sr-users

sr-users@lists.kamailio.org

42 participants
37519 discussions

Setup dmq_usrloc
by Gertjan Wolzak 17 Dec '18

17 Dec '18

Hello Kamailions, Im trying to setup registration replication with the use of the dmq and dmq_usrlog modules. I have configured two servers, configured kamailio on both. When I register a sip client on the one I get a nice KDMQ message on the second server, but the second server reacts with a 404 not found: U 2018/12/17 05:08:38.974495 10.10.0.1:6000 -> 10.10.0.2:6000 #3 KDMQ sip:usrloc@10.10.0.2:6000 SIP/2.0. Via: SIP/2.0/UDP 10.10.0.1:6000;branch=z9hG4bK00ed.e1666b85000000000000000000000000.0. To: <sip:usrloc@10.10.0.2:6000>. From: <sip:usrloc@10.10.0.1:6000>;tag=a2652bb825a097f3b7285d4c70edd51a-0513. CSeq: 10 KDMQ. Call-ID: 35571d0247178282-3780(a)10.88.77.172. Content-Length: 360. User-Agent: kamailio (5.1.6 (x86_64/linux)). Max-Forwards: 1. Content-Type: application/json. . {"action":2,"aor":"test01(a)10.88.77.172","ruid":"uloc-5c177363-ec4-1","c":"sip:test01@10.88.77.140:59293;ob","received":"","path":"","callid":"zjucvmzUT2hNJCIfqeUXEsOaEzyHCs-4","user_agent":"CSipSimple_G3121-26/r2457","instance":"","expires":1545041786,"cseq":24435,"flags":0,"cflags":0,"q":-1,"last_modified":1545040886,"methods":8159,"reg_id":0,"server_id":0} U 2018/12/17 05:08:38.979062 10.10.0.2:6000 -> 10.10.0.1:6000 #4 SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 10.10.0.1:6000;branch=z9hG4bK00ed.e1666b85000000000000000000000000.0;rport=6000. To: <sip:usrloc@10.10.0.2:6000>;tag=02f7a7b91eadf5e07797231cb2f64326-b794. From: <sip:usrloc@10.10.0.1:6000>;tag=a2652bb825a097f3b7285d4c70edd51a-0513. CSeq: 10 KDMQ. Call-ID: 35571d0247178282-3780(a)10.88.77.172. Server: kamailio (5.1.6 (x86_64/linux)). Content-Length: 0. . I think I have to configure a bit more than just the parameters of the dmq and dqm_usrloc modules. Ive googled, looked in the Sip routing with kamailio book, but no luck. Did not see any examples in the kamailio source, with regards to the dmq_usrloc module. I am not looking for direct answers, have not provided you with enough information for that, but I would appreciate information on how to correctly configure registration replication using the dmq/dmq_usrloc module... Thank you. With regards, Gertjan

1 0

Reminding again about private/direct emails from mailing lists discussions
by Daniel-Constantin Mierla 17 Dec '18

17 Dec '18

I see that still a lot of emails are sent directly to me after a discussion started on public mailing lists (subject with mailing lists prefix, but the sender header is not the mailing list daemon), I would like to remind that, at least in my case, writing emails directly without cc-ing the mailing list is highly not recommended and there is practically 0 chance to get an answer from myself. The rule is also suggested in the mailing lists presentation page: - https://www.kamailio.org/w/mailing-lists/ Unless I asked explicitly for a private email with some specific details, the message will not be noticed in decent time and therefore not replied. I use this email address for many public mailing lists and it is intended only for that usage, I am reacting based on the filters I created, unmatched messages getting the lowest priority. Therefore if you wrote me and expect a response, re-send the message to the appropriate public mailing list. Checking and answering the messages on Kamailio mailing lists are among the top of priorities on my daily activity, so it is high chance you get the response faster when you write there. Note that you can always mask sensitive data (e.g., replace passwords, IP addresses) before sending logs or config file. There are several common sense as well as technical reasons for keeping the discussion on public mailing lists. Here are some along with other suggestions to improve the likeliness of getting an answer: - we are not answering questions on mailing lists to help only one person, but also other people that may have similar issues in the future -- they can find the answers in the archive with a search engine. Also, other people subscribed to the list can learn from conversation or contribute to the discussion, resulting in better outcome - the amount of unexpected messages is very high -- it is practically a zero chance to get to the folder with messages that didn't match any filter (thus your message is not going to be replied) - whenever I find time, the first emails I answer are those coming on mailing lists. They are many there as well, people are traveling or having other personal or business projects, so sometime is good to send a reminder if a question does not get an answer - starting with negative approach or no technical content, like you are going to use something else if you don't get a reply instantly, or this application is missing what so ever tutorial giving exactly the solution you need, or simply asking for full config of complex requirements, is not helping at all, but actually making people to ignore your messages. Start with what you tried and where you got stuck. Everything you get is for free, have appropriate expectation and behavior. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

P-CSCF without Rx interface
by Tsvetan Filev 17 Dec '18

17 Dec '18

Hi All. May be this has been asked before but I was not able to find an answer. I setup IMS with 3 instances of kamailio: P-CSCF, I-CSCF and S-CSCF (I got the config files from here https://github.com/kamailio/kamailio/tree/master/misc/examples/ims). I would like to register two clients to the IMS and make a VoLTE like call. So I want to setup my P-CSCF to send SIP messages to my I-CSCF without Rx interface (without using PCRF). I'm not able to find how to tell P-CSCF where to direct the sip traffic. It obviously should happen in route[REGISTER]. Right now P-CSCF sends all REGISTER requests to itself and goes to a loop. Regards.

3 3

Sending to an announcement server from a failure_route
by Boden Garman 17 Dec '18

17 Dec '18

Hi list, I'm trying to play an announcement for my users when my PSTN gateway responds with a 404 Not Found (Reason: Q.850;cause=3), indicating a wrong number. I'm a bit lost and would appreciate any advice. What I am seeing is two parallel SIP invites, one to my announcement server (hooray!) and one to the PSTN gateway again (which I don't want). How do I stop the original branch from starting again? I've tried changing the order of rewritehostport and append_branch without any difference. I've also tried to use t_relay() instead of the RELAY route. I'm using Kamailio 5.1 and have modified the deafult kamailio.cfg Thanks! Boden I think this is the relevant parts of the config: # PSTN GW routing route[PSTN] { [PSTN regex logic omitted] # classify as an outbound call set_dlg_profile("outbound_calls"); # select a PSTN gateway from the dispatcher if (!ds_select_dst("66", "8")) { send_reply("404", "No destination"); exit; } # failover if the selected PSTN gateway is down t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Try next destinations in failure route failure_route[RTF_DISPATCH] { xlog("inside RTF_DISPATCH"); if (t_is_canceled()) { exit; } # wrong number from PSTN if (t_check_status("404")) { append_branch(); append_hf("X-Application: announcement\r\n"); append_hf("X-Announcement-Playback: wrong-number\r\n"); route(TO_ANNOUNCEMENT); exit; } # next DST - only for 500 or local timeout if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) { xlog("selecting next destination"); if (ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } # Routing to announcement server route[TO_ANNOUNCEMENT] { if (!is_method("INVITE")) { return; } rewritehostport('ANNOUNCEMENT_SERVER:5060'); route(RELAY); exit; }

2 3

Problem with selection Transport Protocol - MultiHomed Problem
by YAS0 CANER 17 Dec '18

17 Dec '18

Hello , i have trouble about selection transport protocol. i have tried so many thing but couldnt find TM selected which IPv? and protocol type (TCP or UDP) For example; i have 2 interface ipv6 and ipv4. i will route my message (INVITE-Register etc.) another node that use TCP and IPV6 , domain name is test1.test. TM module or some module resolve this domain to IPV6:UDP but i want to sent it over IPV6:TCP socket. so i cannot know which protocol is resolved and cannot route the right socket. $du variable show only sip:ali@test1.test Conclusion : There isn't any pseudo value about which destination of IP version will use. Thanks Flow: UAC (IPv4:UDP) ----------> Kamailio (MultiHomed) ----> (TCP:IPv6)Kamailio ---- Kamailo (multiHomed ) cfg ------ auto_bind_ipv6=1 dns_try_ipv6=yes dns_retr_time=1 dns_retr_no=1 dns_cache_flags=4 mhomed=1 fork=yes tcp_source_ipv6 = [fd61:c23c:b3b6::5] listen=tcp:[fd61:c23c:b3b2::5]:5060 listen=udp:[fd61:c23c:b3b2::5]:5060 listen=udp:192.168.1.107:5060 listen=tcp:192.168.1.107:5060 route []{ if($dP=="TCP"){ xlog("L_INFO","[route] [TCP] 1\n"); if (!t_relay_to_tcp()) { xlog("L_ERR", "Error t_relay [route] \n"); t_reply("500","Server - ERROR "); exit; } } else if($dP=="TCP"){ xlog("L_INFO","[route] [TCP] 2\n"); if (!t_relay_to_tcp()) { xlog("L_ERR", "Error t_relay [route] \n"); t_reply("500","Server - ERROR "); exit; } }else if($dP=="UDP"){ xlog("L_INFO","[route] [UDP] 1\n"); if (!t_relay_to_udp()) { xlog("L_ERR", "Error t_relay [route] \n"); t_reply("500","Server - ERROR "); exit; } }else{ xlog("L_INFO","[route] [??] 1\n"); if (!t_relay()) { xlog("L_ERR", "Error t_relay [route] \n"); t_reply("500","Server - ERROR "); exit; } } }

2 5

Re: [SR-Users] sr-users Digest, Vol 163, Issue 15
by Toufic Mobarak 17 Dec '18

17 Dec '18

Tnhxtx Regards, Toufic Mobarak > On Dec 15, 2018, at 6:00 AM, sr-users-request(a)lists.kamailio.org wrote: > > Send sr-users mailing list submissions to > sr-users(a)lists.kamailio.org > > To subscribe or unsubscribe via the World Wide Web, visit > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.kamailio.org > > You can reach the person managing the list at > sr-users-owner(a)lists.kamailio.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: Routing of Audio and Video Stream. (Daniel-Constantin Mierla) > 2. Problem with selection Transport Protocol - MultiHomed > Problem (YAS0 CANER) > 3. Re: Problem with selection Transport Protocol - MultiHomed > Problem (YAS0 CANER) > 4. Re: Routing of Audio and Video Stream. (Sergiu Pojoga) > 5. Re: Problem with selection Transport Protocol - MultiHomed > Problem (Daniel-Constantin Mierla) > 6. Issue on kamailio 4.1.9 of database reconnection (Emmanuel BUU) > 7. Re: Routing of Audio and Video Stream. (Abhijit Dutta) > 8. Re: Routing of Audio and Video Stream. (Abhijit Dutta) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Fri, 14 Dec 2018 12:54:44 +0100 > From: Daniel-Constantin Mierla <miconda(a)gmail.com> > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.kamailio.org>, Abhijit Dutta <duttaa(a)hotmail.com> > Subject: Re: [SR-Users] Routing of Audio and Video Stream. > Message-ID: <26c51f12-5242-4794-5124-f37bc523c7ae(a)gmail.com> > Content-Type: text/plain; charset="windows-1252" > > Hello, > > > I guess you still need to send the INVITE to Asterisk, right? > > > Cheers, > Daniel > > >> On 14.12.18 11:15, Abhijit Dutta wrote: >> Hi All, >> >> Is it possible to redirect only the audio stream towards Asterisk from >> Kamailio (Proxy) via RTPEngine and not to send the video stream to >> Asterisk. I am yet to get the complete reason why my client wants this >> to happen though. >> >> Thanks >> Abhijit >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla -- www.asipto.com > www.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com > Kamailio Advanced Training -- www.asipto.com > >

2 1

Automating installation
by John Tuxies 14 Dec '18

14 Dec '18

Hi all. I am new to this and i would like some help please. I am looking to deploy Kamailio and Asterisk in my systems and i would like to automate the process. I have found a bash script but it it outdated from here: https://github.com/gurumelo/asteriskkamailio i did some changes and modifications to adapt it to newer version of software, but it fails to start Kamailio and continue to Asterisk. I would appreciate if someone could help me with this script and make it work. The systems i am planning to execute it is in Debian 9 and Ubuntu 18.04 server, both 64bit. Sincerely yous, John

8 21

FreeBSD Kamailio pkg
by Sergiu Pojoga 14 Dec '18

14 Dec '18

Did anyone notice that when installing Kamailio from FreeBSD package, by default, it accepts registration with whatever account credentials? Not sure if it's like that with other OS packages (usually I install it from source), but this behavior kind of doesn't go with the popular saying that Kamailio by default "does nothing", lol. Regards, --Sergiu

3 4

Kamailio ERROR: _reply_light(): can't generate xxx reply when a final yyy was sent out
by Olli Attila 13 Dec '18

13 Dec '18

Hello, I'm getting these error messages on kamailio log from time to time: Dec 11 19:18:30 /usr/sbin/kamailio[23913]: ERROR: tm [t_reply.c:482]: _reply_light(): can't generate 487 reply when a final 200 was sent out Dec 8 04:44:56 /usr/sbin/kamailio[14960]: ERROR: tm [t_reply.c:482]: _reply_light(): can't generate 487 reply when a final 486 was sent out Dec 5 07:46:04 /usr/sbin/kamailio[14961]: ERROR: tm [t_reply.c:482]: _reply_light(): can't generate 487 reply when a final 480 was sent out Kamailio is configured to act as a proxy between SBC and a softswitch. According to the signaling we can see that these errors occure when the SBC sends CANCEL message towards Kamailio when at the same time final reply is received from the Softswitch side. The call seems to be teared down nicely and finally Kamailio replies to SBC by sending "200 ok, no more pending brances". I guess Kamailio treats this situation as an error because the sip processing engine has sent the final reply out and at the same time SBC sends the CANCEL to Kamailio. Is there any way we could handle this situation better on Kamailio end? The three example ERROR messages are from cases where B subscriber has answered the call, the B has been busy and the last one is from a case where B subscriber has been temporarily unavailable. Cheers, Olli

2 4

is_e164 logic to detect valid number
by Patrick Wakano 11 Dec '18

11 Dec '18

Hello list, Hope you are all doing fine! I am using the is_e164() function to validate the number we receive, and I come to see that the number +555 was accepted.... After some googling it looks like(it is not very clear though) that 7 digits are the minimum we could have for e164 numbers but after checking the source code, I saw it accepts anything starting with + and having between 2 and 16 numbers. So is it really valid to have a number with just 2 digits? What is the case? Thank you, Kind regards, Patrick Wakano

5 7

Jump to page:

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users