Hi,
I have upgraded the machine from kernel-2.4.x to kernel-2.6.x . Now I am
facing different problem "Got SIP response 488 "Not acceptable here" back
from xx.xx.xx.14" .
OS : CentOS-4.3
Kernel : 2.6.9-42.0.3.ELsmp
GCC : gcc-3.4.6-3
SpanDSP : spandsp-0.0.3pre27/spandsp-20070123
OpenPBX-1.2-RC3
sip.conf
======
t38udptlsupport=yes
t38rtpsupport=yes
t38tcpsupport=yes
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=alaw ;
dtmfmode = inband
Note : I am attaching debug log .
Thanks once again .
Regards,
..Tusar..
Am I missing something ? Pls light on this topics .
i bet you need kernel 2.6 and probably it's time to upgrade that machine.
Max
pbx*CLI> show version
OpenPBX.org 1.2-RC3 built on pbx, a i686 running Linux on 2007-01-28 06:59:30
UTC
<-- SIP read from xx.xx.xx.14:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.14:5060;branch=z9hG4bK71f62fc7;rport
From: "xxxxx82766" <sip:[EMAIL PROTECTED]>;tag=as7acecfa4
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 28 Jan 2007 07:14:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 311
v=0
o=root 11023 11023 IN IP4 xx.xx.xx.14
s=session
c=IN IP4 xx.xx.xx.14
t=0 0
m=audio 25026 RTP/AVP 4 18 8 0 101
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (13 headers 14 lines) ---
Using INVITE request as basis request - [EMAIL PROTECTED]
Sending to xx.xx.xx.14 : 5060 (NAT)
Found peer 'Mercury'
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port xx.xx.xx.14:25026
Found description format G723
Found description format G729
Found description format PCMA
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x10d
(g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Looking for xxxxx30995 in fax2email (domain xx.xx.231.18)
list_route: hop: <sip:[EMAIL PROTECTED]>
Transmitting (NAT) to xx.xx.xx.14:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
xx.xx.xx.14:5060;branch=z9hG4bK71f62fc7;rport;received=xx.xx.xx.14
From: "xxxxx82766" <sip:[EMAIL PROTECTED]>;tag=as7acecfa4
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: OpenPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
---
-- Executing Wait("SIP/xx.xx.xx.14-08231c80", "1") in new stack
-- Executing Set("SIP/xx.xx.xx.14-08231c80", "CALLEDFAX=xxxxx30995") in new
stack
-- Executing FaxDetect("SIP/xx.xx.xx.14-08231c80", "4000") in new stack
We're at xx.xx.231.18 port 16660
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to xx.xx.xx.14:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
xx.xx.xx.14:5060;branch=z9hG4bK71f62fc7;rport;received=xx.xx.xx.14
From: "xxxxx82766" <sip:[EMAIL PROTECTED]>;tag=as7acecfa4
To: <sip:[EMAIL PROTECTED]>;tag=as06f9c56a
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: OpenPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:[EMAIL PROTECTED]>
Content-Type: application/sdp
Content-Length: 243
v=0
o=root 14512 14512 IN IP4 xx.xx.231.18
s=session
c=IN IP4 xx.xx.231.18
=0 0flower*CLI>
m=audio 16660 RTP/AVP 0 122 101
a=rtpmap:0 PCMU/8000
a=rtpmap:122 t38/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
pbx*CLI>
<-- SIP read from xx.xx.xx.14:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.14:5060;branch=z9hG4bK61718ab9;rport
From: "xxxxx82766" <sip:[EMAIL PROTECTED]>;tag=as7acecfa4
To: <sip:[EMAIL PROTECTED]>;tag=as06f9c56a
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
--- (10 headers 0 lines) ---
pbx*CLI>
<-- SIP read from xx.xx.xx.14:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.14:5060;branch=z9hG4bK5bbc4f6b;rport
From: "xxxxx82766" <sip:[EMAIL PROTECTED]>;tag=as7acecfa4
To: <sip:[EMAIL PROTECTED]>;tag=as06f9c56a
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 293
v=0
o=root 11023 11024 IN IP4 xx.xx.167.212
s=session
c=IN IP4 xx.xx.167.212
t=0 0
m=audio 10288 RTP/AVP 4 18 0 101
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (12 headers 13 lines) ---
Using INVITE request as basis request - [EMAIL PROTECTED]
Sending to xx.xx.xx.14 : 5060 (NAT)
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port xx.xx.167.212:10288
Found description format G723
Found description format G729
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x105 (g723|ulaw|g729)/video=0x0
(nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
We're at xx.xx.231.18 port 16660
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to xx.xx.xx.14:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
xx.xx.xx.14:5060;branch=z9hG4bK5bbc4f6b;rport;received=xx.xx.xx.14
From: "xxxxx82766" <sip:[EMAIL PROTECTED]>;tag=as7acecfa4
To: <sip:[EMAIL PROTECTED]>;tag=as06f9c56a
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: OpenPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:[EMAIL PROTECTED]>
Content-Type: application/sdp
Content-Length: 243
v=0
o=root 14512 14513 IN IP4 xx.xx.231.18
s=session
c=IN IP4 xx.xx.231.18
t=0 0
m=audio 16660 RTP/AVP 0 122 101
a=rtpmap:0 PCMU/8000
a=rtpmap:122 t38/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
pbx*CLI>
<-- SIP read from xx.xx.xx.14:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.14:5060;branch=z9hG4bK197b4e79;rport
From: "xxxxx82766" <sip:[EMAIL PROTECTED]>;tag=as7acecfa4
To: <sip:[EMAIL PROTECTED]>;tag=as06f9c56a
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
--- (10 headers 0 lines) ---
set_destination: Parsing <sip:[EMAIL PROTECTED]> for address/port to send to
set_destination: set destination to xx.xx.xx.14, port 5060
T.38 UDPTL is at port xx.xx.231.18:16660
13 headers, 15 lines
Reliably Transmitting (NAT) to xx.xx.xx.14:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xx.xx.231.18:5060;branch=z9hG4bK6ccf3853;rport
From: <sip:[EMAIL PROTECTED]>;tag=as06f9c56a
To: "xxxxx82766" <sip:[EMAIL PROTECTED]>;tag=as7acecfa4
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: OpenPBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-openpbx-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 344
v=0
o=root 14512 14514 IN IP4 xx.xx.231.18
s=session
c=IN IP4 xx.xx.231.18
t=0 0
m=image 16660 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPFEC
---
Jan 28 13:14:08 NOTICE[311309]: app_faxdetect.c:306 detectfax_exec: Redirecting
SIP/xx.xx.xx.14-08231c80 to fax extension [DTMF]
-- Executing SipT38SwitchOver("SIP/xx.xx.xx.14-08231c80", "") in new stack
-- Executing Set("SIP/xx.xx.xx.14-08231c80",
"FAXFILE=/var/spool/fax/xxxxx30995_xxxxx82766_1169968445.1") in new stack
-- Executing RxFAX("SIP/xx.xx.xx.14-08231c80",
"/var/spool/fax/xxxxx30995_xxxxx82766_1169968445.1.tif") in new stack
Jan 28 13:14:09 NOTICE[311309]: rtp.c:356 process_rfc3389: Comfort noise
support incomplete in OpenPBX (RFC 3389). Please turn off on client if
possible. Client IP: xx.xx.167.212
pbx*CLI>
<-- SIP read from xx.xx.xx.14:5060:
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP
xx.xx.231.18:5060;branch=z9hG4bK6ccf3853;received=xx.xx.231.18;rport=5060
From: <sip:[EMAIL PROTECTED]>;tag=as06f9c56a
To: "xxxxx82766" <sip:[EMAIL PROTECTED]>;tag=as7acecfa4
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
--- (10 headers 0 lines) ---
-- Got SIP response 488 "Not acceptable here" back from xx.xx.xx.14
set_destination: Parsing <sip:[EMAIL PROTECTED]> for address/port to send to
set_destination: set destination to xx.xx.xx.14, port 5060
Transmitting (NAT) to xx.xx.xx.14:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xx.xx.231.18:5060;branch=z9hG4bK6ccf3853;rport
From: <sip:[EMAIL PROTECTED]>;tag=as06f9c56a
To: "xxxxx82766" <sip:[EMAIL PROTECTED]>;tag=as7acecfa4
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: OpenPBX
Max-Forwards: 70
Content-Length: 0
---
pbx*CLI>
<-- SIP read from xx.xx.xx.14:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.14:5060;branch=z9hG4bK7f843fe6;rport
From: "xxxxx82766" <sip:[EMAIL PROTECTED]>;tag=as7acecfa4
To: <sip:[EMAIL PROTECTED]>;tag=as06f9c56a
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 216
v=0
o=root 11023 11025 IN IP4 xx.xx.xx.14
s=session
c=IN IP4 xx.xx.xx.14
t=0 0
m=audio 25026 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (12 headers 10 lines) ---
Using INVITE request as basis request - [EMAIL PROTECTED]
Sending to xx.xx.xx.14 : 5060 (NAT)
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port xx.xx.xx.14:25026
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing),
combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
We're at xx.xx.231.18 port 16660
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to xx.xx.xx.14:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
xx.xx.xx.14:5060;branch=z9hG4bK7f843fe6;rport;received=xx.xx.xx.14
From: "xxxxx82766" <sip:[EMAIL PROTECTED]>;tag=as7acecfa4
To: <sip:[EMAIL PROTECTED]>;tag=as06f9c56a
Call-ID: [EMAIL PROTECTED]
CSeq: 104 INVITE
User-Agent: OpenPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:[EMAIL PROTECTED]>
Content-Type: application/sdp
Content-Length: 243
v=0
o=root 14512 14515 IN IP4 xx.xx.231.18
s=session
c=IN IP4 xx.xx.231.18
t=0 0
m=audio 16660 RTP/AVP 0 122 101
a=rtpmap:0 PCMU/8000
a=rtpmap:122 t38/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
pbx*CLI>
<-- SIP read from xx.xx.xx.14:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.14:5060;branch=z9hG4bK3437709e;rport
From: "xxxxx82766" <sip:[EMAIL PROTECTED]>;tag=as7acecfa4
To: <sip:[EMAIL PROTECTED]>;tag=as06f9c56a
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 104 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
--- (10 headers 0 lines) ---
<-- SIP read from xx.xx.xx.14:5060:
BYE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xx.xx.xx.14:5060;branch=z9hG4bK7a419064;rport
From: "xxxxx82766" <sip:[EMAIL PROTECTED]>;tag=as7acecfa4
To: <sip:[EMAIL PROTECTED]>;tag=as06f9c56a
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 105 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
--- (10 headers 0 lines) ---
Sending to xx.xx.xx.14 : 5060 (NAT)
Transmitting (NAT) to xx.xx.xx.14:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
xx.xx.xx.14:5060;branch=z9hG4bK7a419064;rport;received=xx.xx.xx.14
From: "xxxxx82766" <sip:[EMAIL PROTECTED]>;tag=as7acecfa4
To: <sip:[EMAIL PROTECTED]>;tag=as06f9c56a
Call-ID: [EMAIL PROTECTED]
CSeq: 105 BYE
User-Agent: OpenPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
X-OpenPBX-HangupCause: Bearer capability not available
---
Destroying call '[EMAIL PROTECTED]'
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