Hello,
I found it hard to follow what is wrong here, can you get the SIP
traffic on kamailio server and post it here? You can use:
ngrep -d any -qt -W byline port 5060
on your kamailio server.
Then we can see if it something wrong there and who is responsible for it.
Kamailio is hardly changing anything in the sip messages, unless you
instruct it to do so via config. Changing the Contact is for sure a
matter of config file.
Cheers,
Daniel
On 4/29/10 12:16 PM, Konstantin Shpinev wrote:
Hi!
I have a problem with Kamailio.Sometimes (can not catch the moment and conditions of)
situations arise when, calling from AAA to uri XXX call comes to uri YYY.The call comes
from PSTN to the gray subnet of user-agents registered in kamailio.As the PSTN we using
Audiocodes mediant.
Kamailio ver. 1.5.3
Mediant host: 172.19.32.2, Kamailio host: 172.19.32.3
XXX host: 172.19.32.33, YYY host: 192.168.1.203
I put a line in the log section route_branch:
Apr 28 12:59:20 sipproxy /ks/sbin/kamailio[18294]: [INFO] new
branch at sip:XXX@172.19.32.33:5060;user=phone
Then in the log module acc wrote:
Apr 28 12:59:21 sipproxy /ks/sbin/kamailio[18302]: ACC:
transaction answered:
timestamp=1272473961;method=BYE;from_tag=1e76ae41;to_tag=1c1795565414;call_id=17
955646162422000234627(a)172.19.32.2
<mailto:955646162422000234627@172.19.32.2>;code=481;reason=Call/Transaction
Does Not Exist;from_uri=sip:XXX@172.19.32.3
<mailto:sip%3AXXX@172.19.32.3>;from_username=XXX;from_name=;from_domain=
172.19.32.3;to_uri=sip:AAA@172.19.32.2
<mailto:sip%3AAAA@172.19.32.2>;to_username=AAA;to_name=;to_domain=172.19.32.2;request_uri=sip:AAA@172.19.32.2
<mailto:sip%3AAAA@172.19.32.2>;request_username=AAA;route_id=;route_name=;route_type_id=;destination=;calllist_id=
Despite the error, the call passes, but to the another uri.
Here's the log of the mediant:
1:
Apr 28 12:52:31 172.19.32.2 ( lgr_flow)(44244101 ) ----
Outgoing SIP Message to 172.19.32.3:5060 <http://172.19.32.3:5060>
----
Apr 28 12:52:31 172.19.32.2 ACK
sip:XXX@172.19.32.33:5060;user=phone SIP/2.0^M Via: SIP/2.0/UDP
172.19.32.2;branch=z9hG4bKac1897823584^M Max-Forwards: 70^M From:
<sip:AAA@172.19.32.2
<mailto:sip%3AAAA@172.19.32.2>>;tag=1c1795565414^M To:
<sip:XXX@172.19.32.3
<mailto:sip%3AXXX@172.19.32.3>>;tag=1e76ae41^M Call-ID:
17955646162422000234627(a)172.19.32.2
<mailto:17955646162422000234627@172.19.32.2>^M CSeq: 1
ACK Contact: <sip:AAA@172.19.32.2
<mailto:sip%3AAAA@172.19.32.2>>^M Route:
<sip:172.19.32.3;lr;ftag=1c1795565414;vsf=AAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAAA->^M
Supported:
em,timer,replaces,path,early-session,resource-priority^M Allow:
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE^M
*User-Agent: Audiocodes-Sip-Gateway-Mediant
2000/v.5.00A.045.003*^M Content-Length: 0
As you can see, in 1. all is well. But then:
2:
..... ---- Incoming SIP Message from 172.19.32.3:5060
<http://172.19.32.3:5060> ----
....From: <sip:AAA@172.19.32.2
<mailto:sip%3AAAA@172.19.32.2>>;tag=1c1795565414^M Call-ID:
17955646162422000234627(a)172.19.32.2
<mailto:17955646162422000234627@172.19.32.2>^M CSeq: 1 INVITE^M
Allow: INVITE, ACK, CANCEL,OPTIONS, BYE, NOTIFY, REFER, MESSAGE,
OPTIONS, INFO^M Content-Type: application/sdp^M *User-Agent:
Zoiper for Windows rev.1105*^M Content-Length: 232^M ^M v=0^M
*o=Zoiper_user 0 0 IN IP4 192.168.1.203^M s=Zoiper_session^M c=IN
IP4 192.168.1.203^*M t=0 0^M m=audio 8000 RTP/AVP 8 0 101^M
a=rtpmap:8 PCMA/8000^M a=rtpmap:0 PCMU/8000^M a=rtpmap:101
telephone-event/8000^M a=fmtp:101 0-15^M a=sendrecv
And that is where the substitution occurred at the data fields
'Contact' an entirely different user-agent (uri YYY), which is also
registered on Kamailio. That it eventually comes to call. *Repeat: In
second SIP message from Kamailio 'Contact' field is replaced by
'Contact' of another (!) user-agent (YYY).*
There is a suspicion that an error occurs in a block:
if (!lookup("location")) {
switch ($retcode) {
case -3:
...
case -1:
...
case -2:
...
}
}
}
I myself anywhere 'Contact' field are not affected, so all suspected
to*locate()*from module Registrar.
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