0e4129be61afae572d7d5f87060e5d6c@x.25.112.19:5050
CSeq: 102 BYE
Server: VoIP Networks
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
[Dec 23 08:17:20] VERBOSE[2165] chan_sip.c: --- (10 headers 0 lines) ---
[Dec 23 08:17:20] VERBOSE[2165][C-00000002] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
[Dec 23 08:17:20] VERBOSE[2165] chan_sip.c: Really destroying SIP dialog '0e4129be61afae572d7d5f87060e5d6c@x.25.112.19:5050' Method: ACK

Set:

ignoresdpversion=yes

to work round your broken party B.

Thanks David, Love your work! That did indeed fix the issue