PJSIP, T.38 Fax Gateway
Hi list,
I’m trying to receive fax from PSTN, with the following setup:
Fax machine — PSTN — *11 — *13 — IAXmodem + Hylafax
Fax machine is connected to the PSTN, call arrives via ISDN on Asterisk
11.16.0 used as gateway, chan_sip relays the call to Asterisk 13.4.0
receiving via chan_pjsip. I’m trying to have T.38 working between the 2
Asterisk servers: I’ve done that with success with both Asterisk running
11, but I can’t make it work with Asterisk 13. I think the configuration is correct, as the traces below show that T.38 is negotiated correctly, but there is always only one UDPTL packet transmitted from Asterisk-13
to Asterisk-11: wireshark shows UDPTLPacket t30ind: no-signal
Is it a bug in chan_pjsip, or did I miss something?
Here is the SIP trace on the gateway:
== Primary D-Channel on span 2 up
— Accepting call from ‘40483527’ to ‘1041’ on channel 0/1, span 2
— Executing [1041@entrant_rnis:1] NoOp(“DAHDI/i2/40483527-18e”,
“Appel entrant sur ligne RNIS”) in new stack
— Executing [1041@entrant_rnis:2] Set(“DAHDI/i2/40483527-18e”,
“FAXOPT(gateway)=yes”) in new stack
— Executing [1041@entrant_rnis:3] Dial(“DAHDI/i2/40483527-18e”,
“SIP/tiare/1041”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 7740
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.200:5060:
INVITE sip:1041@192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK445324c9
Max-Forwards: 70
From:
To:
Contact:
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.16.0
Date: Thu, 09 Jul 2015 05:02:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer P-Asserted-Identity: “40483527”
Content-Type: application/sdp Content-Length: 233
v=0
o=root 687045483 687045483 IN IP4 192.168.0.10
s=Asterisk PBX 11.16.0
c=IN IP4 192.168.0.10
t=0 0
m=audio 7740 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
– —
— Called SIP/tiare/1041
<--- SIP read from UDP:192.168.0.200:5060 --->
SIP/2.0 100 Trying Via: SIP/2.0/UDP
192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK445324c
9
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
From:
To:
CSeq: 102 INVITE
Server: Asterisk GPL PBX
Content-Length: 0
<------------->
– — (8 headers 0 lines) —
<--- SIP read from UDP:192.168.0.200:5060 --->
SIP/2.0 180 Ringing Via: SIP/2.0/UDP
192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK445324c
9
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
From:
To:
CSeq: 102 INVITE
Server: Asterisk GPL PBX
Contact:
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Content-Length: 0
<------------->
– — (10 headers 0 lines) —
list_route: hop:
— SIP/tiare-00000165 is ringing
<--- SIP read from UDP:192.168.0.200:5060 --->
OPTIONS sip:tiare@gw.sysnux.pf:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.200:5060;rport;branch=z9hG4bKPj76f3fab9-843a-46c5-a840-33411a7
ff33b From:
14-4365-a46e-1678844bee7d To:
Contact:
Call-ID: b5dab0f5-d07b-461b-aa16-a5a9aa93369a CSeq: 22129 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk GPL PBX
Content-Length: 0
<------------->
– — (10 headers 0 lines) —
Sending to 192.168.0.200:5060 (no NAT)
Looking for tiare in none (domain gw.sysnux.pf)
<--- Transmitting (no NAT) to 192.168.0.200:5060 --->
SIP/2.0 404 Not Found Via: SIP/2.0/UDP
192.168.0.200:5060;branch=z9hG4bKPj76f3fab9-843a-46c5-a840-33411a7ff33b;
received=192.168.0.200;rport=5060
From:
14-4365-a46e-1678844bee7d To:
Server: Asterisk PBX 11.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer Accept: application/sdp Content-Length: 0
<------------>
Scheduling destruction of SIP dialog
‘b5dab0f5-d07b-461b-aa16-a5a9aa93369a’ in 32000 ms (Method: OPTIONS)
<--- SIP read from UDP:192.168.0.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK445324c
9
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
From:
To:
CSeq: 102 INVITE
Server: Asterisk GPL PBX
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Contact:
Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 237
v=0
o=- 687045483 687045485 IN IP4 192.168.0.200
s=Asterisk c=IN IP4 192.168.0.200
t=0 0
m=audio 25198 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<------------->
– — (12 headers 12 lines) —
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us – (alaw), peer –
audio=(alaw)/video=(nothing)/text=(nothing), combined – (alaw)
Non-codec capabilities (dtmf): us – 0x1 (telephone-event|), peer – 0x1
(telephone-event|), combined – 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.200:25198
list_route: hop:
set_destination: Parsing
Transmitting (no NAT) to 192.168.0.200:5060:
ACK sip:192.168.0.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK076152f9
Max-Forwards: 70
From:
To:
Contact:
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.16.0
Content-Length: 0
– —
— SIP/tiare-00000165 answered DAHDI/i2/40483527-18e
— Channel 5 detected a CED tone towards the network.
<--- SIP read from UDP:192.168.0.200:5060 --->
INVITE sip:40483527@192.168.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.200:5060;rport;branch=z9hG4bKPjda7d30ac-2b5f-4fa1-93dd-e97cfe5
fd3df From:
To:
Contact:
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
CSeq: 24271 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk GPL PBX
Content-Type: application/sdp Content-Length: 249
v=0
o=- 687045483 687045486 IN IP4 192.168.0.200
s=Asterisk c=IN IP4 192.168.0.200
t=0 0
m=image 4127 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:849
a=T38FaxUdpEC:t38UDPFEC
<------------->
– — (15 headers 11 lines) —
Sending to 192.168.0.200:5060 (no NAT)
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Got T.38 offer in SDP in dialog
622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
Capabilities: us – (alaw), peer –
audio=(nothing)/video=(nothing)/text=(nothing), combined – (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 (no NAT) to 192.168.0.200:5060 --->
SIP/2.0 100 Trying Via: SIP/2.0/UDP
192.168.0.200:5060;branch=z9hG4bKPjda7d30ac-2b5f-4fa1-93dd-e97cfe5fd3df;
received=192.168.0.200;rport=5060
From:
To:
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
CSeq: 24271 INVITE
Server: Asterisk PBX 11.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer Contact:
Content-Length: 0
<------------>
<--- Reliably Transmitting (no NAT) to 192.168.0.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.200:5060;branch=z9hG4bKPjda7d30ac-2b5f-4fa1-93dd-e97cfe5fd3df;
received=192.168.0.200;rport=5060
From:
To:
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
CSeq: 24271 INVITE
Server: Asterisk PBX 11.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer Contact:
Content-Type: application/sdp Content-Length: 285
v=0
o=root 687045483 687045484 IN IP4 192.168.0.10
s=Asterisk PBX 11.16.0
c=IN IP4 192.168.0.10
t=0 0
m=image 4617 udptl t38
c=IN IP4 192.168.0.10
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:849
a=T38FaxUdpEC:t38UDPFEC
<------------>
<--- SIP read from UDP:192.168.0.200:5060 --->
ACK sip:40483527@192.168.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.200:5060;rport;branch=z9hG4bKPj935b2ccd-b675-46b6-8b31-763d80f d9574
From:
To:
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
CSeq: 24271 ACK
Max-Forwards: 70
User-Agent: Asterisk GPL PBX
Content-Length: 0
<------------->
– — (9 headers 0 lines) —
UDPTL (SIP/tiare-00000165): packet from 192.168.0.200:4127 (seq 0, len
8)
Reliably Transmitting (no NAT) to 192.168.0.200:5060:
OPTIONS sip:192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK72712ff5
Max-Forwards: 70
From: “asterisk”
To:
Contact:
Call-ID: 21f0b6115e7d811a6a399b77424cb2b7@192.168.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.16.0
Date: Thu, 09 Jul 2015 05:02:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer Content-Length: 0
– —
<--- SIP read from UDP:192.168.0.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK72712ff
5
Call-ID: 21f0b6115e7d811a6a399b77424cb2b7@192.168.0.10:5060
From: “asterisk”
To:
CSeq: 102 OPTIONS
Accept: application/sdp, application/pidf+xml, application/simple-message-summary, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, message/sipfrag;version=2.0
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub Accept-Encoding: text/plain Accept-Language: en Server: Asterisk GPL PBX
Content-Length: 0
<------------->
– — (13 headers 0 lines) —
Really destroying SIP dialog
’21f0b6115e7d811a6a399b77424cb2b7@192.168.0.10:5060′ Method: OPTIONS
Really destroying SIP dialog ‘b5dab0f5-d07b-461b-aa16-a5a9aa93369a’
Method: OPTIONS
— Span 2: Channel 0/1 got hangup request, cause 16
Scheduling destruction of SIP dialog
‘622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060’ in 6400 ms (Method:
ACK)
set_destination: Parsing
Reliably Transmitting (no NAT) to 192.168.0.200:5060:
BYE sip:192.168.0.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK66a545b1;rport Max-Forwards: 70
From:
To:
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 11.16.0
X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16
Content-Length: 0
– —
<--- SIP read from UDP:192.168.0.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK66a545b
1
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
From:
To:
CSeq: 103 BYE
Server: Asterisk GPL PBX
Content-Length: 0
<------------->
– — (8 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog
‘622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060’ Method: ACK
== Spawn extension (entrant_rnis, 1041, 3) exited non-zero on
‘DAHDI/i2/40483527-18e’
— Hungup ‘DAHDI/i2/40483527-18e’
************************************************************************
***
And here is what happens on Asterisk 13:
<--- Received SIP request (857 bytes) from UDP:192.168.0.10:5060 --->
INVITE sip:1041@192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK445324c9
Max-Forwards: 70
From:
To:
Contact:
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.16.0
Date: Thu, 09 Jul 2015 05:02:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer P-Asserted-Identity: “40483527”
Content-Type: application/sdp Content-Length: 233
v=0
o=root 687045483 687045483 IN IP4 192.168.0.10
s=Asterisk PBX 11.16.0
c=IN IP4 192.168.0.10
t=0 0
m=audio 7740 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<--- Transmitting SIP response (319 bytes) to UDP:192.168.0.10:5060 --->
SIP/2.0 100 Trying Via: SIP/2.0/UDP
192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK445324c
9
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
From:
To:
CSeq: 102 INVITE
Server: Asterisk GPL PBX
Content-Length: 0
— Executing [1041@incoming:1] Gosub(“PJSIP/t0gw-0000073b”,
“stdexten,100,1”) in new stack
— Executing [100@stdexten:1] NoOp(“PJSIP/t0gw-0000073b”, “STDEXTEN
100”) in new stack
— Executing [100@stdexten:2] Set(“PJSIP/t0gw-0000073b”,
“sip=fqygGSWm”) in new stack
— Executing [100@stdexten:3] GotoIf(“PJSIP/t0gw-0000073b”,
“1?sip_ok”) in new stack
— Goto (stdexten,100,5)
— Executing [100@stdexten:5] Set(“PJSIP/t0gw-0000073b”, “ext=100”)
in new stack
— Executing [100@stdexten:6] Set(“PJSIP/t0gw-0000073b”,
“FAXOPT(gateway)=yes”) in new stack
— Executing [100@stdexten:7] Set(“PJSIP/t0gw-0000073b”,
“FAXOPT(faxdetect)=yes”) in new stack
— Executing [100@stdexten:8] Set(“PJSIP/t0gw-0000073b”, “cfvm=”) in new stack
— Executing [100@stdexten:9] GotoIf(“PJSIP/t0gw-0000073b”,
“?:nocfvm”) in new stack
— Goto (stdexten,100,12)
— Executing [100@stdexten:12] Set(“PJSIP/t0gw-0000073b”, “cfim=”)
in new stack
— Executing [100@stdexten:13] GotoIf(“PJSIP/t0gw-0000073b”,
“0?P/t0gw-0,,1”) in new stack
— Executing [100@stdexten:14] GotoIf(“PJSIP/t0gw-0000073b”,
“?:nocfim”) in new stack
— Goto (stdexten,100,19)
— Executing [100@stdexten:19] Set(“PJSIP/t0gw-0000073b”,
“sip=fqygGSWm”) in new stack
— Executing [100@stdexten:20] Dial(“PJSIP/t0gw-0000073b”,
“PJSIP/fqygGSWm,25”) in new stack
— Called PJSIP/fqygGSWm
— PJSIP/fqygGSWm-0000073c is ringing
<--- Transmitting SIP response (507 bytes) to UDP:192.168.0.10:5060 --->
SIP/2.0 180 Ringing Via: SIP/2.0/UDP
192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK445324c
9
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
From:
To:
CSeq: 102 INVITE
Server: Asterisk GPL PBX
Contact:
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Content-Length: 0
<--- Transmitting SIP request (486 bytes) to UDP:192.168.0.10:5060 --->
OPTIONS sip:tiare@gw.sysnux.pf:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.200:5060;rport;branch=z9hG4bKPj76f3fab9-843a-46c5-a840-33411a7
ff33b From:
14-4365-a46e-1678844bee7d To:
Contact:
Call-ID: b5dab0f5-d07b-461b-aa16-a5a9aa93369a CSeq: 22129 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk GPL PBX
Content-Length: 0
<--- Received SIP response (561 bytes) from UDP:192.168.0.10:5060 --->
SIP/2.0 404 Not Found Via: SIP/2.0/UDP
192.168.0.200:5060;branch=z9hG4bKPj76f3fab9-843a-46c5-a840-33411a7ff33b;
received=192.168.0.200;rport=5060
From:
14-4365-a46e-1678844bee7d To:
Server: Asterisk PBX 11.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer Accept: application/sdp Content-Length: 0
— PJSIP/fqygGSWm-0000073c answered PJSIP/t0gw-0000073b
<--- Transmitting SIP response (821 bytes) to UDP:192.168.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK445324c
9
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
From:
To:
CSeq: 102 INVITE
Server: Asterisk GPL PBX
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Contact:
Supported: 100rel, timer, replaces, norefersub Content-Type: application/sdp Content-Length: 237
v=0
o=- 687045483 687045485 IN IP4 192.168.0.200
s=Asterisk c=IN IP4 192.168.0.200
t=0 0
m=audio 25198 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
— Channel PJSIP/t0gw-0000073b joined ‘simple_bridge’ basic-bridge
<09793536-f012-4d31-a293-8df93639b90c>
— Channel PJSIP/fqygGSWm-0000073c joined ‘simple_bridge’
basic-bridge <09793536-f012-4d31-a293-8df93639b90c>
<--- Received SIP request (408 bytes) from UDP:192.168.0.10:5060 --->
ACK sip:192.168.0.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK076152f9
Max-Forwards: 70
From:
To:
Contact:
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.16.0
Content-Length: 0
== Redirecting ‘PJSIP/t0gw-0000073b’ to fax extension due to CNG detec tion
— Channel PJSIP/t0gw-0000073b left ‘simple_bridge’ basic-bridge
<09793536-f012-4d31-a293-8df93639b90c>
— Channel PJSIP/fqygGSWm-0000073c left ‘simple_bridge’ basic-bridge
<09793536-f012-4d31-a293-8df93639b90c>
— Executing [fax@stdexten:1] NoOp(“PJSIP/t0gw-0000073b”, “FAXIN
(100) “” <40483527> -> <> <> <40483527> <0>“) in new stack
— Executing [fax@stdexten:2] Dial(“PJSIP/t0gw-0000073b”,
“IAX2/iaxmodem0/100”) in new stack
— Called IAX2/iaxmodem0/100
— Call accepted by 127.0.0.1:4570 (format alaw)
— Format for call is (alaw)
— IAX2/iaxmodem0-7773 is ringing
— IAX2/iaxmodem0-7773 answered PJSIP/t0gw-0000073b
— Channel PJSIP/t0gw-0000073b joined ‘simple_bridge’ basic-bridge
<0328da69-07f5-4270-8fa4-8178649c9906>
— Channel IAX2/iaxmodem0-7773 joined ‘simple_bridge’ basic-bridge
<0328da69-07f5-4270-8fa4-8178649c9906>
<--- Transmitting SIP request (927 bytes) to UDP:192.168.0.10:5060 --->
INVITE sip:40483527@192.168.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.200:5060;rport;branch=z9hG4bKPjda7d30ac-2b5f-4fa1-93dd-e97cfe5
fd3df From:
To:
Contact:
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
CSeq: 24271 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk GPL PBX
Content-Type: application/sdp Content-Length: 249
v=0
o=- 687045483 687045486 IN IP4 192.168.0.200
s=Asterisk c=IN IP4 192.168.0.200
t=0 0
m=image 4127 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:849
a=T38FaxUdpEC:t38UDPFEC
<--- Received SIP response (560 bytes) from UDP:192.168.0.10:5060 --->
SIP/2.0 100 Trying Via: SIP/2.0/UDP
192.168.0.200:5060;branch=z9hG4bKPjda7d30ac-2b5f-4fa1-93dd-e97cfe5fd3df;
received=192.168.0.200;rport=5060
From:
To:
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
CSeq: 24271 INVITE
Server: Asterisk PBX 11.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer Contact:
Content-Length: 0
<--- Received SIP response (874 bytes) from UDP:192.168.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.200:5060;branch=z9hG4bKPjda7d30ac-2b5f-4fa1-93dd-e97cfe5fd3df;
received=192.168.0.200;rport=5060
From:
To:
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
CSeq: 24271 INVITE
Server: Asterisk PBX 11.16.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer Contact:
Content-Type: application/sdp Content-Length: 285
v=0
o=root 687045483 687045484 IN IP4 192.168.0.10
s=Asterisk PBX 11.16.0
c=IN IP4 192.168.0.10
t=0 0
m=image 4617 udptl t38
c=IN IP4 192.168.0.10
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:849
a=T38FaxUdpEC:t38UDPFEC
<--- Transmitting SIP request (409 bytes) to UDP:192.168.0.10:5060 --->
ACK sip:40483527@192.168.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.200:5060;rport;branch=z9hG4bKPj935b2ccd-b675-46b6-8b31-763d80f d9574
From:
To:
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
CSeq: 24271 ACK
Max-Forwards: 70
User-Agent: Asterisk GPL PBX
Content-Length: 0
UDPTL (PJSIP/t0gw-0000073b): packet to 192.168.0.10:4617 (seq 0, len 8)
<--- Received SIP request (533 bytes) from UDP:192.168.0.10:5060 --->
OPTIONS sip:192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK72712ff5
Max-Forwards: 70
From: “asterisk”
To:
Contact:
Call-ID: 21f0b6115e7d811a6a399b77424cb2b7@192.168.0.10:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.16.0
Date: Thu, 09 Jul 2015 05:02:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer Content-Length: 0
<--- Transmitting SIP response (829 bytes) to UDP:192.168.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK72712ff
5
Call-ID: 21f0b6115e7d811a6a399b77424cb2b7@192.168.0.10:5060
From: “asterisk”
To:
CSeq: 102 OPTIONS
Accept: application/sdp, application/pidf+xml, application/simple-message-summary, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, message/sipfrag;version=2.0
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
Supported: 100rel, timer, replaces, norefersub Accept-Encoding: text/plain Accept-Language: en Server: Asterisk GPL PBX
Content-Length: 0
<--- Received SIP request (444 bytes) from UDP:192.168.0.10:5060 --->
BYE sip:192.168.0.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK66a545b1;rport Max-Forwards: 70
From:
To:
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 11.16.0
X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16
Content-Length: 0
<--- Transmitting SIP response (353 bytes) to UDP:192.168.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK66a545b
1
Call-ID: 622d9e2260f4cb36405204ea341f9024@192.168.0.10:5060
From:
To:
CSeq: 103 BYE
Server: Asterisk GPL PBX
Content-Length: 0
— Channel PJSIP/t0gw-0000073b left ‘simple_bridge’ basic-bridge
<0328da69-07f5-4270-8fa4-8178649c9906>
— Channel IAX2/iaxmodem0-7773 left ‘simple_bridge’ basic-bridge
<0328da69-07f5-4270-8fa4-8178649c9906>
== Spawn extension (stdexten, fax, 2) exited non-zero on
‘PJSIP/t0gw-0000073b’
— Hungup ‘IAX2/iaxmodem0-7773’
Thanks,
– —
Jean-Denis Girard
SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10.40 / GSM: +689 87.79.75.27
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