[asterisk-bugs] [Asterisk 0012437]: Asterisk negotiates only T.38 when answering even if the other end offers audio
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Mar 4 09:11:47 CST 2009
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=12437
======================================================================
Reported By: marsosa
Assigned To: file
======================================================================
Project: Asterisk
Issue ID: 12437
Category: Channels/chan_sip/T.38
Reproducibility: always
Severity: major
Priority: normal
Status: ready for testing
Asterisk Version: 1.4.18
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2008-04-14 08:31 CDT
Last Modified: 2009-03-04 09:11 CST
======================================================================
Summary: Asterisk negotiates only T.38 when answering even if
the other end offers audio
Description:
One of our gateways (audiocodes mp-118) offers ulaw,g729 and t.38 when an
incoming call is sent to asterisk, and asterisk answer() with t.38 only,
instead of using ulaw. T.38 is enabled on the gateway because this is
needed for reinvites, if i disable it, the call works ok but fails later
when the ata wants to do reinvite for receiving faxes with t.38 '488 not
acceptable'.
The main problem here is that, after answering with t.38, asterisk sends
invites with t.38 only to the ip phones, and they rejected with not
acceptable.
======================================================================
----------------------------------------------------------------------
(0101188) pinga-fogo (reporter) - 2009-03-04 09:11
http://bugs.digium.com/view.php?id=12437#c101188
----------------------------------------------------------------------
<--- SIP read from 200.146.79.165:1571 --->
INVITE sip:FAX_SDI at 189.114.245.124 SIP/2.0
Via: SIP/2.0/UDP 200.146.79.165:1571;branch=z9hG4bK5ibiuu204o301ggfv5k0.1
t: "Camilo MGA t38" <sip:0904430322095 at gvt.com.br:5060;user=phone>
f:
<sip:4433055130 at gvt.com.br:5060;user=phone>;tag=SDeehl201-c0a-13c4-420681-2bb94f8c-420681
i: SDeehl201-dd829eb2bac4b242ca1a53cf2ba12e69-v300g00
CSeq: 1 INVITE
Max-Forwards: 138
x-nt-corr-id: 11fd2095f361670a523c580a9534ffc169a16c032 at 200.175.10.194
X-Nortel-Profile: CTA
x-nt-location: 120224
Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK
m:
<sip:4433055130 at 200.146.79.165:1571;nt_end_pt=YM0+~K12.1zqa131CQit~Kb2P0~Rn2rcevNknice~NC_2SS1y8J+90~EbtYMig~WD9QOkbO.1j83Q8SD1MXU7t.BC1paf6.DbO9o1Tcb1M2tietG~~iE4q44894;nt_server_host=200.175.10.194;transport=udp>
Remote-Party-ID:
<sip:4433055130 at 10.140.131.208;user=phone>;screen=yes;party=calling
k: 100rel,com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
User-Agent: CS2000_NGSS/9.0
l: 420
c: application/sdp
X-Omni-User: sip:ip4440015253f at gvt.com.br
v=0
o=PVG 1236168590700 1236168590700 IN IP4 200.146.79.165
s=-
p=+1 6135555555
c=IN IP4 200.146.79.165
t=0 0
m=audio 10506 RTP/AVP 18 8 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=fmtp:18 annexb=no
m=image 11518 udptl t38
a=T38FaxVersion:0
a=T38FaxMaxBuffer:1100
a=T38FaxMaxDatagram:612
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (18 headers 18 lines) ---
Sending to 200.146.79.165 : 1571 (no NAT)
Using INVITE request as basis request -
SDeehl201-dd829eb2bac4b242ca1a53cf2ba12e69-v300g00
Found peer 'ip4440015253f'
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 101
Got T.38 offer in SDP in dialog
SDeehl201-dd829eb2bac4b242ca1a53cf2ba12e69-v300g00
Peer audio RTP is at port 200.146.79.165:10506
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x108
(alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 200.146.79.165:10506
Looking for FAX_SDI in sip-did (domain 189.114.245.124)
list_route: hop:
<sip:4433055130 at 200.146.79.165:1571;nt_end_pt=YM0+~K12.1zqa131CQit~Kb2P0~Rn2rcevNknice~NC_2SS1y8J+90~EbtYMig~WD9QOkbO.1j83Q8SD1MXU7t.BC1paf6.DbO9o1Tcb1M2tietG~~iE4q44894;nt_server_host=200.175.10.194;transport=udp>
<--- Transmitting (no NAT) to 200.146.79.165:1571 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
200.146.79.165:1571;branch=z9hG4bK5ibiuu204o301ggfv5k0.1;received=200.146.79.165
From:
<sip:4433055130 at gvt.com.br:5060;user=phone>;tag=SDeehl201-c0a-13c4-420681-2bb94f8c-420681
To: "Camilo MGA t38" <sip:0904430322095 at gvt.com.br:5060;user=phone>
Call-ID: SDeehl201-dd829eb2bac4b242ca1a53cf2ba12e69-v300g00
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:FAX_SDI at 189.114.245.124>
Content-Length: 0
<------------>
-- Executing [FAX_SDI at sip-did:1] NoOp("SIP/ip4440015253f-0069bf00",
"") in new stack
-- Executing [FAX_SDI at sip-did:2] Set("SIP/ip4440015253f-0069bf00",
"pseudodid="Camilo MGA t38"
<sip:0904430322095 at gvt.com.br:5060;user=phone>") in new stack
-- Executing [FAX_SDI at sip-did:3] Set("SIP/ip4440015253f-0069bf00",
"pseudodid=Camilo MGA t38 <sip:0904430322095") in new stack
-- Executing [FAX_SDI at sip-did:4] Set("SIP/ip4440015253f-0069bf00",
"pseudodid=0904430322095") in new stack
-- Executing [FAX_SDI at sip-did:5] NoOp("SIP/ip4440015253f-0069bf00",
"2095") in new stack
-- Executing [FAX_SDI at sip-did:6] Goto("SIP/ip4440015253f-0069bf00",
"sip-did|2095|1") in new stack
-- Goto (sip-did,2095,1)
-- Executing [2095 at sip-did:1] Goto("SIP/ip4440015253f-0069bf00",
"entrante-analog|s|1") in new stack
-- Goto (entrante-analog,s,1)
-- Executing [s at entrante-analog:1] Wait("SIP/ip4440015253f-0069bf00",
"2") in new stack
-- Executing [s at entrante-analog:2] Set("SIP/ip4440015253f-0069bf00",
"CIDName="Externa"") in new stack
-- Executing [s at entrante-analog:3] Macro("SIP/ip4440015253f-0069bf00",
"stdexten|s|SIP/5002&SIP/5008|tTj") in new stack
-- Executing [s at macro-stdexten:1]
SIPAddHeader("SIP/ip4440015253f-0069bf00", "Alert-Info: Bellcore-r1") in
new stack
-- Executing [s at macro-stdexten:2] Dial("SIP/ip4440015253f-0069bf00",
"SIP/5002&SIP/5008|30|tTj") in new stack
[Mar 4 11:07:03] WARNING[27302]: chan_sip.c:3081 sip_call: No audio
format found to offer. Cancelling call to 5002
-- Couldn't call 5002
Scheduling destruction of SIP dialog
'08640e3e4d83fac85cfc139c03665669 at 172.16.5.194' in 32000 ms (Method:
INVITE)
[Mar 4 11:07:03] WARNING[27302]: chan_sip.c:3081 sip_call: No audio
format found to offer. Cancelling call to 5008
-- Couldn't call 5008
Scheduling destruction of SIP dialog
'33c51f7b43cd0214355165cc3f41c9cb at 172.16.5.194' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
-- Executing [s at macro-stdexten:3] Goto("SIP/ip4440015253f-0069bf00",
"s-CHANUNAVAIL|1") in new stack
-- Goto (macro-stdexten,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL at macro-stdexten:1]
Goto("SIP/ip4440015253f-0069bf00", "s-NOANSWER|1") in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing [s-NOANSWER at macro-stdexten:1]
Playback("SIP/ip4440015253f-0069bf00",
"/var/lib/asterisk/sounds/camilo/astcc-noanswer") in new stack
[Mar 4 11:07:03] NOTICE[27302]: chan_sip.c:6679 add_sdp: add audio = 1
add t38 = 1
Audio is at 189.114.245.124 port 14892
Adding codec 0x8 (alaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 200.146.79.165:1571 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
200.146.79.165:1571;branch=z9hG4bK5ibiuu204o301ggfv5k0.1;received=200.146.79.165
From:
<sip:4433055130 at gvt.com.br:5060;user=phone>;tag=SDeehl201-c0a-13c4-420681-2bb94f8c-420681
To: "Camilo MGA t38"
<sip:0904430322095 at gvt.com.br:5060;user=phone>;tag=as61d79fef
Call-ID: SDeehl201-dd829eb2bac4b242ca1a53cf2ba12e69-v300g00
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:FAX_SDI at 189.114.245.124>
Content-Type: application/sdp
Content-Length: 478
v=0
o=root 27158 27158 IN IP4 189.114.245.124
s=session
c=IN IP4 189.114.245.124
t=0 0
m=audio 14892 RTP/AVP 8 18 101
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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
m=image 27898 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:612
a=T38FaxMaxDatagram:612
a=T38FaxUdpEC:t38UDPRedundancy
<------------>
-- <SIP/ip4440015253f-0069bf00> Playing
'/var/lib/asterisk/sounds/camilo/astcc-noanswer' (language 'en')
srvsmbig*CLI>
<--- SIP read from 200.146.79.165:1571 --->
ACK sip:FAX_SDI at 189.114.245.124 SIP/2.0
Via: SIP/2.0/UDP 200.146.79.165:1571;branch=z9hG4bKdkrkmn0068i06i4q60s0.1
t: "Camilo MGA t38"
<sip:0904430322095 at gvt.com.br:5060;user=phone>;tag=as61d79fef
f:
<sip:4433055130 at gvt.com.br:5060;user=phone>;tag=SDeehl201-c0a-13c4-420681-2bb94f8c-420681
i: SDeehl201-dd829eb2bac4b242ca1a53cf2ba12e69-v300g00
CSeq: 1 ACK
Max-Forwards: 68
Allow:
ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK,UPDATE
m:
<sip:4433055130 at 200.146.79.165:1571;nt_end_pt=YM0+~K12.1zqa131CQit~Kb2P0~Rn2rcevNknice~NC_2SS1y8J+90~EbtYMig~WD9QOkbO.1j83Q8SD1MXU7t.BC1paf6.DbO9o1Tcb1M2tietG~~iE4q44894;nt_server_host=200.175.10.194;transport=udp>
k: 100rel
User-Agent: CS2000_NGSS/9.0
l: 0
<------------->
--- (12 headers 0 lines) ---
-- Executing [s-NOANSWER at macro-stdexten:2]
Goto("SIP/ip4440015253f-0069bf00", "entrante-analog|s|1") in new stack
-- Goto (entrante-analog,s,1)
== Channel 'SIP/ip4440015253f-0069bf00' jumping out of macro 'stdexten'
-- Executing [s at entrante-analog:1] Wait("SIP/ip4440015253f-0069bf00",
"2") in new stack
-- Executing [s at entrante-analog:2] Set("SIP/ip4440015253f-0069bf00",
"CIDName="Externa"") in new stack
-- Executing [s at entrante-analog:3] Macro("SIP/ip4440015253f-0069bf00",
"stdexten|s|SIP/5002&SIP/5008|tTj") in new stack
-- Executing [s at macro-stdexten:1]
SIPAddHeader("SIP/ip4440015253f-0069bf00", "Alert-Info: Bellcore-r1") in
new stack
-- Executing [s at macro-stdexten:2] Dial("SIP/ip4440015253f-0069bf00",
"SIP/5002&SIP/5008|30|tTj") in new stack
[Mar 4 11:07:07] WARNING[27302]: chan_sip.c:3081 sip_call: No audio
format found to offer. Cancelling call to 5002
-- Couldn't call 5002
Scheduling destruction of SIP dialog
'7b04956b3f054c2052a5dace38404ee9 at 172.16.5.194' in 32000 ms (Method:
INVITE)
[Mar 4 11:07:07] WARNING[27302]: chan_sip.c:3081 sip_call: No audio
format found to offer. Cancelling call to 5008
-- Couldn't call 5008
Scheduling destruction of SIP dialog
'1a0a549270aa43e526be1ad12f20d081 at 172.16.5.194' in 32000 ms (Method:
INVITE)
== Everyone is busy/congested at this time (0:0/0/0)
-- Executing [s at macro-stdexten:3] Goto("SIP/ip4440015253f-0069bf00",
"s-CHANUNAVAIL|1") in new stack
-- Goto (macro-stdexten,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL at macro-stdexten:1]
Goto("SIP/ip4440015253f-0069bf00", "s-NOANSWER|1") in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing [s-NOANSWER at macro-stdexten:1]
Playback("SIP/ip4440015253f-0069bf00",
"/var/lib/asterisk/sounds/camilo/astcc-noanswer") in new stack
-- <SIP/ip4440015253f-0069bf00> Playing
'/var/lib/asterisk/sounds/camilo/astcc-noanswer' (language 'en')
Really destroying SIP dialog 'BD21-16CF-46684823DBE7893D776A-005 at SipHost'
Method: REGISTER
Really destroying SIP dialog 'BD21-16CF-4668482379FABBEFDA77-007 at SipHost'
Method: REGISTER
Really destroying SIP dialog 'BD21-16CF-46684823EF67CE8DDEF0-008 at SipHost'
Method: REGISTER
srvsmbig*CLI>
<--- SIP read from 200.146.79.165:1571 --->
BYE sip:FAX_SDI at 189.114.245.124 SIP/2.0
Via: SIP/2.0/UDP
200.146.79.165:1571;branch=z9hG4bKdkrkmn0068i06i4q60s0cd0000010.1
t: "Camilo MGA t38"
<sip:0904430322095 at gvt.com.br:5060;user=phone>;tag=as61d79fef
f:
<sip:4433055130 at gvt.com.br:5060;user=phone>;tag=SDeehl201-c0a-13c4-420681-2bb94f8c-420681
i: SDeehl201-dd829eb2bac4b242ca1a53cf2ba12e69-v300g00
CSeq: 2 BYE
Max-Forwards: 68
Allow:
ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK,UPDATE
Reason: Q.850;cause=16;text="Normal call clearing"
k: 100rel,com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
User-Agent: CS2000_NGSS/9.0
l: 0
<------------->
--- (12 headers 0 lines) ---
Sending to 200.146.79.165 : 1571 (no NAT)
<--- Transmitting (no NAT) to 200.146.79.165:1571 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
200.146.79.165:1571;branch=z9hG4bKdkrkmn0068i06i4q60s0cd0000010.1;received=200.146.79.165
From:
<sip:4433055130 at gvt.com.br:5060;user=phone>;tag=SDeehl201-c0a-13c4-420681-2bb94f8c-420681
To: "Camilo MGA t38"
<sip:0904430322095 at gvt.com.br:5060;user=phone>;tag=as61d79fef
Call-ID: SDeehl201-dd829eb2bac4b242ca1a53cf2ba12e69-v300g00
CSeq: 2 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
'SIP/ip4440015253f-0069bf00' in macro 'stdexten'
== Spawn extension (entrante-analog, s, 3) exited non-zero on
'SIP/ip4440015253f-0069bf00'
Really destroying SIP dialog
'SDeehl201-dd829eb2bac4b242ca1a53cf2ba12e69-v300g00' Method: BYE
Really destroying SIP dialog
'287f4e10044a36207914f5a75f73e799 at 172.16.5.194' Method: INVITE
Really destroying SIP dialog
'05c12fa00842f38f2deb608a4d8a846b at 172.16.5.194' Method: INVITE
Issue History
Date Modified Username Field Change
======================================================================
2009-03-04 09:11 pinga-fogo Note Added: 0101188
======================================================================
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