[asterisk-bugs] [Asterisk 0008078]: T38 relay doesn't work between Audiocodes Tulip AC494 ATAs
noreply at bugs.digium.com
noreply at bugs.digium.com
Wed Sep 26 16:28:45 CDT 2007
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=8078
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Reported By: candlerb
Assigned To: oej
======================================================================
Project: Asterisk
Issue ID: 8078
Category: Channels/chan_sip/T.38
Reproducibility: always
Severity: major
Priority: normal
Status: feedback
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): trunk
SVN Revision (number only!): 44249
Disclaimer on File?: No
Request Review:
======================================================================
Date Submitted: 10-03-2006 07:32 CDT
Last Modified: 09-26-2007 16:28 CDT
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Summary: T38 relay doesn't work between Audiocodes Tulip
AC494 ATAs
Description:
When trying to use T38 fax relay between a pair of Audiocodes AC494's, when
fax is detected and one ATA sends a reinvite with
m=image 5004 UDPTL t38
a=T38FaxVersion:0
...
Asterisk rejects it with "488 Not acceptable here" and logs "Unsupported
SDP media type in offer: image 5004 UDPTL t38". As a result, fax relay
fails completely.
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Relationships ID Summary
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related to 0007679 T.38 passthrough is not working between...
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jon - 09-26-07 16:28
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Ahh, apparently the packet I really wanted wasn't within the scope of the
command: sip debug peer testphone
You can delete 20070926_debug.txt, look at this instead
pbx*CLI> sip debug ip 216.82.224.202
SIP Debugging Enabled for IP: 216.82.224.202
-- Executing [dial-15082937963 at local:5] Set("SIP/testphone-0844ae70",
"CALLERID(all)="Elephant Outlook" <+18638774177>") in new stack
-- Executing [dial-15082937963 at local:6]
Dial("SIP/testphone-0844ae70",
"SIP/bandwidth/+15082937963") in new stack Video is at 192.168.10.225 port
18900 Audio is at 192.168.10.225 port 12894 Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP Adding codec 0x80000 (h263) to SDP Adding
non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to
216.82.224.202:5060:
INVITE sip:+15082937963 at 216.82.224.202 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK7e490e03;rport
From: "Elephant Outlook" <sip:+18638774177 at 4.68.250.148>;tag=as0bcd3341
To: <sip:+15082937963 at 216.82.224.202>
Contact: <sip:+18638774177 at 192.168.10.225>
Call-ID: 047caabf60378e05225c67c25441df80 at 4.68.250.148
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 26 Sep 2007 21:43:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 339
v=0
o=root 16400 16400 IN IP4 192.168.10.225 s=session c=IN IP4
192.168.10.225
b=CT:384
t=0 0
m=audio 12894 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=video 18900 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv
---
-- Called bandwidth/+15552937963
<--- SIP read from 216.82.224.202:5060 ---> SIP/2.0 100 bandwidth.com has
received your request
Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK7e490e03;rport=5060
From: "Elephant Outlook" <sip:+18638774177 at 4.68.250.148>;tag=as0bcd3341
To: <sip:+15552937963 at 216.82.224.202>
Call-ID: 047caabf60378e05225c67c25441df80 at 4.68.250.148
CSeq: 102 INVITE
Server: Bandwidth.com TRM (bw7.gold.13)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
pbx*CLI>
<--- SIP read from 216.82.224.202:5060 ---> SIP/2.0 183 Session Progress
v: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK7e490e03;rport=5060
Record-Route:
<sip:216.82.224.202;lr;ftag=as0bcd3341;vsf=AAAAABMHCw4EDwMGAwZ3A24CFhgKGwIbARoJNDg->
P-Charging-Vector:icid-value=default~a187189b7133a7da19724dd436a55426;icid-generated-at=204.13.236.129;orig-ioi=default~;term-ioi=default~;Charge=sip%3A8638774177%404.68.250.148
From: "Elephant Outlook" <sip:+18638774177 at 4.68.250.148>;tag=as0bcd3341
t:
<sip:+15552937963 at 216.82.224.202>;tag=61bd0a692dcabf2a933b95ffd8f47375
i: 047caabf60378e05225c67c25441df80 at 4.68.250.148
CSeq: 102 INVITE
Server: DC-SIP/1.2
k: timer
m: <sip:5552937963 at 204.13.236.129:5060;transport=udp>
c: application/sdp
l: 290
v=0
o=- 194901 19490100 IN IP4 204.13.236.130
s=-
t=0 0
m=image 44742 udptl t38
c=IN IP4 204.13.236.130
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
=T38FaxMaxDatagram:72
<------------->
--- (13 headers 13 lines) ---
[Sep 26 17:43:04] WARNING[25857]: chan_sip.c:5017 process_sdp:
Unsupported SDP media type in offer: image 44742 udptl t38
-- SIP/bandwidth-084fecf8 is making progress passing it to
SIP/testphone-0844ae70 Scheduling destruction of SIP dialog
'047caabf60378e05225c67c25441df80 at 4.68.250.148' in 32000 ms (Method:
INVITE)
Reliably Transmitting (no NAT) to 216.82.224.202:5060:
CANCEL sip:+15552937963 at 216.82.224.202 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK7e490e03;rport
From: "Elephant Outlook" <sip:+18638774177 at 4.68.250.148>;tag=as0bcd3341
To: <sip:+15552937963 at 216.82.224.202>
Call-ID: 047caabf60378e05225c67c25441df80 at 4.68.250.148
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Scheduling destruction of SIP dialog
'047caabf60378e05225c67c25441df80 at 4.68.250.148' in 32000 ms (Method:
INVITE)
== Spawn extension (local, dial-15552937963, 6) exited non-zero on
'SIP/testphone-0844ae70'
Really destroying SIP dialog 'b01d298b-7082aaba-dd93fa81 at 192.168.2.21'
Method: ACK
pbx*CLI>
<--- SIP read from 216.82.224.202:5060 ---> SIP/2.0 200 canceling
Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK7e490e03;rport=5060
From: "Elephant Outlook" <sip:+18638774177 at 4.68.250.148>;tag=as0bcd3341
To:
<sip:+15552937963 at 216.82.224.202>;tag=6c8d6b8eaf3981f68fb5449c71ab6dfd-392d
Call-ID: 047caabf60378e05225c67c25441df80 at 4.68.250.148
CSeq: 102 CANCEL
Server: Bandwidth.com TRM (bw7.gold.13)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
pbx*CLI>
<--- SIP read from 216.82.224.202:5060 ---> SIP/2.0 487 Request
Terminated
v: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK7e490e03;rport=5060
Record-Route:
<sip:216.82.224.202;lr;ftag=as0bcd3341;vsf=AAAAABMHCw4EDwMGAwZ3A24CFhgKGwIbARoJNDg->
P-Charging-Vector:icid-value=default~a187189b7133a7da19724dd436a55426;icid-generated-at=204.13.236.129;orig-ioi=default~;term-ioi=default~;Charge=sip%3A8638774177%404.68.250.148
From: "Elephant Outlook" <sip:+18638774177 at 4.68.250.148>;tag=as0bcd3341
t:
<sip:+15552937963 at 216.82.224.202>;tag=61bd0a692dcabf2a933b95ffd8f47375
i: 047caabf60378e05225c67c25441df80 at 4.68.250.148
CSeq: 102 INVITE
Server: DC-SIP/1.2
k: timer
m: <sip:5552937963 at 204.13.236.129:5060;transport=udp>
l: 0
<------------->
--- (12 headers 0 lines) ---
Transmitting (no NAT) to 216.82.224.202:5060:
ACK sip:+15552937963 at 216.82.224.202 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.225:5060;branch=z9hG4bK7e490e03;rport
From: "Elephant Outlook" <sip:+18638774177 at 4.68.250.148>;tag=as0bcd3341
To:
<sip:+15552937963 at 216.82.224.202>;tag=61bd0a692dcabf2a933b95ffd8f47375
Contact: <sip:+18638774177 at 192.168.10.225>
Call-ID: 047caabf60378e05225c67c25441df80 at 4.68.250.148
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Really destroying SIP dialog
'047caabf60378e05225c67c25441df80 at 4.68.250.148' Method: INVITE
pbx*CLI>
Issue History
Date Modified Username Field Change
======================================================================
09-26-07 16:28 jon Note Added: 0071133
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