Re: [SR-Users] when mySQL is unreachable ...
On 4/13/10 8:39 PM, Alex rsm wrote: Hi, My Kamailio server uses an external mysql database with perl script to proxy traffic. When mysql server is unreachable, Kamailio do not respond with 100 Trying and instead responses with 484 (Address Incomplete) to the INVITE message. Is this a normal behavior? I need to response with 503 when mysql is unreachable. Can I do it in kamailio.cfg or I need to implement it in the perl script? kamailio is replying automatically in very few cases. In case you use default confing, then 484 address incomplete is coming from the check for calls not having a username in r-uri. 100 trying is sent by t_relay(). You can send 503 from config as you need. mysql connectivity is hidden behind some modules, like authentication, which will return false and from config will be a new challenge (in this particular case). mysql module has support for auto-connect. Cheers, Daniel Thanks, A R The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. Get busy. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4 ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla * http://www.asipto.com/ * http://twitter.com/miconda * http://www.linkedin.com/in/danielconstantinmierla ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] [NAT] BYE not received to caller
Hi
I need some guidelines to troubleshoot the following issue:
a) A is behind NAT
b) B is not behind NAT
c) A calls B, SIP INVITE is sent over TCP
d) A's firewall does NAT and changes the source port to let's say p1
e) B releases the call and sends BYE over UDP
f) Kamailio sends the BYE to A, over UDP, to the NATed source port p1 ?!
2 comments:
- This scenario works perfectly when A is the one who disconnects the call.
and of course when no NAT is involved everything works ok
- if works when I comment the line fix_nated_contact() in the route[NAT]
block:
route[NAT]{
#!ifdef WITH_NAT
force_rport();
#if (nat_uac_test(19)) {
if (nat_uac_test(3)) {
if (method==REGISTER) {
setbflag(10);
fix_nated_register();
} else {
#fix_nated_contact();
[...]
Any hint is very welcome.
Cheers
Pascal
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Re: [SR-Users] [NAT] BYE not received to caller
Klaus
I attached the ngrep you asked me to this email.
Regards,
Pascal
On Wed, Apr 14, 2010 at 12:21 PM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
The contact after fix_nated_contact() should also contain ;transport=tcp.
Thus, Kamailio should relay the BYE with TCP.
Can you show an ngrep dump (ngrep -W byline -t -q -P port 5060) of the
problematic scenario?
regards
klaus
PS: A more standard-conform way of rewriting the SDP is to use the
add_contact_alias() and handle_ruri_alias() functions:
http://sip-router.org/docbook/sip-router/branch/master/modules_k/nathelper/nathelper.html#id2601711
regards
klaus
Am 14.04.2010 12:06, schrieb Pascal Maugeri:
Hi
I need some guidelines to troubleshoot the following issue:
a) A is behind NAT
b) B is not behind NAT
c) A calls B, SIP INVITE is sent over TCP
d) A's firewall does NAT and changes the source port to let's say p1
e) B releases the call and sends BYE over UDP
f) Kamailio sends the BYE to A, over UDP, to the NATed source port p1 ?!
2 comments:
- This scenario works perfectly when A is the one who disconnects the
call. and of course when no NAT is involved everything works ok
- if works when I comment the line fix_nated_contact() in the
route[NAT] block:
route[NAT]{
#!ifdef WITH_NAT
force_rport();
#if (nat_uac_test(19)) {
if (nat_uac_test(3)) {
if (method==REGISTER) {
setbflag(10);
fix_nated_register();
} else {
#fix_nated_contact();
[...]
Any hint is very welcome.
Cheers
Pascal
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interface: eth0 (213.151.105.168/255.255.255.252)
filter: (ip or ip6) and ( port )
T 2010/04/14 13:04:09.058608 80.36.XXX.XXX:27270 - 213.151.XXX.XXX: [A]
T 2010/04/14 13:04:09.095454 80.36.XXX.XXX:27270 - 213.151.XXX.XXX: [A]
INVITE sip:ad...@test.net;transport=tcp SIP/2.0
Call-ID: 7a0ca2997d08c9f64b58e3b5db289...@10.1.1.22
CSeq: 1 INVITE
From: miquel sip:miq...@test.net;tag=89376139
To: sip:ad...@test.net
Via: SIP/2.0/TCP 10.1.1.22:5060;branch=z9hG4bKVuwkLMxhU
Max-Forwards: 70
Contact: sip:miq...@10.1.1.22:5060;+g.poc.talkburst;+g.poc.groupad
P-Preferred-Identity: miquel sip:miq...@test.net
Accept-Contact: *;+g.poc.talkburst;require;explicit
Supported: timer
Session-Expires: 1800;refresher=uac
Require: pref
User-Agent: PoC-client/OMA1.0 GenakerOMAPoCSDK/2.0
Content-Type: multipart/mixed;boundary=Genaker-boundary
Content-Length: 783
--Genaker-boundary
Content-Type: application/sdp
v=0
o=miquel 1271242986331 1271242986331 IN IP4 10.1.1.22
s=adhoc
c=IN IP4 10.1.1.22
t=0 0
m=audio 4502 RTP/AVP 105
a=rtcp:4503 IN IP4 10.1.1.22
a=sendrecv
a=rtpmap:105 AMR/8000
a=ptime:100
a=maxptime:400
a=fmtp:105 mode-set=0,1,2,3,4,5,6,7; octet-align=1
m=application 4503 udp TBCP
a=fmtp:TBCP queuing=0; tb_priority=1; timestamp=0; tb_granted=0;
poc_sess_priority=1; poc_lock=0
--Genaker-boundary
Content-Type: application/resource-lists+xml
?xml version=1.0 encoding=UTF-8?
resource-lists xmlns
T 2010/04/14 13:04:09.098795 80.36.XXX.XXX:27270 - 213.151.XXX.XXX: [AP]
=urn:ietf:params:xml:ns:resource-lists
xmlns:xsi=http://www.w3.org/2001/XMLSchema-instance;
list
entry uri=sip:m...@test.net /
/list
/resource-lists
--Genaker-boundary--
T 2010/04/14 13:04:09.100679 213.151.XXX.XXX: - 80.36.XXX.XXX:27270 [AP]
SIP/2.0 100 trying -- your call is important to us
Call-ID: 7a0ca2997d08c9f64b58e3b5db289...@10.1.1.22
CSeq: 1 INVITE
From: miquel sip:miq...@test.net;tag=89376139
To: sip:ad...@test.net
Via: SIP/2.0/TCP
10.1.1.22:5060;branch=z9hG4bKVuwkLMxhU;rport=27270;received=80.36.XXX.XXX
Server: Genaker SIP Proxy
Content-Length: 0
Warning: 392 213.151.XXX.XXX: Noisy feedback tells: pid=3770
req_src_ip=80.36.XXX.XXX req_src_port=27270
in_uri=sip:ad...@test.net;transport=tcp out_uri=sip:ad...@213.151.xxx.xxx:5070
via_cnt==1
U 2010/04/14 13:04:09.175387 213.151.XXX.XXX: - 79.149.232.74:1167
INVITE sip:m...@79.149.232.74:1167 SIP/2.0
Record-Route: sip:213.151.XXX.XXX:;lr=on
Call-ID: 107d1c4710e1e8fdf28aba6b22efe...@213.151.xxx.xxx
CSeq: 1 INVITE
From: miquel sip:miq...@test.net;tag=43317743
To: sip:m...@test.net
Via: SIP/2.0/UDP 213.151.XXX.XXX:;branch=z9hG4bK369d.3ceb9f8.0
Via: SIP/2.0/UDP 213.151.XXX.XXX:5070;rport=5070;branch=z9hG4bKQausL84Yz
Max-Forwards: 69
Accept-Contact: *;+g.poc.talkburst;require;explicit
User-Agent: PoC-serv/OMA1.0 Canard-1.0
Supported: 100rel,norefersub,timer
Referred-By: miquel sip:miq...@test.net
Contact:
sip:adhoc+1271243049...@213.151.xxx.xxx:5070;+g.poc.talkburst;session=1-1;isfocus
Allow: INVITE, SUBSCRIBE, BYE, ACK, CANCEL, REFER
Re: [SR-Users] syntax error with kamailio
alexis heron wrote:
Hi,
I have a problem to routing with kamailio. When I restart kamailio I
have this error message :
0(710) : core [cfg.y:3328]: parse error in config file
/usr/local/etc/kamailio/kamailio.cfg, line 640, column 2-3: syntax error
0(710) : core [cfg.y:3328]: parse error in config file
/usr/local/etc/kamailio/kamailio.cfg, line 640, column 2-3:
ERROR: bad config file (2 errors)
here is my line 640 :
if (uri=~sip:9[0-9][0...@.*) { //line 640
log(1, Matched Cisco Call Manager);
route(4);
};
Can you help me please
thank you
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thank you all for your help my SIPtrunk is mounted thanks to you. My
telephone routing is now operational.
thank you very much for your help
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Re: [SR-Users] Failure to load module db_mysql.so due to undefined symbol: log
On Wednesday 14 April 2010, Pratab Ali wrote: I read the reply from Daniel and as he suggested I got kamailio 3.0.0 using GIT this morning. However, I don't know if I actually got (Commit: 3a25f8327c on 3.0). I am unfamiliar with GIT as I use CVS. I will recheck to see if I got the branch you mention. Hi Pratab, just do a 'git log' in the top level of the checked out git tree. It should display the change from Daniel as the first (or now second) log entry. Cheers, Henning ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Failure to load module db_mysql.so due to undefined symbol: log
Pratab Ali wrote: Hi, Thanks, I just followed your git instructions and it shows that I have Daniel's change. I will try a clean rebuild and run again. Thanks. pratab If a clean build doesn't solve can you add to modules/db_mysql/Makefile a -lm (link the math library) to the LIBS variable (right after -lz) Can you test with this? Thanks Marius ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Failure to load module db_mysql.so due to undefined symbol: log
On Wednesday 14 April 2010, Pratab Ali wrote: Your suggestion with regards to appending -lm after -lz worked. I no longer get the undefined symbol: log error. Thanks! However, now I have kamailio unable to connect to mysql because I've not told it to use the correct port. This I can fix from reading the documentation. Hi Pratab, i think there could be a problem in the Makefile for the db_mysql module of sip-router. In Kamailio we used to autodetect the libraries. Can you please run the following command on your system in a shell and send me the output? The command is mysql_config --libs Cheers, Henning ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] reply when drop() in branch route
i call drop() in branch route and it is the only branch left. the branch gets correctly dropped, i.e., the request is not send out, but the reply to UAC is strange: U 2010/04/14 21:03:43.750712 192.98.102.10:5060 - 192.98.103.2:5074 SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/TM). Via: SIP/2.0/UDP 192.98.103.2:5074;rport=5074;branch=z9hG4bKaoxzewgj. To: sip:0407058...@192.98.102.10;tag=c02670ad1171fe45d9ff9a27d6c2cb82-e03b. From: sip:+35816234...@foo.bar;tag=bclsw. Call-ID: doibilvynrtq...@localhost. CSeq: 134 INVITE. Server: SIP Proxy (3.0.99-dev1 (i386/linux)). Content-Length: 0. why this kind of reply, because the proxy didn't even try to send the request to next ho? is it possible to somehow choose in the script, which reply to send? in this example, i'm using lcr to choose gws and none of them was suitable, so something like '503 service not available' would be more appropriate. -- juha ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] reply when drop() in branch route
2010/4/14 Juha Heinanen j...@tutpro.com: i call drop() in branch route and it is the only branch left. the branch gets correctly dropped, i.e., the request is not send out, but the reply to UAC is strange: U 2010/04/14 21:03:43.750712 192.98.102.10:5060 - 192.98.103.2:5074 SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/TM). Via: SIP/2.0/UDP 192.98.103.2:5074;rport=5074;branch=z9hG4bKaoxzewgj. To: sip:0407058...@192.98.102.10;tag=c02670ad1171fe45d9ff9a27d6c2cb82-e03b. From: sip:+35816234...@foo.bar;tag=bclsw. Call-ID: doibilvynrtq...@localhost. CSeq: 134 INVITE. Server: SIP Proxy (3.0.99-dev1 (i386/linux)). Content-Length: 0. why this kind of reply, because the proxy didn't even try to send the request to next ho? is it possible to somehow choose in the script, which reply to send? in this example, i'm using lcr to choose gws and none of them was suitable, so something like '503 service not available' would be more appropriate. What about checking in failure_route if the final response is locally generated (and also the winning status so you get the 477 and can act according)? -- Iñaki Baz Castillo i...@aliax.net ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
