[asterisk-bugs] [Asterisk 0016327]: Asterisk responds 488 - Not acceptable here on T38 reinvite
Asterisk Bug Tracker
noreply at bugs.digium.com
Thu Dec 3 03:59:01 CST 2009
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=16327
======================================================================
Reported By: serje
Assigned To: kpfleming
======================================================================
Project: Asterisk
Issue ID: 16327
Category: Channels/chan_sip/T.38
Reproducibility: always
Severity: major
Priority: normal
Status: assigned
Asterisk Version: Older 1.6.1 - please test a newer version
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2009-11-26 03:48 CST
Last Modified: 2009-12-03 03:59 CST
======================================================================
Summary: Asterisk responds 488 - Not acceptable here on T38
reinvite
Description:
I'm trying to send a fax from asterisk through Audiocodes Mediant1000
gateway to PSTN using T38.
The call is being set up in audio, then Mediant sends reinvite with T38 -
and asterisk responds "488 Not acceptable here".
This can be observed in 1.6.0.15, 1.6.1.5, 1.6.1.9 (current 1.6 trunk and
latest releases nave another bug (is is on bug tracker) - they can't
establish connection on sip trunk, so I couldn't verify, whether T38 issue
present in them.
======================================================================
----------------------------------------------------------------------
(0114644) linuxrulez (reporter) - 2009-12-03 03:59
https://issues.asterisk.org/view.php?id=16327#c114644
----------------------------------------------------------------------
Confirming for 1.6.2r231696:
[Dec 3 11:39:52] -- Accepting AUTHENTICATED call from XXXXX:
> requested format = alaw,
> requested prefs = (alaw|ulaw|slin|gsm|g729),
> actual format = alaw,
> host prefs = (alaw|slin|gsm|g729),
> priority = mine
[Dec 3 11:39:52] -- Executing [test at from_voip:1]
FaxGateway("IAX2/test-5099", "SIP/faxer/703,10,R") in new stack
[Dec 3 11:39:52] == Using SIP RTP CoS mark 5
[Dec 3 11:39:52] == Using UDPTL CoS mark 5
[Dec 3 11:39:52] Audio is at XXXX port 15636
[Dec 3 11:39:52] Adding codec 0x2 (gsm) to SDP
[Dec 3 11:39:52] Adding codec 0x4 (ulaw) to SDP
[Dec 3 11:39:52] Adding codec 0x8 (alaw) to SDP
[Dec 3 11:39:52] Adding non-codec 0x1 (telephone-event) to SDP
[Dec 3 11:39:52] Reliably Transmitting (no NAT) to XXXXXX:5061:
INVITE sip:703 at XXX:5061 SIP/2.0
Via: SIP/2.0/UDP XXXXX:5060;branch=z9hG4bK588ad1dd;rport
Max-Forwards: 70
From: "703" <sip:703 at XXXX>;tag=as31a0cb3e
To: <sip:703 at XXXXX:5061>
Contact: <sip:703 at XXXXX>
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321 at XXXX
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.6.2-r231696M
Date: Thu, 03 Dec 2009 09:39:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 330
v=0
o=root 1106053413 1106053413 IN IP4 XXXXX
s=Asterisk PBX SVN-branch-1.6.2-r231696M
c=IN IP4 XXXXXX
t=0 0
m=audio 15636 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
[Dec 3 11:39:52]
<--- SIP read from UDP:XXXXXX:5061 --->
SIP/2.0 100 Trying
CSeq: 102 INVITE
Via: SIP/2.0/UDP
XXXX:5060;branch=z9hG4bK588ad1dd;rport=5060;received=XXXXXXX
From: "703" <sip:703 at XXXXX>;tag=as31a0cb3e
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321 at XXXXX
To: <sip:703 at XXXXX:5061>
Contact: <sip:703 at XXXXX:5061>
Content-Length: 0
<------------->
[Dec 3 11:39:52] --- (8 headers 0 lines) ---
[Dec 3 11:39:52]
<--- SIP read from UDP:XXXXX:5061 --->
SIP/2.0 180 Ringing
CSeq: 102 INVITE
Via: SIP/2.0/UDP
XXXX:5060;branch=z9hG4bK588ad1dd;rport=5060;received=XXXX
User-Agent: T38Modem/1.2.1
From: "703" <sip:703 at XXXX>;tag=as31a0cb3e
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321 at XXXXX
Organization: Vyacheslav Frolov
To: <sip:703 at XXXX:5060>;tag=2070e879-5dde-de11-8f00-001111ef9b5e
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Length: 0
<------------->
[Dec 3 11:39:52] --- (10 headers 0 lines) ---
[Dec 3 11:39:52]
<--- SIP read from UDP:XXXXXX:5061 --->
SIP/2.0 200 OK
CSeq: 102 INVITE
Via: SIP/2.0/UDP
XXXX:5060;branch=z9hG4bK588ad1dd;rport=5060;received=XXXXXXXX
User-Agent: T38Modem/1.2.1
From: "703" <sip:703 at XXXXXX>;tag=as31a0cb3e
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321 at XXXXXXXXX
Organization: Vyacheslav Frolov
To: <sip:703 at XXXXXx:5060>;tag=2070e879-5dde-de11-8f00-001111ef9b5e
Contact: <sip:703 at XXXXXXx:5061>
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Type: application/sdp
Content-Length: 231
v=0
o=- 1259833192 1 IN IP4 XXXXXX
s=T38Modem/1.2.1
c=IN IP4 XXXXXX
t=0 0
m=audio 5002 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=maxptime:240
<------------->
[Dec 3 11:39:52] --- (12 headers 11 lines) ---
[Dec 3 11:39:52] Found RTP audio format 0
[Dec 3 11:39:52] Found RTP audio format 101
[Dec 3 11:39:52] Found audio description format PCMU for ID 0
[Dec 3 11:39:52] Found audio description format telephone-event for ID
101
[Dec 3 11:39:52] Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer -
audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4
(ulaw)
[Dec 3 11:39:52] Non-codec capabilities (dtmf): us - 0x1
(telephone-event), peer - 0x1 (telephone-event), combined - 0x1
(telephone-event)
[Dec 3 11:39:52] Peer audio RTP is at port XXXX:5002
[Dec 3 11:39:52] list_route: hop: <sip:703 at XXXX:5061>
[Dec 3 11:39:52] set_destination: Parsing <sip:703 at XXX:5061> for
address/port to send to
[Dec 3 11:39:52] set_destination: set destination to XXXX, port 5061
[Dec 3 11:39:52] Transmitting (no NAT) to XXXX:5061:
ACK sip:703 at XXXXx:5061 SIP/2.0
Via: SIP/2.0/UDP XXXXX:5060;branch=z9hG4bK3fa801fb;rport
Max-Forwards: 70
From: "703" <sip:703 at XXXX>;tag=as31a0cb3e
To: <sip:703 at XXXX:5061>;tag=2070e879-5dde-de11-8f00-001111ef9b5e
Contact: <sip:703 at XXXXX>
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321 at XXXXX
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-branch-1.6.2-r231696M
Content-Length: 0
---
[Dec 3 11:39:53]
<--- SIP read from UDP:XXXX:5061 --->
INVITE sip:703 at XXXXXXXX SIP/2.0
CSeq: 2 INVITE
Via: SIP/2.0/UDP
XXXXXXX:5061;branch=z9hG4bK0a75947a-5dde-de11-8f00-001111ef9b5e;rport
User-Agent: T38Modem/1.2.1
From: <sip:703 at XXXX:5060>;tag=2070e879-5dde-de11-8f00-001111ef9b5e
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321 at XXXXX
Organization: Vyacheslav Frolov
To: "703" <sip:703 at XXXXXX>;tag=as31a0cb3e
Contact: <sip:703 at XXXXX:5061>
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Type: application/sdp
Content-Length: 187
Max-Forwards: 70
v=0
o=- 1259833192 2 IN IP4 XXXXX
s=T38Modem/1.2.1
c=IN IP4 XXXXXXX
t=0 0
m=image 5002 udptl t38
a=sendrecv
a=T38FaxVersion:0
a=T38FaxRateManagement:transferredTCF
<------------->
[Dec 3 11:39:53] --- (13 headers 9 lines) ---
[Dec 3 11:39:53] Sending to XXXXXX : 5061 (no NAT)
[Dec 3 11:39:53] Got T.38 offer in SDP in dialog
4ddcdbca767a3f4842ad448c5a7c3321 at XXXXXX
[Dec 3 11:39:53] Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer -
audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0
(nothing)
[Dec 3 11:39:53] Non-codec capabilities (dtmf): us - 0x1
(telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Dec 3 11:39:53] Got T.38 Re-invite without audio. Keeping RTP active
during T.38 session.
[Dec 3 11:39:53] WARNING[22525]: udptl.c:766 calculate_far_max_ifp: (no
tag): Cannot calculate far_max_ifp before far_max_datagram has been set.
[Dec 3 11:39:53]
<--- Transmitting (no NAT) to XXXXX:5061 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
XXXXx:5061;branch=z9hG4bK0a75947a-5dde-de11-8f00-001111ef9b5e;received=XXXXXx;rport=5061
From: <sip:703 at XXXXX:5060>;tag=2070e879-5dde-de11-8f00-001111ef9b5e
To: "703" <sip:703 at XXXXX>;tag=as31a0cb3e
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321 at XXXXXX
CSeq: 2 INVITE
Server: Asterisk PBX SVN-branch-1.6.2-r231696M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:703 at XXXXX>
Content-Length: 0
<--- Reliably Transmitting (no NAT) to XXXXX:5061 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP
XXXXXX:5061;branch=z9hG4bK0a75947a-5dde-de11-8f00-001111ef9b5e;received=XXXXX;rport=5061
From: <sip:703 at XXXXXX:5060>;tag=2070e879-5dde-de11-8f00-001111ef9b5e
To: "703" <sip:703 at XXXXXXX>;tag=as31a0cb3e
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321 at XXXXx
CSeq: 2 INVITE
Server: Asterisk PBX SVN-branch-1.6.2-r231696M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
<------------>
[Dec 3 11:39:58]
<--- SIP read from UDP:XXXXX:5061 --->
ACK sip:703 at XXXXx SIP/2.0
CSeq: 2 ACK
Via: SIP/2.0/UDP
XXXXXx:5061;branch=z9hG4bK0a75947a-5dde-de11-8f00-001111ef9b5e;rport
From: <sip:703 at XXXXXXx:5060>;tag=2070e879-5dde-de11-8f00-001111ef9b5e
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321 at XXXXXXX
To: "703" <sip:703 at XXXXXx>;tag=as31a0cb3e
Content-Length: 0
Max-Forwards: 70
<------------->
[Dec 3 11:39:58] --- (8 headers 0 lines) ---
< ONLY SILENCE HERE, no FAX tones (SIP connected to T38modem, connected to
FaxGetty from HylaFAX+>
[Dec 3 11:40:07] Scheduling destruction of SIP dialog
'4ddcdbca767a3f4842ad448c5a7c3321 at XXXXX' in 32000 ms (Method: ACK)
[Dec 3 11:40:07] set_destination: Parsing <sip:703 at XXXX:5061> for
address/port to send to
[Dec 3 11:40:07] set_destination: set destination to XXXXx, port 5061
[Dec 3 11:40:07] Reliably Transmitting (no NAT) to XXXX:5061:
BYE sip:703 at XXXXXxx:5061 SIP/2.0
Via: SIP/2.0/UDP XXXx:5060;branch=z9hG4bK01f98963;rport
Max-Forwards: 70
From: "703" <sip:703 at XXXXXX>;tag=as31a0cb3e
To: <sip:703 at XXXXX:5060>;tag=2070e879-5dde-de11-8f00-001111ef9b5e
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321 at XXXXX
CSeq: 103 BYE
User-Agent: Asterisk PBX SVN-branch-1.6.2-r231696M
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
[Dec 3 11:40:07] -- Hungup 'IAX2/test-5099'
[Dec 3 11:40:07]
<--- SIP read from UDP:XXXXx:5061 --->
SIP/2.0 200 OK
CSeq: 103 BYE
Via: SIP/2.0/UDP XXXXXx:5060;branch=z9hG4bK01f98963;rport
From: "703" <sip:703 at XXXXXX>;tag=as31a0cb3e
Call-ID: 4ddcdbca767a3f4842ad448c5a7c3321 at XXXXXx
To: <sip:703 at XXXXXx:5060>;tag=2070e879-5dde-de11-8f00-001111ef9b5e
Content-Length: 0
<------------->
[Dec 3 11:40:07] --- (7 headers 0 lines) ---
[Dec 3 11:40:07] SIP Response message for INCOMING dialog BYE arrived
[Dec 3 11:40:07] Really destroying SIP dialog
'4ddcdbca767a3f4842ad448c5a7c3321 at XXXXX' Method: ACK
Issue History
Date Modified Username Field Change
======================================================================
2009-12-03 03:59 linuxrulez Note Added: 0114644
======================================================================
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