[asterisk-users] PJSIP T.38 issues
Jean-Denis Girard
jd.girard at sysnux.pf
Sun Jul 26 22:15:07 CDT 2015
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Hi list,
2 weks ago I asked questions about PJSIP and T.38 but got no replies. I
upgraded Asterisk to git as of yesterday (309dd2a), and I'm still having
the same issues.
In the trace below, I'm sending a fax from Hylafax server through
iaxmodem on Asterisk-13 (tiare) to a second Asterisk (11.18.0 t0gw)
connected to the PSTN via ISDN; the call is to my test fax machine,
connected to the PSTN. chan_pjsip is used on Asterisk-13, and chan_sip
is used on Asterisk-11.
This how endpoint t0gw (Asterisk-11) is configured on tiare (Asterisk-13
):
tiare*CLI> pjsip show endpoint t0gw
...
t38_udptl : true
t38_udptl_ec : fec
t38_udptl_ipv6 : false
t38_udptl_maxdatagram : 400
t38_udptl_nat : false
...
Could someone explain why I'm getting "Not acceptable" below?
-- Accepting AUTHENTICATED call from 127.0.0.1:4570:
-- > requested format = slin,
-- > requested prefs = (),
-- > actual format = slin,
-- > host prefs = (slin),
-- > priority = mine
-- Executing [40ZZZZZZ at fax-sortant:1] NoOp("IAX2/iaxmodem0-7838", "
calls 40ZZZZZZ (local)") in new stack
-- Executing [40ZZZZZZ at fax-sortant:2] Set("IAX2/iaxmodem0-7838",
"FAXOPT(gateway)=yes") in new stack
-- Executing [40ZZZZZZ at fax-sortant:3] Dial("IAX2/iaxmodem0-7838",
"PJSIP/40ZZZZZZ at t0gw") in new stack
-- Called PJSIP/40ZZZZZZ at t0gw
<--- Transmitting SIP request (936 bytes) to UDP:192.168.0.10:5060 --->
INVITE sip:40ZZZZZZ at gw.sysnux.pf SIP/2.0
Via: SIP/2.0/UDP
192.168.0.200:5060;rport;branch=z9hG4bKPjba38816d-b5fe-4d5f-8bda-d0968e3
8e5f1
From: "SysNux"
<sip:+68940XXXXXX at 192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
5
To: <sip:40ZZZZZZ at gw.sysnux.pf>
Contact: <sip:63035284-ad7d-484f-8e54-f5ea54f39104 at 192.168.0.200:5060>
Call-ID: 57283616-94e0-4052-bdff-491b31fdd229
CSeq: 31693 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: 238
v=0
o=- 1710591484 1710591484 IN IP4 192.168.0.200
s=Asterisk
c=IN IP4 192.168.0.200
t=0 0
m=audio 8834 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
<--- Received SIP response (585 bytes) from UDP:192.168.0.10:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.0.200:5060;branch=z9hG4bKPjba38816d-b5fe-4d5f-8bda-d0968e38e5f1;
received=192.168.0.200;rport=5060
From: "SysNux"
<sip:+68940XXXXXX at 192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
5
To: <sip:40ZZZZZZ at gw.sysnux.pf>
Call-ID: 57283616-94e0-4052-bdff-491b31fdd229
CSeq: 31693 INVITE
Server: Asterisk PBX 11.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:40ZZZZZZ at 192.168.0.10:5060>
Content-Length: 0
<--- Received SIP response (895 bytes) from UDP:192.168.0.10:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP
192.168.0.200:5060;branch=z9hG4bKPjba38816d-b5fe-4d5f-8bda-d0968e38e5f1;
received=192.168.0.200;rport=5060
From: "SysNux"
<sip:+68940XXXXXX at 192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
5
To: <sip:40ZZZZZZ at gw.sysnux.pf>;tag=as7bba6b0d
Call-ID: 57283616-94e0-4052-bdff-491b31fdd229
CSeq: 31693 INVITE
Server: Asterisk PBX 11.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:40ZZZZZZ at 192.168.0.10:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 236
v=0
o=root 2087714374 2087714374 IN IP4 192.168.0.10
s=Asterisk PBX 11.18.0
c=IN IP4 192.168.0.10
t=0 0
m=audio 16834 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
-- PJSIP/t0gw-0000001a is making progress passing it to
IAX2/iaxmodem0-7838
<--- Received SIP response (601 bytes) from UDP:192.168.0.10:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
192.168.0.200:5060;branch=z9hG4bKPjba38816d-b5fe-4d5f-8bda-d0968e38e5f1;
received=192.168.0.200;rport=5060
From: "SysNux"
<sip:+68940XXXXXX at 192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
5
To: <sip:40ZZZZZZ at gw.sysnux.pf>;tag=as7bba6b0d
Call-ID: 57283616-94e0-4052-bdff-491b31fdd229
CSeq: 31693 INVITE
Server: Asterisk PBX 11.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:40ZZZZZZ at 192.168.0.10:5060>
Content-Length: 0
-- PJSIP/t0gw-0000001a is ringing
<--- Received SIP response (881 bytes) from UDP:192.168.0.10:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.200:5060;branch=z9hG4bKPjba38816d-b5fe-4d5f-8bda-d0968e38e5f1;
received=192.168.0.200;rport=5060
From: "SysNux"
<sip:+68940XXXXXX at 192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
5
To: <sip:40ZZZZZZ at gw.sysnux.pf>;tag=as7bba6b0d
Call-ID: 57283616-94e0-4052-bdff-491b31fdd229
CSeq: 31693 INVITE
Server: Asterisk PBX 11.18.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:40ZZZZZZ at 192.168.0.10:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 236
v=0
o=root 2087714374 2087714374 IN IP4 192.168.0.10
s=Asterisk PBX 11.18.0
c=IN IP4 192.168.0.10
t=0 0
m=audio 16834 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 request (412 bytes) to UDP:192.168.0.10:5060 --->
ACK sip:40ZZZZZZ at 192.168.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.0.200:5060;rport;branch=z9hG4bKPj8504e505-1222-4747-955f-4788fef
f58d1
From: "SysNux"
<sip:+68940XXXXXX at 192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
5
To: <sip:40ZZZZZZ at gw.sysnux.pf>;tag=as7bba6b0d
Call-ID: 57283616-94e0-4052-bdff-491b31fdd229
CSeq: 31693 ACK
Max-Forwards: 70
User-Agent: Asterisk GPL PBX
Content-Length: 0
-- PJSIP/t0gw-0000001a answered IAX2/iaxmodem0-7838
-- Channel PJSIP/t0gw-0000001a joined 'simple_bridge' basic-bridge
<56a7726f-44a3-4df3-aee0-d21020aa5be1>
-- Channel IAX2/iaxmodem0-7838 joined 'simple_bridge' basic-bridge
<56a7726f-44a3-4df3-aee0-d21020aa5be1>
<--- Received SIP request (954 bytes) from UDP:192.168.0.10:5060 --->
UPDATE sip:63035284-ad7d-484f-8e54-f5ea54f39104 at 192.168.0.200:5060 SIP/2
.0
Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK4fd84f17;rport
Max-Forwards: 70
From: <sip:40ZZZZZZ at gw.sysnux.pf>;tag=as7bba6b0d
To: "SysNux"
<sip:+68940XXXXXX at 192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
5
Contact: <sip:40ZZZZZZ at 192.168.0.10:5060>
Call-ID: 57283616-94e0-4052-bdff-491b31fdd229
CSeq: 102 UPDATE
User-Agent: Asterisk PBX 11.18.0
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: 287
v=0
o=root 2087714374 2087714375 IN IP4 192.168.0.10
s=Asterisk PBX 11.18.0
c=IN IP4 192.168.0.10
t=0 0
m=image 5720 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 response (376 bytes) to UDP:192.168.0.10:5060 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/UDP
192.168.0.10:5060;rport=5060;received=192.168.0.10;branch=z9hG4bK4fd84f1
7
Call-ID: 57283616-94e0-4052-bdff-491b31fdd229
From: <sip:40ZZZZZZ at gw.sysnux.pf>;tag=as7bba6b0d
To: "SysNux"
<sip:+68940XXXXXX at 192.168.0.200>;tag=22d8369f-061d-4232-9c4d-5068e81bc5c
5
CSeq: 102 UPDATE
Server: Asterisk GPL PBX
Content-Length: 0
Is anyone successfully using chan_pjsip and iaxmodem?
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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