[SR-Users] Preparing for the next major release
Hi! Daniel has set a code freeze date for the next major release and the Kamailio community needs to work together to prepare the release. That means you as well! There will be a lot of work in the coming weeks to merge code from SIP router into the Kamailio core to complete the merger process. We also have a large set of new modules and functions to test. The creation of a new release is much more than code - it's testing, documentation proof-reading and feedback. With a large and complex piece of software like Kamailio we can't test all functions with all possible combinations of configurations and connections to third party products - this is where YOU can help out! Read more on http://www.kamailio.org/w/2012/12/preparations-for-new-release/ and join us in creating the new release! /O ___ 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] Kamailio Load Balancing, problems ACK and BYE.
Hi everyone,
I am a beginner in using Kamailio. I am currently building a load balancer
for multiple SIP Servers, all of them are Kamailio. The Load balancer can
help end-users REGISTER and INVITE normally, but has problems with ACK and
BYE.
Messages 180 Ringing and 200 OK sent successfully to the Caller, but the ACK
failed, is transmitted to the SIP Server then stop here, doesn't go back to
Load balancing. (Caller's BYE -> LB -> Proxy then stop here). Because
doesn't receive ACK, the Called continuously sent 200 call OK to Caller
(sent successfully) and wait for confirmation ACK (after arrival to the SIP
Server, ACK doesn't come back to Load Balancing). Because the Called party
doesn't receive ACK, the call will be automatically disconnected after 30
seconds.
With BYE have rather, if the Caller hangs up first, BYE is transferred over
SIP Server and returned to Load balancer, but Load Balancing can't sent it
to the Called (Caller's BYE -> LB -> Proxy -> LB then do not go to the side
of called). Caller had picked off the phone, but the Caller is not received
BYE and after 30 seconds the call will automatically disconnect (by ACK
errors). If the called party hangs up first, HUNG UP button of the called
even useless, and I don't see any packet of the Called party's BYE message
in Wireshark.
I use X-Lite and EyeBeam for two end users, the configuration of the SIP
Server is the default configuration of Kamailio.
DISPATHCHER.LIST:
I have 3 Kamailio SIP Proxies, all of them OK, share one DB located in Proxy
2. But for capturing packets more convenient, I turn off the Proxy 2 and 3.
# group sip addresses of your * units
# Kamailio SIP Proxies/Registrars
1 sip:192.168.3.11:5060
#1 sip:192.168.3.12:5060
#1 sip:192.168.3.13:5060
Previously I used the Stateless Load Balancing configuration by default, but
could not INVITE:
# -- dispatcher params --
modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list")
modparam("dispatcher", "force_dst", 1)
route{
if ( !mf_process_maxfwd_header("10") )
{
sl_send_reply("483","To Many Hops");
drop();
};
ds_select_dst("1", "0");
record_route();
forward();
# t_relay();
}
Then I changed the following, sent INVITE and setup call successfully, but
met ACK and BYE error above.
if ( is_method("INVITE")) {
# Not Dispatch INVITE from SIP Proxies
if ( ds_is_from_list("1") ) {
t_relay();
} else if ( !ds_is_from_list("1") ) {
route(DISPATCH);
}
} else {
# If not INVITE
route(DISPATCH);
}
}
route[DISPATCH] {
# dispatch destinations
ds_select_dst("1", "0");
record_route();
t_relay();
#forward();
}
Please tell me How to modify in the configuration of Proxy and Load
Balancing?
Thank so much for helping!
Best Regard
Nguyen Anh Tuan,
The latest 200 OK from Called after Ringing : LB->Caller
Via: SIP/2.0/UDP
172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-a7349502ae55
cf46-1--d87543-;rport=9040
Record-Route:
Record-Route:
Contact:
To: "1004";tag=8e782d60
From: "1001 registrar.sip.vn";tag=7c72f24e
Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 435
Caller's ACK (1001@192.168.3.88) to Load Balancer (192.168.3.10):
Via: SIP/2.0/UDP
172.24.104.159:9040;branch=z9hG4bK-d87543-6f1c8659392a0773-1--d87543-;rport
Max-Forwards: 70
Route:
Route:
Contact:
To: "1004";tag=8e782d60
From: "1001 registrar.sip.vn";tag=7c72f24e
Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc.
CSeq: 2 ACK
Proxy-Authorization: Digest
username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P
32lS",uri="sip:1...@registrar.sip.vn",response="ec2a4c8832bc7bf7665811c26ec9
b292",algorithm=MD5
User-Agent: eyeBeam release 1003s stamp 31159
Content-Length: 0
ACK 192.168.3.10:5060 (Load Balance) to 192.168.3.11:5060 (Proxy) then
STOP HERE (Not transfer further)
Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP
172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-6f1c8659392a
0773-1--d87543-;rport=9040
Max-Forwards: 68
Contact:
To: "1004";tag=8e782d60
From: "1001 registrar.sip.vn";tag=7c72f24e
Call-
[SR-Users] Kamailio Load Balancing, problems ACK and BYE.
Hi everyone,
I am a beginner in using Kamailio. I am currently building a load balancer
for multiple SIP Servers, all of them are Kamailio. The Load balancer can
help end-users REGISTER and INVITE normally, but has problems with ACK and
BYE.
Messages 180 Ringing and 200 OK sent successfully to the Caller, but the ACK
failed, is transmitted to the SIP Server then stop here, doesn't go back to
Load balancing. (Caller's BYE -> LB -> Proxy then stop here). Because
doesn't receive ACK, the Called continuously sent 200 call OK to Caller
(sent successfully) and wait for confirmation ACK (after arrival to the SIP
Server, ACK doesn't come back to Load Balancing). Because the Called party
doesn't receive ACK, the call will be automatically disconnected after 30
seconds.
With BYE have rather, if the Caller hangs up first, BYE is transferred over
SIP Server and returned to Load balancer, but Load Balancing can't sent it
to the Called (Caller's BYE -> LB -> Proxy -> LB then do not go to the side
of called). Caller had picked off the phone, but the Caller is not received
BYE and after 30 seconds the call will automatically disconnect (by ACK
errors). If the called party hangs up first, HUNG UP button of the called
even useless, and I don't see any packet of the Called party's BYE message
in Wireshark.
I use X-Lite and EyeBeam for two end users, the configuration of the SIP
Server is the default configuration of Kamailio.
DISPATHCHER.LIST:
I have 3 Kamailio SIP Proxies, all of them OK, share one DB located in Proxy
2. But for capturing packets more convenient, I turn off the Proxy 2 and 3.
# group sip addresses of your * units
# Kamailio SIP Proxies/Registrars
1 sip:192.168.3.11:5060
#1 sip:192.168.3.12:5060
#1 sip:192.168.3.13:5060
Previously I used the Stateless Load Balancing configuration by default, but
could not INVITE:
# -- dispatcher params --
modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list")
modparam("dispatcher", "force_dst", 1)
route{
if ( !mf_process_maxfwd_header("10") )
{
sl_send_reply("483","To Many Hops");
drop();
};
ds_select_dst("1", "0");
record_route();
forward();
# t_relay();
}
Then I changed the following, sent INVITE and setup call successfully, but
met ACK and BYE error above.
if ( is_method("INVITE")) {
# Not Dispatch INVITE from SIP Proxies
if ( ds_is_from_list("1") ) {
t_relay();
} else if ( !ds_is_from_list("1") ) {
route(DISPATCH);
}
} else {
# If not INVITE
route(DISPATCH);
}
}
route[DISPATCH] {
# dispatch destinations
ds_select_dst("1", "0");
record_route();
t_relay();
#forward();
}
Please tell me How to modify in the configuration of Proxy and Load
Balancing?
Thank so much for helping!
Best Regard
Nguyen Anh Tuan,
The latest 200 OK from Called after Ringing : LB->Caller
Via: SIP/2.0/UDP
172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-a7349502ae55
cf46-1--d87543-;rport=9040
Record-Route:
Record-Route:
Contact:
To: "1004";tag=8e782d60
From: "1001 registrar.sip.vn";tag=7c72f24e
Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1102q stamp 51814
Content-Length: 435
Caller's ACK (1001@192.168.3.88) to Load Balancer (192.168.3.10):
Via: SIP/2.0/UDP
172.24.104.159:9040;branch=z9hG4bK-d87543-6f1c8659392a0773-1--d87543-;rport
Max-Forwards: 70
Route:
Route:
Contact:
To: "1004";tag=8e782d60
From: "1001 registrar.sip.vn";tag=7c72f24e
Call-ID: 1926bb7b58289f2fODg1OTZlZGE5MTAxN2JhOWJjNTZmOGQ1OTljOTJiNjc.
CSeq: 2 ACK
Proxy-Authorization: Digest
username="1001",realm="registrar.sip.vn",nonce="T4uyKE+LsPwWJBt3JseE7OVdd58P
32lS",uri="sip:1...@registrar.sip.vn",response="ec2a4c8832bc7bf7665811c26ec9
b292",algorithm=MD5
User-Agent: eyeBeam release 1003s stamp 31159
Content-Length: 0
ACK 192.168.3.10:5060 (Load Balance) to 192.168.3.11:5060 (Proxy) then
STOP HERE (Not transfer further)
Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP 192.168.3.10;branch=z9hG4bKcydzigwkX
Via: SIP/2.0/UDP
172.24.104.159:9040;received=192.168.3.88;branch=z9hG4bK-d87543-6f1c8659392a
0773-1--d87543-;rport=9040
Max-Forwards: 68
Contact:
To: "1004";tag=8e782d60
From: "1001 registrar.sip.vn";tag=7c72f24e
Call-
Re: [SR-Users] Internal/External SIP setup
Awesome. Thank you so much ;) On Wed, Dec 5, 2012 at 12:06 AM, Uriel Rozenbaum wrote: > The answer is YES. The setup you talk about requires a little work on the > routing domain, but with Kamailio and RTPProxy you'll be fine. > > On Tue, Dec 4, 2012 at 8:00 PM, Zero Aggression < > zeroaggress...@googlemail.com> wrote: > >> hanks so muc > > > > ___ > 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 > > ___ 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] Segmentation fault in sip-capture
something in tcp stack, maybe need initialize some modules before. But who cares ? :-) regards, Alexandr From: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Owen Lynch Sent: Tuesday, December 04, 2012 9:21 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: Re: [SR-Users] Segmentation fault in sip-capture Hi Alexandr, thanks, that seems to have fixed it. But I am curious, why should having TLS enabled cause kamailio to crash in this configuration? Owen On 3 December 2012 22:27, Alexandr Dubovikov wrote: Hi Owen, you don't need TLS for sip capturing, please undef WITH_TLS and try again, Wbr, Alexandr 2012/12/3 Owen Lynch On 3 December 2012 10:20, Alexandr Dubovikov wrote: Hi Olwen, siptrace use UDP to send HEP packets, but here is a problem in tcp stack. Can you please provide your config for the capture node and config for remote kamailio. Wbr, Alexandr From: sr-users-boun...@lists.sip-router.org [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Owen Lynch Sent: Sunday, December 02, 2012 9:43 PM To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users Mailing List Subject: [SR-Users] Segmentation fault in sip-capture Hi, I am running Kamailio 3.3 as a Homer capture agent, the sip-capture module receives HEP packets from a sip-trace module in a remote kamailio instance. Periodically the capture agent dies. The log and core dump suggest the initial problem is an invalid HEP msg received, but kamailio crashes when trying to send an error reply. Can anyone shed any light on this? Thanks, Owen Lynch Back trace: Program terminated with signal 11, Segmentation fault. #0 0x0813c916 in send2child (tcpconn=0xb55c1c60) at tcp_main.c:3967 3967tcp_main.c: No such file or directory. in tcp_main.c Missing separate debuginfos, use: debuginfo-install glibc-2.12-1.80.el6.i686 keyutils-libs-1.4-4.el6.i686 krb5-libs-1.9-33.el6.i686 libcom_err-1.41.12-12.el6.i686 libselinux-2.0.94-5.3.el6.i686 mysql-libs-5.1.61-4.el6.i686 nss-softokn-freebl-3.12.9-11.el6.i686 openssl-1.0.0-20.el6_2.5.i686 zlib-1.2.3-27.el6.i686 (gdb) bt #0 0x0813c916 in send2child (tcpconn=0xb55c1c60) at tcp_main.c:3967 #1 0x08136e78 in handle_tcpconn_ev (fm=, ev=, idx=) at tcp_main.c:4310 #2 handle_io (fm=, ev=, idx=) at tcp_main.c:4362 #3 0x0813ea3c in io_wait_loop_epoll () at io_wait.h:1092 #4 tcp_main_loop () at tcp_main.c:4656 #5 0x080a502b in main_loop () at main.c:1727 #6 0x080a6c00 in main (argc=11, argv=0xbfeafd14) at main.c:2546 Log: Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: : [pass_fd.c:288]: ERROR: receive_fd: recvmsg on 10 failed: Connection reset by peer Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: ERROR: [tcp_main.c:2347]: BUG: tcp_send: failed to get fd(receive_fd): Connection reset by peer (104) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: ERROR: sl [../../forward.h:193]: msg_send: ERROR: tcp_send failed Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: ERROR: *** cfgtrace: c=[/usr/local/etc/kamailio/kamailio.cfg] l=704 a=3 n=exit Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: [xavp.c:365]: destroying xavp list (nil) Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: [receive.c:293]: receive_msg: cleaning up Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: ERROR: sipcapture [sipcapture.c:682]: ERROR: sipcapture:hep_msg_received: not supported version or bad length: v:[10] l:[10] vs [40] Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: [parser/msg_parser.c:634]: SIP Reply (status): Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: [parser/msg_parser.c:636]: version: Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: [parser/msg_parser.c:638]: status: <500> Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: [parser/msg_parser.c:640]: reason: Nov 28 12:24:24 flowadmin /usr/local/sbin/kamailio[19568]: DEBUG: [parser/parse_via.c:1286]: Foun
[SR-Users] db_cluster configuration
Hello, i am configuring db_cluster module in kamailio 3.3
I have configured in roun-robin read between two MySQL servers but the load
is not distributed equally
My configuration is that:
modparam("db_cluster", "connection", "con1=>mysql://USER1:PASS1@1.1.1.1/DB")
modparam("db_cluster", "connection", "con2=>mysql://USER2:PASS2@2.2.2.2/DB")
modparam("db_cluster", "cluster", "cls1=>con1=9r9p;con2=9r9p")
And the auth module with:
modparam("auth_db", "db_url", "cluster://cls1")
But i have enabled the mysql logs and the major part of the querys are
targeted to the second server
Is there any cache by peer?
What am i doing wrong?
Thanks in advance.
___
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] Freezing for next major release
On 12/5/12 5:06 PM, Alex Balashov wrote: On 12/05/2012 11:05 AM, Daniel-Constantin Mierla wrote: It seems that many people are already using devel version to play with websockets, meaning that we are staying very well with testing. I would add to this only an emphasis on the importance of not forgetting to test everything else, amidst all the excitement and attention on websockets. :-) My note tried to underline that we seemed to get new people on board of testing/using devel branch, based on public forums feedback, which was not the same for the past releases. But I guess they use websockets module without the core :-), so that part still has to be done, of course... Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ 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] modules_s/eval - survey of usage
Hello, is anyone using the modules_s/eval module? In the context of merging modules_s/* to modules/*, this one lacks the docbook xml files for readme. It has a text readme file, from where I understand it does script operations using a stack and polish notation expressions. Probably it was developed before the expressions support in config file language, at this moment all its operations can be done directly by the interpreter. I am considering moving the module to obsolete/ folder. If anyone is using it and wants to keep it in modules/*, then he/she has to convert the readme text into docbook format (which is not complex at all, just takes a bit of time, probably like 60 min or even less). If there is no answer in one week about usage, will be sent to obsolete/, from where it can be rescued in case of need (just to be clear for everybody that code is not lost). Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ 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] Freezing for next major release
On 12/05/2012 11:05 AM, Daniel-Constantin Mierla wrote: It seems that many people are already using devel version to play with websockets, meaning that we are staying very well with testing. I would add to this only an emphasis on the importance of not forgetting to test everything else, amidst all the excitement and attention on websockets. :-) -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ ___ 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] Freezing for next major release
Hello, based on last irc devel meeting and feedback afterwards, most of the people seem to be fine releasing next major release a bit earlier, in order to: - make websockets support and the other new features available in a stable version - allow proper development time frame for other planned new features that wouldn't have been ready if the release would be 1-2 months later So, considering the winter holidays, I propose to freeze development on January 7, 2013, with full release about one month later. It seems that many people are already using devel version to play with websockets, meaning that we are staying very well with testing. Other opinions? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ 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] [sr-dev] dns resolver issue (RFC3263)
Hi Daniel I wrote in my first patch announcing email, that i didn't test the patched dns resolution without cache. I only tested with dns cache. This is the reason why i didn't recognize this problem. You are right I made a mistake, but now it is corrected. Many Thanks, Misi On 2012-12-04 17:47, Daniel-Constantin Mierla wrote: Hello, I was looking to the patch and I spotted that you didn't assign anymore a value to he variable -- next is the specific part of the diff: - /* fallback to normal srv lookup */ - he=srv_sip_resolvehost(name, 0, port, proto, 0, 0); + /* fallback to srv lookup */ + no_naptr_srv_sip_resolvehost(name,port,proto); Shouldn't be like: he = no_naptr_srv_sip_resolvehost(name,port,proto); Cheers, Daniel On 11/30/12 10:31 AM, MÉSZÁROS Mihály wrote: Hi, On 2012-11-30 09:07, Daniel-Constantin Mierla wrote: Hello, On 11/19/12 10:18 AM, MÉSZÁROS Mihály wrote: Hi Daniel, On 2012-11-14 12:51, Daniel-Constantin Mierla wrote: Hello, On 11/12/12 10:50 AM, MÉSZÁROS Mihály wrote: Hi, I made some progress. As I stated before, I made a patch and submitted to git branch misi/dns_srv. I tested with dns cache. It works for me. I made it also available for case if "no dns cache" is used too, but it isn't tested yet. Please review my commit, and let me know if any corrections needed. if nobody does it meanwhile, I can look over it next week and also check properly what's all about this discussion, currently being out of the office. After you had time to review it, please let me know your thoughts. unfortunately I had no time to look at it yet. Hopefully I will find some soon. Btw, is it complete? IIRC, I saw something like it still has to be extended. It is complete and working patch. If there are no NAPTR records to a domain, then according to the local protocol preference it orders protocols and it tries to resolve SRV records according this ordered list. If there is no order then the order is udp,tcp,tls,sctp,.. SRV records are resolved in order Kamailio dns protocol preference. My algorithm picks and returns with the first protocol resolvable SRV record, so it sets from SRV the port and protocol. (Of course if there are no SRV at all then it fallbacks to host resolving so dns "A" record.) It is big step forward comparing to current Kamailio behavior where it is using strictly udp only and after it stops searching SRV records at all, and go for "A" record! As i wrote in my patch announcing email it is a step further on the way to conforming with RFC3263, but my patch not handling fallback if there are SRV-s for multiple protocols in DNS. In such case only and only if the first protocol is temporary not available or fails we are not falling back to other protocol but falling back to host resolving so "A" record (and/or ). Can you send meg the iirc message what was there exactly? Is there any other problem in it? I guess no just what i explained above. I am eagerly waiting your review and comment. Thanks in advance! Misi -- Daniel-Constantin Mierla -http://www.asipto.com http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda ___ 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] syslog message about invalid domain name
Daniel-Constantin Mierla writes: > looking at the code history, you added this check in the commit > 019ab5e2d6730b764b20a890f9a3b5f9237b6338 . > > It is about checking if it is a valid domain, to avoid doing queries on > invalid values. In this particular case is probably related to presence > of characters [, ], and :. ok, thanks for reminding me. i'll change the code so that the info message is not printed. -- 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] syslog message about invalid domain name
Hello, looking at the code history, you added this check in the commit 019ab5e2d6730b764b20a890f9a3b5f9237b6338 . It is about checking if it is a valid domain, to avoid doing queries on invalid values. In this particular case is probably related to presence of characters [, ], and :. Cheers, Daniel On 12/5/12 9:04 AM, Juha Heinanen wrote: when i t_relay in-dialog bye: Request-Line: BYE sip:0x8eb94d0@[2002:c062:660a::1]:5050;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP [2002:C062:660A:0:0:0:0:1];branch=z9hG4bKde53.6aaed77ea908953bbb133e5021d74036.0 Via: SIP/2.0/UDP [::1]:5090;received=0:0:0:0:0:0:0:1;branch=z9hG4bKq~ouJaGs;rport=5090 From: ;tag=5A29C828-50BEFE55000C7118-B4FF9B70 To: "" ;tag=92e26ffbfbc16e53 CSeq: 10 BYE Call-ID: 28d1e4872cc5f83e User-Agent: Sip Express Media Server (1.6.0 (x86/linux)) Max-Forwards: 69 Content-Length: 0 with $du set to sip:[2002:C062:660A:0:0:0:0:1]:59933;transport=tcp, i get to syslog: Dec 5 09:53:48 sip /usr/sbin/sip-proxy[1947]: INFO: [resolve.c:729]: invalid domain name '[2002:C062:660A:0:0:0:0:1]' since everything is and works as expected, why is the info message produced and how can i get rid of it? -- 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 -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ___ 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] syslog message about invalid domain name
when i t_relay in-dialog bye: Request-Line: BYE sip:0x8eb94d0@[2002:c062:660a::1]:5050;transport=tcp SIP/2.0 Message Header Via: SIP/2.0/TCP [2002:C062:660A:0:0:0:0:1];branch=z9hG4bKde53.6aaed77ea908953bbb133e5021d74036.0 Via: SIP/2.0/UDP [::1]:5090;received=0:0:0:0:0:0:0:1;branch=z9hG4bKq~ouJaGs;rport=5090 From: ;tag=5A29C828-50BEFE55000C7118-B4FF9B70 To: "" ;tag=92e26ffbfbc16e53 CSeq: 10 BYE Call-ID: 28d1e4872cc5f83e User-Agent: Sip Express Media Server (1.6.0 (x86/linux)) Max-Forwards: 69 Content-Length: 0 with $du set to sip:[2002:C062:660A:0:0:0:0:1]:59933;transport=tcp, i get to syslog: Dec 5 09:53:48 sip /usr/sbin/sip-proxy[1947]: INFO: [resolve.c:729]: invalid domain name '[2002:C062:660A:0:0:0:0:1]' since everything is and works as expected, why is the info message produced and how can i get rid of it? -- 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
