sr-users February 2018

sr-users@lists.kamailio.org

90 participants
112 discussions

ways to track all messages of a dialog
by Fabian Borot 09 Mar '18

09 Mar '18

I need to track all messages (SIP Requests, SIP Responses) for some SIP dialogs. Let's say I need to log something specific on each state for calls initiated from a certain IP (initial request, temporary replies, rejection with a negative response, call was accepted with 200, termination with BYE, termination with CANCEL etc. ). I am thinking on using flags but I am not sure after reading the tutorial on flags (http://www.kamailio.org/dokuwiki/doku.php/tutorials:openser-flag-operations) if that is going to work properly. The doc says: "They provide a very easy and fast way of keeping states during processing a request or during a transaction". I am afraid that the ACK if the call is accepted and the BYE won't be tracked properly because they are not part of the initial transaction. Also the doc does not mention if the flag is preserved among branches spawned during the processing of the call. The idea is for example: request_route { if (is_method("INVITE")){ if ($si == "192.168.168.100"){ #condition that would set the flag, INVITE came from "my_IP" #flag/mark dialog setflag(1); #log call from "my_IP" was received } route("INCOMING); } If(loose_route() ){ if (is_method("ACK")){ if(isflagset(1)){ #log ACK related to "my_IP" was received } } if (is_method("CANCEL")){ if(isflagset(1)){ #log CANCEL related to "my_IP" was received } } if (is_method("BYE")){ if(isflagset(1)){ #log BYE related to "my_IP" was received } } } } Then : route["INCOMING"] { # do some processing # do some processing t_onreply("INCOMING"); t_on_failure("INCOMING"); t_relay(); } onreply_route[INCOMING] { #do some processing if(isflagset(1)){ #log a reply related to "my_IP" was received } } failure_route[INCOMING] { #do some processing if(isflagset(1)){ #log a negative reply related to "my_IP" was received } } Will the flags work on this case ? My 2nd approach is to use a htable and use the call_id for my calls as the key. The on each route section (reply, failure_reply, ACK processing etc.) find the $ci on the htable and if it is there do the appropriate action. But I need to delete the key:value once the call ends or leave it autoexpire with a expire value bigger than the max_call_duration_time that we have in our system.

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Resend Authenticate REGISTER to two or more Asterisk PBX
by Social Boh 08 Mar '18

08 Mar '18

Hello, I'd like know which is the best way to resend a Authenticate REGISTER to two or more Asterisk. Repeating rewritehostport and forward commands for the number of Asterisk Servers, there is a problem with the Via Header; Kamailio add one with his IP for each forward. I try to remove the first Via before send the REGISTER to the second Asterisk without success. Any hint or tip? Thank you Regards -- --- I'm SoCIaL, MayBe

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Kamailio Memory Settings
by Duarte Rocha 08 Mar '18

08 Mar '18

Hey. I have some questions about memory settings. I can see the default values with "kamailio -I" and change the current values with "kamailio -M 12 -m 128". How do i see the active settings and how do i change the default value? I want to setup a proxy capable to handle 150CPS, 6k active calls, with Dispatcher, Database, Number Normalizations and other operations. What are good settings for CPU, RAM Memory, PKG_SIZE and SHM_SIZE? Thanks for the help, Best Regards, Duarte Rocha

2 1

Failed auth
by Aidar Kamalov 08 Mar '18

08 Mar '18

Hello! I have kamailio 5.0.5 and some cisco spa 50x phones. Sometime phone can't register to kamailio. REGISTER packets with right nonce,reponse, etc gets to kamailio but it response 401 Unauthorized with code -3 (user not found). After some attempts(or time, phone send register every 1200 seconds) or after reboot phone it register succesfully Feb 13 11:20:02 sipchel /usr/sbin/kamailio[110432]: NOTICE: <script>: AUTH: REGISTER [a470211a-b7d8faa5(a)172.16.3.1] Feb 13 11:20:02 sipchel /usr/sbin/kamailio[110432]: NOTICE: <script>: AUTH: REGISTER FAILED (code: -5) fd=voip.xxxx.org, adu=<null>, aa=<null>, ar=<null>, au=<null>, ad=<null>, aU=<null>, Au=33855(a)voip.xxxx.org [ a470211a-b7d8faa5(a)172.16.3.1] Feb 13 11:20:02 sipchel /usr/sbin/kamailio[110433]: NOTICE: <script>: AUTH: REGISTER [a470211a-b7d8faa5(a)172.16.3.1] Feb 13 11:20:02 sipchel /usr/sbin/kamailio[110433]: NOTICE: <script>: AUTH: REGISTER FAILED (code: -3) fd=voip.xxxx.org, adu=sip:voip.xxxx.org, aa=MD5, ar=voip.xxxx.org, au=33855, ad=, aU=33855, Au=33855(a)voip.xxxx.org [ a470211a-b7d8faa5(a)172.16.3.1] Feb 13 11:20:02 sipchel /usr/sbin/kamailio[110431]: NOTICE: <script>: AUTH: REGISTER [a470211a-b7d8faa5(a)172.16.3.1] Feb 13 11:20:02 sipchel /usr/sbin/kamailio[110431]: NOTICE: <script>: AUTH: REGISTER FAILED (code: -3) fd=voip.xxxx.org, adu=sip:voip.xxxx.org, aa=MD5, ar=voip.xxxx.org, au=33855, ad=, aU=33855, Au=33855(a)voip.xxxx.org [ a470211a-b7d8faa5(a)172.16.3.1] -- Aydar A. Kamalov

2 1

Issue starting with kamctl
by Ruben Lezama 08 Mar '18

08 Mar '18

Hi, I am having issues to start Kamailio using kamctl. root@kamailio2:/etc/systemd/system# kamctl restart INFO: Stopping Kamailio : ERROR: No PID file found (/var/run/kamailio.pid)! Kamailio probably not running INFO: check with 'ps axw | /bin/egrep kamailio' Using #Kamailio starts works. I can register peers and make calls between them. But I cannot update configuration. root@kamailio2:/var/log# kamailio Listening on udp: 127.0.0.1:5060 udp: 192.168.1.100:5060 tcp: 127.0.0.1:5060 tcp: 192.168.1.100:5060 Aliases: tcp: kamailio2:5060 tcp: localhost:5060 udp: kamailio2:5060 udp: localhost:5060 I would appreciate any help provided. Attached the syslog output Ruben

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KeepAlive Module
by Social Boh 08 Mar '18

08 Mar '18

Hello, I'm using keepalive module with drouting module... the event route works fine: NOTICE: <script>: Gateway up: OPTIONS sip:1.2.3.4 NOTICE: <script>: Gateway up: OPTIONS sip:5.6.7.8 but kamcmd command not: *kamcmd keepalive.list* foobar 42 foobar 42 { text: foobar number: 42 } { list: { nn: 17 nn: 22 } } { uri: sip:1.2.3.4 owner: drouting last checked: Wed Feb 21 11:20:12 2018 *last up: Wed Dec 31 19:00:00 1969* *last down: Wed Feb 21 11:20:13 2018* } { uri: sip:5.6.7.8 owner: drouting last checked: Wed Feb 21 11:20:12 2018 *last up: Wed Dec 31 19:00:00 1969* *last down: Wed Feb 21 11:20:12 2018* Thank you Regards... -- --- I'm SoCIaL, MayBe

2 1

Strongswan on same Server in EC2 as kamailio 4.4
by Jonathan Hunter 08 Mar '18

08 Mar '18

Hi Guys, I have an issue with sending a BYE message back down a strongswan VPN tunnel on a server in the Amazon EC2 environment. To get the configuration working correctly I have the Public IP address bound to a loopback virtual interface, and I also have kamailio listening to it. Calls establish without issue, but when the called party send a BYE, I cant seem to force the SIP message down the correct interface. Due to client confidentiality I need to be careful what information I send but does anyone have a similar configuration set up, and working, I am just looking for reasons as to why force_socket and mhomed=1 doesnt appear to force the BYE message via the Public Interface but the private IP address on the server as its a NAT'd environment. So essentially call establishes; CLIENT---------(VIA IPSEC)-INVITE----->(UDP)KAMAILIO PUBLIC IP-------->(UDP)KAMAILIO PRIVATE IP(TCP)------------------------->CLIENT B | | | | |<---------(VIA IPSEC)-100 TRYING----|<----------100 TRYING---------------|<----------100 TRYING----------------------------| | | | | |<---------(VIA IPSEC)-180 RINGING---|<----------180 RINGING--------------|<----------180 RINGING---------------------------| | | | | |<---------(VIA IPSEC)-200OK --------|<----------200OK -------------------|<----------200OK --------------------------------| | | | | |----------(VIA IPSEC)-ACk---------> |-----------ACk--------------------->|-----------ACk---------------------------------->| How can I ensure the BYE is forced out via the PUBLIC IP? So I do this; CLIENT---------(VIA IPSEC)-INVITE----->(UDP)KAMAILIO PUBLIC IP-------->(UDP)KAMAILIO PRIVATE IP(TCP)------------------------->CLIENT B |<---------BYE (VIA IPSEC)----------------------|<----------BYE ----------------------------|<----------BYE -------------------| As it appears to be going from Private IP and not being pushed down the IPSEC tunnel CLIENT---------(VIA IPSEC)-INVITE----->KAMAILIO PUBLIC IP-------->KAMAILIO PRIVATE IP------------------------->CLIENT B |<---------BYE ----------------------<----------BYE ----------------------------|<----------BYE -------------------| on the BYE I have tried; force_send_socket(KAMAILIO PUBLIC IP); However doesnt appear to work, is this an issue for ip routing on the server to push the BYE correctly down the IPSEC tunnel? or is it configuration related? Any comments welcome if people have had this issue before. Many thanks Jon

2 1

Planing next Kamailio releases - v4.4.7 v5.0.6 and v5.1.2
by Daniel-Constantin Mierla 01 Mar '18

01 Mar '18

Hello, I am considering to do releases from last three stable branches during next week. v4.4.7 is going to be released out of latest code in branch 4.4 to mark the end of official maintenance for series v4.4.x. The version 4.4.0 was released in March 2016, so it had more or less two years of maintenance. Commits can still go in 4.4 branch if developers wants to push there, end of official maintenance means that there is no plan to do packaging and release another version from the branch in the future, so you will have to build it from sources or generate packages yourself. v5.0.6 and v5.1.2 are the usual minor releases we do from the last two stable branches. If there are any issues you are aware of and not reported on bug tracker, create the items as soon as possible to get a better chance to review and eventually fix. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 3

Siremis and kamailio 5.x.x
by Olli Attila 01 Mar '18

01 Mar '18

Hello, Question about Kamailio web GUI (Siremis): I believe the Siremis updates are still a ongoing work or has there been any updates when the new Siremis update (compatible with Kamailio 5.x.x) will be released? https://www.kamailio.org/w/2017/07/siremis-ongoing-updates/ Cheers, Olli

2 1

Kamailio ims with TCP
by Amar Tinawi 01 Mar '18

01 Mar '18

Hello list For some reason i couldn't register on kamailio ims when i use only TCP protocols between nodes, the error was solved when i enabled UDP between I-CSCF and S-CSCF is it mandatory for those two nodes to talk over UDP only ? or i'm missing something ? and in general for sip protocol which is better ? UDP or TCP ? Regards,

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sr-users February 2018