Hello,
On 10/24/13 3:49 PM, anfecora wrote:
Hi all, can anyone help me to find out what is wrong
with my setup, i
have an asterisk behind a kamailio, kamailio is proxying all packages
to the outside.
when the call is bridge it gets disconnected after a few seconds, it
seems that our voip carrier is sending a bye because we didn't answer
to their 200 ok properly, but as the trace shows we did only that
kamailio is answering to the contact header ip not the ip that is
sending the ok.
the ACK has to be forwarded where the INVITE was forwarded, not where
the 200ok was forwarded.
We need full ngrep trace, from INVITE to BYE to see the contact and
record-route headers in INVITE/200ok in order to state what part of the
routing is going wrong there.
Cheers,
Daniel
I am sorry i sent a too long message before i will try skim it a bit.
any help is appreciated .
thanks.
my setup
request_route {
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
}
if(is_method("OPTIONS")) {
# send reply for each options request
sl_send_reply("200", "ok");
exit();
}
if(method=="BYE") {
#Account BYE transactions
};
if (method=="CANCEL") {
if (t_check_trans()) t_relay();
exit;
};
if (loose_route()) {
t_relay();
exit;
}
if (is_method("INVITE")) {
record_route();
}
f (!t_relay_to_udp("3.1.1.1", "5060")) {
sl_reply_error();
exit;
};
exit
};
here is a trace to a call made to a hotel.
i had changed the real ips for obvious reasons.
thanks.
asterisk ip 1.1.1.1
kamailio internal 1.1.1.2
kamailio external 2.0.0.1
Voip Carrier 3.1.1.1
voip contact ip 3.1.1.2
U 2013/10/23 17:26:20.354248 1.1.1.2:5060 <http://1.1.1.2:5060> ->
3.1.1.2:5060 <http://3.1.1.2:5060>
ACK sip:76890723276341079@3.1.1.2:5060
<http://sip:76890723276341079@3.1.1.2:5060> SIP/2.0.
Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport=5060.
Route:
<sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Max-Forwards: 16.
From: "+19812457865" <sip:+19812457865@1.1.1.1
<mailto:sip%3A%2B19812457865@1.1.1.1>>;tag=as4bc322e9.
To: <sip:76890723276341079@3.1.1.2
<mailto:sip%3A76890723276341079@3.1.1.2>>;tag=3591552407-393967.
Contact: <sip:+19812457865@1.1.1.1:5060
<http://sip:+19812457865@1.1.1.1:5060>>.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060
<http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060>.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 1.8.15-cert2.
Content-Length: 0.
.
U 2013/10/23 17:26:36.355580 3.1.1.1:5060 <http://3.1.1.1:5060> ->
1.1.1.2:5060 <http://1.1.1.2:5060>
BYE sip:+19812457865@1.1.1.1:5060
<http://sip:+19812457865@1.1.1.1:5060> SIP/2.0.
Max-Forwards: 69.
Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
To: "+19812457865" <sip:+19812457865@1.1.1.1
<mailto:sip%3A%2B19812457865@1.1.1.1>>;tag=as4bc322e9.
From: <sip:76890723276341079@3.1.1.1
<mailto:sip%3A76890723276341079@3.1.1.1>>;tag=3591552407-393967.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060
<http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060>.
CSeq: 2 BYE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO,
REFER, SUBSCRIBE, PRACK, UPDATE.
Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0.
Via: SIP/2.0/UDP
3.1.1.2:5060;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952.
Contact: <sip:76890723276341079@3.1.1.2:5060
<http://sip:76890723276341079@3.1.1.2:5060>>.
Content-Length: 0.
U 2013/10/23 17:26:20.354248 1.1.1.2:5060 <http://1.1.1.2:5060> ->
3.1.1.2:5060 <http://3.1.1.2:5060>
ACK sip:76890723276341079@3.1.1.2:5060
<http://sip:76890723276341079@3.1.1.2:5060> SIP/2.0.
Via: SIP/2.0/UDP 2.0.0.1:5060;branch=z9hG4bKcydzigwkX.
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK280c4e9c;rport=5060.
Route:
<sip:2.0.0.1;lr=on;ftag=as4bc322e9>,<sip:3.1.1.1;lr;ftag=as4bc322e9;did=8b8.d7ef5a05>.
Max-Forwards: 16.
From: "+19812457865" <sip:+19812457865@1.1.1.1
<mailto:sip%3A%2B19812457865@1.1.1.1>>;tag=as4bc322e9.
To: <sip:76890723276341079@3.1.1.2
<mailto:sip%3A76890723276341079@3.1.1.2>>;tag=3591552407-393967.
Contact: <sip:+19812457865@1.1.1.1:5060
<http://sip:+19812457865@1.1.1.1:5060>>.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060
<http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060>.
CSeq: 102 ACK.
User-Agent: Asterisk PBX 1.8.15-cert2.
Content-Length: 0.
.
U 2013/10/23 17:26:36.355580 3.1.1.1:5060 <http://3.1.1.1:5060> ->
1.1.1.2:5060 <http://1.1.1.2:5060>
BYE sip:+19812457865@1.1.1.1:5060
<http://sip:+19812457865@1.1.1.1:5060> SIP/2.0.
Max-Forwards: 69.
Route: <sip:2.0.0.1;lr=on;ftag=as4bc322e9>.
To: "+19812457865" <sip:+19812457865@1.1.1.1
<mailto:sip%3A%2B19812457865@1.1.1.1>>;tag=as4bc322e9.
From: <sip:76890723276341079@3.1.1.1
<mailto:sip%3A76890723276341079@3.1.1.1>>;tag=3591552407-393967.
Call-ID: 7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060
<http://7d0ca48c1d48c14d104fac1f59194ae0@1.1.1.1:5060>.
CSeq: 2 BYE.
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO,
REFER, SUBSCRIBE, PRACK, UPDATE.
Via: SIP/2.0/UDP 3.1.1.1:5060;branch=z9hG4bKce8a.db93afa3.0.
Via: SIP/2.0/UDP
3.1.1.2:5060;branch=z9hG4bK96003ecbb11f5deaf6014235140e6952.
Contact: <sip:76890723276341079@3.1.1.2:5060
<http://sip:76890723276341079@3.1.1.2:5060>>.
Content-Length: 0.
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