sr-users January 2024

sr-users@lists.kamailio.org

49 participants
59 discussions

by mtohidi67＠gmail.com

Hello. I have installed Kamailio version devel on Ubuntu 22.04 and use the mysql database on the same machine. when i run systemctl status kamailio.service ● kamailio.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio; generated) Active: active (exited) since Sat 2024-01-20 11:35:09 +0330; 21min ago Docs: man:systemd-sysv-generator(8) CPU: 48ms /usr/local/sbin/kamailio[173481]: ERROR: usrloc [dlist.c:846]: register_udomain(): failed to open database connection /usr/local/sbin/kamailio[173481]: ERROR: registrar [registrar.c:767]: domain_fixup(): failed to register domain /usr/local/sbin/kamailio[173481]: ERROR: [core/route.c:1193]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:717 /usr/local/sbin/kamailio[173481]: ERROR: [core/rvalue.c:3818]: fix_rval_expr(): failure in cfg at line: 717 col: 22 /usr/local/sbin/kamailio[173481]: ERROR: [core/rvalue.c:3818]: fix_rval_expr(): failure in cfg at line: 717 col: 22 /usr/local/sbin/kamailio[173481]: ERROR: [core/route.c:1193]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:720 /usr/local/sbin/kamailio[173481]: INFO: [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized kamailio[173463]: * already running kamailio[173463]: ...done. systemd[1]: Started LSB: Start the Kamailio SIP proxy server. In /usr/local/etc/kamailio/kamctlrc, I uncomment these lines: SIP_DOMAIN=ims.mnc55.mcc999.farazNET.org DBENGINE=MYSQL DBHOST=localhost in /usr/local/etc/kamailio/kamailio.cfg i apply this changes: auto_aliases=no alias="ims.mnc55.mcc999.farazNET.org" listen=udp:192.168.75.185:5060 listen=tcp:192.168.75.185:5060 how can i solved it?

3 months, 2 weeks

rr, loose_route and corex alias_subdomains

by Jon Bonilla (Manwe)

Hi I'm having an issue with the double rr for mst direct routing. I'm using subdomains like 23234.mydomain.com with a wildcard certificate and I set corex modparam alias_subdomains to mydomain.com But loose_route doesn't seem to detect the rr as myself I get the message: rr [loose.c:804]: rr_do_force_send_socket(): no socket found to match second RR (sip:20141.mydomain.com:5063;transport=tls;ftag=as7107d0fe;lr;r2=on;vsf=AAAAAAAAAAADCQMaABhMS1NYRh0cRllbR0RRQG91cC5lcw--;nat=yes;pet=mst) Regarding my setup is more or less like this: kamailio.local.cfg: #!substdef "!MSTDOMAIN!mydomain.com!g" kamailio.cfg: modparam("corex", "alias_subdomains", "MSTDOMAIN") But in the cli: kamctl rpc corex.list_aliases { "jsonrpc": "2.0", "result": [ ], "id": 1319303 } I don't remember to have this issue when I was developing the mst integration. Either I missed it or can it be a regression? kamailio version is 5.6.4 at the moment. Any ideas? cheers Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

3 months, 3 weeks

Just need some basic help

by Jere Cassidy

Hello! Thanks in advance for any help you can provide. I have a simple setup here and am struggling a bit with getting a final working solution. This is just a single direction flow from a SIP provider inbound to servers running HMPelements with CTI integration (sip endpoint). Essentially a different DN / dial number would go to a different backend server. There is no registration involved. The kamailio server therefore needs to sit between the SIP provider and the backend servers, it will be a NAT situation where private RFC1918 addresses will be used on the backend after the server receives it with public IP information. So that's it. What is the simplest way to accomplish this? If it would help - here's what ive done so far: For the Kamailio server, I have an AWS EC2 instance with a public IP bound/assigned to the main private IP. During troubleshooting, I've added another private IP to the NIC so that I could throw up another listener. I was thinking I'd need to accept the SIP on the first listener with the "advertise" directive. The second listener would not have an advertise directive so perhaps could be used to talk to the backends. I defined the WITH NAT and rtp proxy settings (pointing to the first listener below). The RTP Proxy daemon also knows about the y.y.y.y public address below. In the kamailio.cfg, I have a listener setup : listen=udp:172.24.40.28:5060 advertise y.y.y.y:5060 listen=udp:172.24.40.26:5060 then threw in a route(CMS) in the request route and created the following route block: route[CMS] { #check URI matches phone xxxxxxxxxx if (uri=~"^sip:\+1xxxxxxxxxx@") { rewritehostport("10.12.18.193:5060"); forward(uri:host, uri:port); exit; }; } I get the SIP delivered to the backend host! but the INVITE headers are stil pointing to public IPs and the SDP info is not adjusted to be a private IP as well - as follows: z.z.z.z = sip provider , y.y.y.y = public ip (aws ec2 ip). xxxxxxxxx = calling phone, ######## = called phone (matches routeCMS above) INVITE sip:+1##########@10.12.18.193:5060 SIP/2.0 Record-Route: <sip:y.y.y.y;lr> Via: SIP/2.0/UDP y.y.y.y:5060;branch=z9hG4bK28b7.56b42fdeec9f10cd13030b3711b16f4d.0 Via: SIP/2.0/UDP z.z.z.z:11000;received=z.z.z.z;rport=11000;branch=z9hG4bKQ2yQKgv7NtKQF Max-Forwards: 48 From: "" <sip:+1xxxxxxxxxx@z.z.z.z;isup-oli=0>;tag=tUF15gaNg6N6j To: <sip:+1##########@y.y.y.y> Call-ID: 9ee7e073-c7fc-46ac-a839-f3c77eaed903 CSeq: 78244364 INVITE Contact: <sip:mod_sofia@z.z.z.z:11000> User-Agent: 2600Hz Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY Supported: path, replaces Allow-Events: talk, hold, conference, refer Privacy: none Content-Type: application/sdp Content-Disposition: session Content-Length: 278 X-FS-Core-UUID: 4e0b2928-d913-42c4-9809-f77a36d281be X-FS-Support: update_display,send_info P-Asserted-Identity: "" <sip:+1xxxxxxxxx@z.z.z.z> v=0 o=FreeSWITCH 1705572686 1705572687 IN IP4 z.z.z.z Thanks much!

3 months, 3 weeks

Re: add a SIP custom header on a SIP response generated locally

by frédéric Gaisnon

Hi, Yes I've tried using append_to_reply() in several routes (tm:branch-failure, sl:local-response, tm:local-response, onreply_route, failure_route) but it was not applied. It works fine if a response is received or if it is generated by the configuration script but not if this response is locally generated by a kamailio module. Regards, Frédéric Gaisnon > Custom headers to the reply can be added using append_to_reply() function from textops module. Have you tried that? > > Regards, > > Fred Posner > p: +1 (352) 664-3733 > > On Jan 18, 2024, at 2:07 AM, frédéric Gaisnon via sr-users < sr-users(a)lists.kamailio.org> wrote: > > > > Hello, > > I want to add a SIP custom header on a SIP response generated locally (on the 408 generated when fr_inv_timeout is expired for example) > > I've tried functions append_hf and append_to_reply in event_route[tm:branch-failure:XXX] or event_route [tm:local-response] without any success. > > Do you have any suggestions to accomplish this ? > > Regards, > > Frédéric Gaisnon

3 months, 3 weeks

trigger different branch route on branch added with append_branch

by Benoit Panizzon

Hi List Imagine the following situation: Two Kamailio registrar proxies handling the same domain. Location Information synced via DMQ. So if a device registers we don't know on which of the two registrar proxies. Core proxy is dispatching calls to either one of those registrars. Core also performs AOR database lookup and adds an X- header telling which AOR the call is destined to. Registrar performs lookup of that AOR in the location database and routes the call to destination. Problem: The call could be routed to the the registrar which does not hold the active socket to that client. Solution: reg_fetch_contacts, cycle the contacts and append_branch if the contact has a local socket. Next Problem: I use a failure_route to sent the call to voicemail, if not registered, busy etc. This would trigger, if we have no local socket for the registration which is on the other registrar. So I have to find a way to send the call to the other registrar. I guess I can append_branch a branch pointing to the URI of the other registrar. This would then succeed in reaching the device. Also using the failure_route would work this way if we get a remote error from behind the other registrar or no registration at all. Next Problem: I use a branch trigger to remove undesired header I do not want to send to the customer. Now imagine that AOR has two contacts, one on registered on each registrar proxy. So I end up with two destinations added with append_branch to the destination set. One points to the device of the locally registered customer device, the other one points to the other registrar where to find the other device. I use a branch route trigger to remove the X- Headers towards the customer. But I need to X- Header containing the destination AOR to still be sent to the other registrar. Is there a way to trigger a branch route for only some of the destinations added with append_branch? Or to trigger different branch routes for different destinations in one set added with append_branch? Or am I doing this completely wrong and there is an easier way? ;-) Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

3 months, 3 weeks

How to change uri for db_mysql ?

by iglov

Sup guyz! On debian 12 i have kamailio 5.6.3 and when I try to use db_mysql module I always get an error: kamailio[2842268]: ERROR: db_mysql [km_my_con.c:163]: db_mysql_new_connection(): driver error: Can't connect to local server through socket '/run/mysqld/mysqld.sock' I have no idea where it finds this socket cuz I don’t have it and don’t want to. Where I can change it?

3 months, 3 weeks

add a SIP custom header on a SIP response generated locally

by frédéric Gaisnon

Hello, I want to add a SIP custom header on a SIP response generated locally (on the 408 generated when fr_inv_timeout is expired for example) I've tried functions append_hf and append_to_reply in event_route[tm:branch-failure:XXX] or event_route [tm:local-response] without any success. Do you have any suggestions to accomplish this ? Regards, Frédéric Gaisnon

3 months, 3 weeks

Kamailio v5.7.4 Released

by Daniel-Constantin Mierla

Hello, Kamailio SIP Server v5.7.4 stable release is out. This is a maintenance release of the latest stable branch, 5.7, that includes fixes since the release of v5.7.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.7.x. Deployments running previous v5.7.x versions are strongly recommended to be upgraded to v5.7.4. For more details about version 5.7.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2024/01/kamailio-v5-7-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio Advanced Training, February 20-22, 2024 -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

3 months, 3 weeks

Releasing v5.7.4

by Daniel-Constantin Mierla

Hello, I am considering to release Kamailio v5.7.4 (out of branch 5.7) on Thursday, Jan 18, 2024. If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy, Training and Development Services -- asipto.com Kamailio Advanced Training, February 20-22, 2024 -- asipto.com Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

3 months, 3 weeks

IMS Register failure

by srinidhi 2wewew

I am trying to get imsdroid register to pcscf . But failing with 504 server timeout . PCSCF IP : 192.168.1.28 imsdroid IP : 192.168.1.212 Please help to figure out the reaso for register failure . I have attached pcap file and pcscf log file

3 months, 3 weeks

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sr-users January 2024