[OpenSIPS-Users] 488 Not acceptable here

[troxlinux]

Hi list , I am making some tests with a server opensips and  adds him
the rtpproxy for the nat, the problem is that when adding the nat and
to call to an extension that  don't answer it doesn't jump me to the
asterisk voicemail and it shows me an error 488

I explain that in the same server opensips I have installed asterisk
, in the asterisk cli when the call is not answered he throws me this
error:


WARNING[3178]: chan_sip.c:5201 process_sdp: Unable to lookup host in
c= line, 'IN IP4 192.168.10.3192.168.10.3'

the sdp writes it twice , as I can avoid this?

## log sip##


#
U +0.019539 192.168.10.30:5064 -> 192.168.10.3:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.3:5060;branch=0
From: sip:pinger at 192.168.10.3;tag=cd0baa81
To: sip:192.168.10.30:5064;tag=a8c59398c8984470
Call-ID: 9528c331-0c6e3641-f at 192.168.10.3
CSeq: 1 OPTIONS
User-Agent: Grandstream GXP2020 1.1.6.16
Contact: <sip:201 at 192.168.10.30:5064;transport=udp>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Supported: replaces, timer
Content-Length: 0


#
U +2.000872 192.168.10.3:5060 -> 192.168.10.3:5070
INVITE sip:u201 at 192.168.10.3:5070 SIP/2.0
Record-Route: <sip:192.168.10.3;lr=on;ftag=42d5a8fbdbb60640o0>
Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bKb977.26ab18f7.1
Via: SIP/2.0/UDP
192.168.10.19:5060;rport=5060;received=192.168.10.19;branch=z9hG4bK-a8ea22ed
From: <sip:200 at 192.168.10.3>;tag=42d5a8fbdbb60640o0
To: "Opensips-14x" <sip:201 at 192.168.10.3>
Call-ID: f6dccfd7-7f5fad14 at 192.168.10.19
CSeq: 102 INVITE
Max-Forwards: 69
Contact: <sip:200 at 192.168.10.19:5060;nat=yes;nat=yes>
Expires: 240
User-Agent: Linksys/SPA942-6.1.3(a)
Content-Length: 263
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
P-hint: inbound->inbound
P-hint: Route[20]: Rtpproxy
P-hint: Route[20]: Rtpproxy

v=0
o=- 811136 811136 IN IP4 192.168.10.19
s=-
c=IN IP4 192.168.10.3192.168.10.3
t=0 0
m=audio 3500435006 RTP/AVP 18 101
a=rtpmap:18 G729a/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
a=nortpproxy:yes
a=nortpproxy:yes

#
U +0.000123 192.168.10.3:5060 -> 192.168.10.30:5064
CANCEL sip:201 at 192.168.10.30:5064;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bKb977.26ab18f7.0
From: <sip:200 at 192.168.10.3>;tag=42d5a8fbdbb60640o0
Call-ID: f6dccfd7-7f5fad14 at 192.168.10.19
To: "Opensips-14x" <sip:201 at 192.168.10.3>
CSeq: 102 CANCEL
Max-Forwards: 70
User-Agent: OpenSIPS (1.4.5-notls (i386/linux))
Content-Length: 0


#
U +0.001572 192.168.10.3:5070 -> 192.168.10.3:5060
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP
192.168.10.3;branch=z9hG4bKb977.26ab18f7.1;received=192.168.10.3
Via: SIP/2.0/UDP
192.168.10.19:5060;rport=5060;received=192.168.10.19;branch=z9hG4bK-a8ea22ed
From: <sip:200 at 192.168.10.3>;tag=42d5a8fbdbb60640o0
To: "Opensips-14x" <sip:201 at 192.168.10.3>;tag=as50300fb2
Call-ID: f6dccfd7-7f5fad14 at 192.168.10.19
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


#
U +0.000244 192.168.10.3:5060 -> 192.168.10.3:5070
ACK sip:u201 at 192.168.10.3:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.3;branch=z9hG4bKb977.26ab18f7.1
From: <sip:200 at 192.168.10.3>;tag=42d5a8fbdbb60640o0
Call-ID: f6dccfd7-7f5fad14 at 192.168.10.19
To: "Opensips-14x" <sip:201 at 192.168.10.3>;tag=as50300fb2
CSeq: 102 ACK
Max-Forwards: 70
User-Agent: OpenSIPS (1.4.5-notls (i386/linux))
Content-Length: 0


regardss

-- 
rickygm

http://gnuforever.homelinux.com