This issue is already fixed in the 1.2 svn.
Regards,
Ovidiu Sas
On 5/7/07, mailinglist-openser-users
<mailinglist-openser-users(a)e-qual.fr> wrote:
Version: OpenSer 1.2.0
I have a phone (using G711 as voice codec ) using openser as outbound
proxy, openser has 2 routes
- gateway1 (does not allow G711)
- gateway2
When the phone try a call, openser try first gateway1 and that fails with
error code "488 Not acceptable here" ( which is normal because G711 is not
allowed on that gateway ). Then Openser then try gateway2 but it seems that
Openser is generating wrong SIP packets especially, it is using header
"transport=udp..t" instead of "transport=udp".
Is this a known bug on LCR fallback ?
Note that on the first gateway1, one digit is stripped before trying the
call; on the second gateway, no digits are stripped.
Below is an extract of the SIP dialog ( ngrep output in attachement ):
o Phone to OpenSer
U phone:5060 -> openser:5060
INVITE sip:xxxxxxxx@sip1.e-qual.fr SIP/2.0.
U openser:5060 -> phone:5060
SIP/2.0 100 Giving a try.
o OpenSer to Gateway1
U openser:5060 -> gateway1:5060
INVITE sip:xxxxxxxx@gateway1:5060;transport=udp SIP/2.0.
U gateway1:5060 -> openser:5060
SIP/2.0 488 Not acceptable here.
U openser:5060 -> gateway1:5060
ACK sip:516442431@gateway1:5060;transport=udp SIP/2.0.
o OpenSer to Gateway2
U openser:5060 -> gateway2:5060
INVITE sip:xxxxxxxx@gateway2:5060;transport=udp..t SIP/2.0.
U openser:5060 -> gateway2:5060
INVITE sip:xxxxxxxx@gateway2:5060;transport=udp..t SIP/2.0.
U openser:5060 -> gateway2:5060
INVITE sip:xxxxxxxx@gateway2:5060;transport=udp..t SIP/2.0.
Regards,
Jerome
--
Jerome
gallois:/var/log# sudo /usr/bin/ngrep -W byline "" "port 5060"
|more
filter: (ip or ip6) and ( port 5060 )
#
U phone:5060 -> openser:5060
INVITE sip:xxxxxxxx@sip1.e-qual.fr SIP/2.0.
Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone.
Contact: <sip:phone>.
Content-Length: 221.
Content-Type: application/sdp.
CSeq: 1 INVITE.
From: <sip:sip1.e-qual.fr>;tag=c0a80843-16.
Max-Forwards: 70.
Session-GUID: 875967334-858992696-842282596-875782963.
Supported: 100rel.
To: <sip:xxxxxxxx@sip1.e-qual.fr>.
User-Agent: Quintum/1.0.0.
Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c.
Quintum:
0c01030b0237370501001301000f0b413031322d323032334332.
.
v=0.
o=Quintum 17 17 IN IP4 phone.
s=VoipCall.
c=IN IP4 phone.
t=0 0.
m=audio 10274 RTP/AVP 4 101.
c=IN IP4 phone.
a=rtpmap:4 g723/8000/1.
a=ptime:30.
a=rtpmap:101 telephone-event/8000/1.
a=sendrecv.
#
U openser:5060 -> phone:5060
SIP/2.0 100 Giving a try.
Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone.
CSeq: 1 INVITE.
From: <sip:sip1.e-qual.fr>;tag=c0a80843-16.
To: <sip:xxxxxxxx@sip1.e-qual.fr>.
Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c.
Server: OpenSER (1.2.0-notls (i386/linux)).
Content-Length: 0.
.
#
U openser:5060 -> gateway1:5060
INVITE sip:xxxxxxxxx@gateway1:5060;transport=udp SIP/2.0.
Record-Route: <sip:openser;lr=on;ftag=c0a80843-16>.
Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone.
Contact: <sip:phone>.
Content-Length: 221.
Content-Type: application/sdp.
CSeq: 1 INVITE.
From: <sip:sip1.e-qual.fr>;tag=c0a80843-16.
Max-Forwards: 69.
Session-GUID: 875967334-858992696-842282596-875782963.
Supported: 100rel.
To: <sip:xxxxxxxx@sip1.e-qual.fr>.
User-Agent: Quintum/1.0.0.
Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.0.
Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c.
Quintum:
0c01030b0237370501001301000f0b413031322d323032334332.
P-hint: outbound PSTN.
.
v=0.
o=Quintum 17 17 IN IP4 phone.
s=VoipCall.
c=IN IP4 phone.
t=0 0.
m=audio 10274 RTP/AVP 4 101.
c=IN IP4 phone.
a=rtpmap:4 g723/8000/1.
a=ptime:30.
a=rtpmap:101 telephone-event/8000/1.
a=sendrecv.
#
U gateway1:5060 -> openser:5060
SIP/2.0 488 Not acceptable here.
Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.0;received=openser.
Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c.
From: <sip:sip1.e-qual.fr>;tag=c0a80843-16.
To: <sip:xxxxxxxx@sip1.e-qual.fr>;tag=as714f5f56.
Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone.
CSeq: 1 INVITE.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Contact: <sip:xxxxxxxxx@gateway1>.
Content-Length: 0.
.
#
U openser:5060 -> gateway1:5060
ACK sip:xxxxxxxxx@gateway1:5060;transport=udp SIP/2.0.
Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.0.
From: <sip:sip1.e-qual.fr>;tag=c0a80843-16.
Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone.
To: <sip:xxxxxxxx@sip1.e-qual.fr>;tag=as714f5f56.
CSeq: 1 ACK.
User-Agent: OpenSER (1.2.0-notls (i386/linux)).
Content-Length: 0.
.
#
U openser:5060 -> gateway2:5060
INVITE sip:xxxxxxxx@gateway2:5060;transport=udp..t SIP/2.0.
Record-Route: <sip:openser;lr=on;ftag=c0a80843-16>.
Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone.
Contact: <sip:phone>.
Content-Length: 221.
Content-Type: application/sdp.
CSeq: 1 INVITE.
From: <sip:sip1.e-qual.fr>;tag=c0a80843-16.
Max-Forwards: 69.
Session-GUID: 875967334-858992696-842282596-875782963.
Supported: 100rel.
To: <sip:xxxxxxxx@sip1.e-qual.fr>.
User-Agent: Quintum/1.0.0.
Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.1.
Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c.
Quintum:
0c01030b0237370501001301000f0b413031322d323032334332.
P-hint: outbound PSTN.
.
v=0.
o=Quintum 17 17 IN IP4 phone.
s=VoipCall.
c=IN IP4 phone.
t=0 0.
m=audio 10274 RTP/AVP 4 101.
c=IN IP4 phone.
a=rtpmap:4 g723/8000/1.
a=ptime:30.
a=rtpmap:101 telephone-event/8000/1.
a=sendrecv.
#
U openser:5060 -> gateway2:5060
INVITE sip:xxxxxxxx@gateway2:5060;transport=udp..t SIP/2.0.
Record-Route: <sip:openser;lr=on;ftag=c0a80843-16>.
Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone.
Contact: <sip:phone>.
Content-Length: 221.
Content-Type: application/sdp.
CSeq: 1 INVITE.
From: <sip:sip1.e-qual.fr>;tag=c0a80843-16.
Max-Forwards: 69.
Session-GUID: 875967334-858992696-842282596-875782963.
Supported: 100rel.
To: <sip:xxxxxxxx@sip1.e-qual.fr>.
User-Agent: Quintum/1.0.0.
Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.1.
Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c.
Quintum:
0c01030b0237370501001301000f0b413031322d323032334332.
P-hint: outbound PSTN.
.
v=0.
o=Quintum 17 17 IN IP4 phone.
s=VoipCall.
c=IN IP4 phone.
t=0 0.
m=audio 10274 RTP/AVP 4 101.
c=IN IP4 phone.
a=rtpmap:4 g723/8000/1.
a=ptime:30.
a=rtpmap:101 telephone-event/8000/1.
a=sendrecv.
#
U openser:5060 -> gateway2:5060
INVITE sip:xxxxxxxx@gateway2:5060;transport=udp..t SIP/2.0.
Record-Route: <sip:openser;lr=on;ftag=c0a80843-16>.
Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone.
Contact: <sip:phone>.
Content-Length: 221.
Content-Type: application/sdp.
CSeq: 1 INVITE.
From: <sip:sip1.e-qual.fr>;tag=c0a80843-16.
Max-Forwards: 69.
Session-GUID: 875967334-858992696-842282596-875782963.
Supported: 100rel.
To: <sip:xxxxxxxx@sip1.e-qual.fr>.
User-Agent: Quintum/1.0.0.
Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.1.
Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c.
Quintum:
0c01030b0237370501001301000f0b413031322d323032334332.
P-hint: outbound PSTN.
.
v=0.
o=Quintum 17 17 IN IP4 phone.
s=VoipCall.
c=IN IP4 phone.
t=0 0.
m=audio 10274 RTP/AVP 4 101.
c=IN IP4 phone.
a=rtpmap:4 g723/8000/1.
a=ptime:30.
a=rtpmap:101 telephone-event/8000/1.
a=sendrecv.
#
U openser:5060 -> gateway2:5060
INVITE sip:xxxxxxxx@gateway2:5060;transport=udp..t SIP/2.0.
Record-Route: <sip:openser;lr=on;ftag=c0a80843-16>.
Call-ID: call-717C8C1D-CEDE-2910-0D00-26@phone.
Contact: <sip:phone>.
Content-Length: 221.
Content-Type: application/sdp.
CSeq: 1 INVITE.
From: <sip:sip1.e-qual.fr>;tag=c0a80843-16.
Max-Forwards: 69.
Session-GUID: 875967334-858992696-842282596-875782963.
Supported: 100rel.
To: <sip:xxxxxxxx@sip1.e-qual.fr>.
User-Agent: Quintum/1.0.0.
Via: SIP/2.0/UDP openser;branch=z9hG4bKe7a2.b7f82867.1.
Via: SIP/2.0/UDP phone;branch=z9hG4bK-tenor-c0a8-0843-001c.
Quintum:
0c01030b0237370501001301000f0b413031322d323032334332.
P-hint: outbound PSTN.
.
v=0.
o=Quintum 17 17 IN IP4 phone.
s=VoipCall.
c=IN IP4 phone.
t=0 0.
m=audio 10274 RTP/AVP 4 101.
c=IN IP4 phone.
a=rtpmap:4 g723/8000/1.
a=ptime:30.
a=rtpmap:101 telephone-event/8000/1.
a=sendrecv.
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