12 Nov
2008
12 Nov
'08
8:33 a.m.
Hi list, I am giving him returned script from openser trying to
find the problem to this message " 488 acceptable Not here" , I have my openser
integrated recently with asterisk and the voicemail recently adds rtpproxy to solve
problems of nat and was almost a success, some details to improve, but after adding the
rtpproxy when I call to UAC and it does not answer the call on the telephone shows a 488
message to me No Aceptable, it does not jump to the voicemai..
I have 2 networks cards in my server, and looking at the sip log, I see twice in the sdp
the interfaz it public
any idea as solving this problem?
regards
rickygm
#
U +1.675624 192.168.10.1:5060 -> 192.168.10.1:5070
INVITE sip:u120@192.168.10.1:5070 SIP/2.0
Record-Route: <sip:192.168.10.1;lr=on;ftag=69372fe200a41663>
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.1
Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
To: <sip:120@192.168.10.1>
Contact: <sip:119@192.168.10.28:5060;nat=yes;nat=yes>
Supported: replaces, timer, path
Call-ID: 7c55e4667d473f76(a)192.168.10.28
CSeq: 12216 INVITE
User-Agent: Grandstream GXV3000 1.1.3.14
Max-Forwards: 69
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 604
P-hint: inbound->inbound
P-hint: Route[20]: Rtpproxy
P-hint: Route[20]: Rtpproxy
v=0
o=119 8000 8001 IN IP4 192.168.10.28
s=SIP Call
c=IN IP4 192.168.1.64192.168.1.64
t=0 0
m=audio 3504835048 RTP/AVP 18 4 3 2 0 101
a=sendrecv
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 3505035050 RTP/AVP 99 34
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428014; packetization-mode=0;
sprop-parameter-sets=Z0KADJWgUH5A,aM4CXID=
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=2 MaxBR=1280
a=framerate:20
a=nortpproxy:yes
a=nortpproxy:yes
#
U +0.000052 192.168.10.1:5060 -> 192.168.10.27:5060
CANCEL sip:120@192.168.10.27:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
Call-ID: 7c55e4667d473f76(a)192.168.10.28
To: <sip:120@192.168.10.1>
CSeq: 12216 CANCEL
Max-Forwards: 70
User-Agent: OpenSER (1.3.2-notls (i386/linux))
Content-Length: 0
#
U +0.000411 192.168.10.1:5070 -> 192.168.10.1:5060
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.1;received=192.168.10.1
Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
To: <sip:120@192.168.10.1>;tag=as4f1165d5
Call-ID: 7c55e4667d473f76(a)192.168.10.28
CSeq: 12216 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
#
U +0.000120 192.168.10.1:5060 -> 192.168.10.1:5070
ACK sip:u120@192.168.10.1:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.1
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
Call-ID: 7c55e4667d473f76(a)192.168.10.28
To: <sip:120@192.168.10.1>;tag=as4f1165d5
CSeq: 12216 ACK
Max-Forwards: 70
User-Agent: OpenSER (1.3.2-notls (i386/linux))
Content-Length: 0
#
U +0.000144 192.168.10.1:5060 -> 192.168.10.28:5060
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
To: <sip:120@192.168.10.1>;tag=as4f1165d5
Call-ID: 7c55e4667d473f76(a)192.168.10.28
CSeq: 12216 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
P-hint: Onreply-route - fixcontact
#
U +0.000787 192.168.10.27:5060 -> 192.168.10.1:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
To: <sip:120@192.168.10.1>;tag=fb538483d0333de1
Call-ID: 7c55e4667d473f76(a)192.168.10.28
CSeq: 12216 CANCEL
User-Agent: Grandstream GXV3000 1.1.3.14
Supported: replaces, timer, 100rel, path
Content-Length: 0
#
U +0.000544 192.168.10.27:5060 -> 192.168.10.1:5060
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0
Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3
Record-Route: <sip:192.168.10.1;lr=on;ftag=69372fe200a41663>
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
To: <sip:120@192.168.10.1>;tag=fb538483d0333de1
Call-ID: 7c55e4667d473f76(a)192.168.10.28
CSeq: 12216 INVITE
User-Agent: Grandstream GXV3000 1.1.3.14
Content-Length: 0
#
U +0.000067 192.168.10.1:5060 -> 192.168.10.27:5060
ACK sip:120@192.168.10.27:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
Call-ID: 7c55e4667d473f76(a)192.168.10.28
To: <sip:120@192.168.10.1>;tag=fb538483d0333de1
CSeq: 12216 ACK
Max-Forwards: 70
User-Agent: OpenSER (1.3.2-notls (i386/linux))
Content-Length: 0
#
U +0.001390 192.168.10.28:5060 -> 192.168.10.1:5060
ACK sip:120@192.168.10.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bK286620c52595f4b3
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
To: <sip:120@192.168.10.1>;tag=as4f1165d5
Contact: <sip:119@192.168.10.28:5060>
Proxy-Authorization: Digest username="119", realm="192.168.10.1",
algorithm=MD5, uri="sip:120@192.168.10.1",
nonce="4919edaf0a873106bfd0b18f347709360f00f7eb",
response="a5a26c257b491718def2b43488a39853"
Call-ID: 7c55e4667d473f76(a)192.168.10.28
CSeq: 12216 ACK
User-Agent: Grandstream GXV3000 1.1.3.14
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
#
U +0.497023 192.168.10.1:5060 -> 192.168.10.28:5060
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
To: <sip:120@192.168.10.1>;tag=as4f1165d5
Call-ID: 7c55e4667d473f76(a)192.168.10.28
CSeq: 12216 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
P-hint: Onreply-route - fixcontact
#
U +0.999813 192.168.10.1:5060 -> 192.168.10.28:5060
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
To: <sip:120@192.168.10.1>;tag=as4f1165d5
Call-ID: 7c55e4667d473f76(a)192.168.10.28
CSeq: 12216 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
P-hint: Onreply-route - fixcontact
exit
67 received, 0 dropped
12 Nov
12 Nov
11:26 a.m.
Hi Ricky!
looks like you activate the rtp proxy two times for the same request.
Fix your config.
regards
klaus
Ricky Gutierrez schrieb:
Hi list, I am giving him returned script from openser
trying to find
the problem to this message " 488 acceptable Not here" , I have my
openser integrated recently with asterisk and the voicemail recently
adds rtpproxy to solve problems of nat and was almost a success, some
details to improve, but after adding the rtpproxy when I call to UAC and
it does not answer the call on the telephone shows a 488 message to me
No Aceptable, it does not jump to the voicemai..
I have 2 networks cards in my server, and looking at the sip log, I see
twice in the sdp the interfaz it public
any idea as solving this problem?
regards
rickygm
#
U +1.675624 192.168.10.1:5060 -> 192.168.10.1:5070
INVITE sip:u120@192.168.10.1:5070 SIP/2.0
Record-Route: <sip:192.168.10.1;lr=on;ftag=69372fe200a41663>
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.1
Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3
From:
<sip:119@192.168.10.1>;tag=69372fe200a41663
To: <sip:120@192.168.10.1>
Contact: <sip:119@192.168.10.28:5060;nat=yes;nat=yes>
Supported: replaces, timer, path
Call-ID: 7c55e4667d473f76(a)192.168.10.28
CSeq: 12216 INVITE
User-Agent: Grandstream GXV3000 1.1.3.14
Max-Forwards: 69
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 604
P-hint: inbound->inbound
P-hint: Route[20]: Rtpproxy
P-hint: Route[20]: Rtpproxy
v=0
o=119 8000 8001 IN IP4 192.168.10.28
s=SIP Call
c=IN IP4 192.168.1.64192.168.1.64
t=0 0
m=audio 3504835048 RTP/AVP 18 4 3 2 0 101
a=sendrecv
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 3505035050 RTP/AVP 99 34
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=428014; packetization-mode=0;
sprop-parameter-sets=Z0KADJWgUH5A,aM4CXID=
a=rtpmap:34 H263/90000
a=fmtp:34 CIF=2 MaxBR=1280
a=framerate:20
a=nortpproxy:yes
a=nortpproxy:yes
#
U +0.000052 192.168.10.1:5060 -> 192.168.10.27:5060
CANCEL sip:120@192.168.10.27:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
Call-ID: 7c55e4667d473f76(a)192.168.10.28
To: <sip:120@192.168.10.1>
CSeq: 12216 CANCEL
Max-Forwards: 70
User-Agent: OpenSER (1.3.2-notls (i386/linux))
Content-Length: 0
#
U +0.000411 192.168.10.1:5070 -> 192.168.10.1:5060
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.1;received=192.168.10.1
Via:
SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
To: <sip:120@192.168.10.1>;tag=as4f1165d5
Call-ID: 7c55e4667d473f76(a)192.168.10.28
CSeq: 12216 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
#
U +0.000120 192.168.10.1:5060 -> 192.168.10.1:5070
ACK sip:u120@192.168.10.1:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.1
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
Call-ID: 7c55e4667d473f76(a)192.168.10.28
To: <sip:120@192.168.10.1>;tag=as4f1165d5
CSeq: 12216 ACK
Max-Forwards: 70
User-Agent: OpenSER (1.3.2-notls (i386/linux))
Content-Length: 0
#
U +0.000144 192.168.10.1:5060 -> 192.168.10.28:5060
SIP/2.0 488 Not acceptable here
Via:
SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
To: <sip:120@192.168.10.1>;tag=as4f1165d5
Call-ID: 7c55e4667d473f76(a)192.168.10.28
CSeq: 12216 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
P-hint: Onreply-route - fixcontact
#
U +0.000787 192.168.10.27:5060 -> 192.168.10.1:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
To: <sip:120@192.168.10.1>;tag=fb538483d0333de1
Call-ID: 7c55e4667d473f76(a)192.168.10.28
CSeq: 12216 CANCEL
User-Agent: Grandstream GXV3000 1.1.3.14
Supported: replaces, timer, 100rel, path
Content-Length: 0
#
U +0.000544 192.168.10.27:5060 ->
192.168.10.1:5060
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0
Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3
Record-Route: <sip:192.168.10.1;lr=on;ftag=69372fe200a41663>
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
To: <sip:120@192.168.10.1>;tag=fb538483d0333de1
Call-ID: 7c55e4667d473f76(a)192.168.10.28
CSeq: 12216 INVITE
User-Agent: Grandstream GXV3000 1.1.3.14
Content-Length: 0
#
U +0.000067 192.168.10.1:5060 -> 192.168.10.27:5060
ACK sip:120@192.168.10.27:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1;branch=z9hG4bK78e7.cd89b213.0
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
Call-ID: 7c55e4667d473f76(a)192.168.10.28
To: <sip:120@192.168.10.1>;tag=fb538483d0333de1
CSeq: 12216 ACK
Max-Forwards: 70
User-Agent: OpenSER (1.3.2-notls (i386/linux))
Content-Length: 0
#
U +0.001390 192.168.10.28:5060 -> 192.168.10.1:5060
ACK sip:120@192.168.10.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.28:5060;branch=z9hG4bK286620c52595f4b3
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
To: <sip:120@192.168.10.1>;tag=as4f1165d5
Contact: <sip:119@192.168.10.28:5060>
Proxy-Authorization: Digest username="119", realm="192.168.10.1",
algorithm=MD5, uri="sip:120@192.168.10.1",
nonce="4919edaf0a873106bfd0b18f347709360f00f7eb",
response="a5a26c257b491718def2b43488a39853"
Call-ID: 7c55e4667d473f76(a)192.168.10.28
CSeq: 12216 ACK
User-Agent: Grandstream GXV3000 1.1.3.14
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
#
U +0.497023 192.168.10.1:5060 -> 192.168.10.28:5060
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP
192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
To: <sip:120@192.168.10.1>;tag=as4f1165d5
Call-ID: 7c55e4667d473f76(a)192.168.10.28
CSeq: 12216 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
P-hint: Onreply-route - fixcontact
#
U +0.999813 192.168.10.1:5060 -> 192.168.10.28:5060
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.10.28:5060;rport=5060;branch=z9hG4bK286620c52595f4b3
From: <sip:119@192.168.10.1>;tag=69372fe200a41663
To: <sip:120@192.168.10.1>;tag=as4f1165d5
Call-ID: 7c55e4667d473f76(a)192.168.10.28
CSeq: 12216 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
P-hint: Onreply-route
- fixcontact
exit
67 received, 0 dropped
------------------------------------------------------------------------
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7:58 p.m.
hi Klaus, you could explain this part more in detail to me? , I do
not have much experience with rtpproxy but it would be thankful for an
explanation ..
best regards
rickygm
________________________________
Wednesday, November 12, 2008 3:26:08 AM Klaus Darilion wrote:
Hi Ricky!
looks like you activate the rtp proxy two times for the same request. Fix your config.
13 Nov
13 Nov
10:56 a.m.
force_rtp_proxy() replaces the IP ind the SDP with the IP of the rtp proxy.
if you have in your config:
force_rtp_proxy();
...
...
...
force_rtp_proxy();
t_relay():
then the SDP is corrupted - like in your case.
Create a config file with the wizard on sipwise.com and analyse it -
then you will know how correct NAT traversal works.
klaus
Ricky Gutierrez schrieb:
hi Klaus, you could explain this part more in detail to me? , I do not
have much experience with rtpproxy but it would be thankful for an
explanation ..
best regards
rickygm
------------------------------------------------------------------------
** Wednesday, November 12, 2008 3:26:08 AM Klaus Darilion wrote:
Hi Ricky!
looks like you activate the rtp proxy two times for the same request.
Fix your config.
5855
days inactive
5856
days old
3 comments
2 participants
participants (2)
-
Klaus Darilion -
Ricky Gutierrez