sr-users June 2020

sr-users@lists.kamailio.org

71 participants
105 discussions

by Trần Đức Thành

Hi, Anyone can help Sincerely, ---- TRAN DUC THANH Technician ZLINK TECHNOLOGY CO., Ltd Add: 4th Floor, 14 Phao Dai Lang Str, Dong Da Dist., Hanoi, Vietnam 186 Bis Tran Quang Khai, Dist 1, HCM, Vietnam Tel: (84-24) 7 309 99 68 / (84-28) 7309 99 68 Mobile: (84)988612049 - (84)904706808 - Skype ID: thanhtd07 Email: [ mailto:tranthanh@voip.com.vn | tranthanh(a)voip.com.vn ] ZLINK - VoIP Solution - Software Licensing - Server&Network Device [ http://voip.com.vn/ | VoIP.com.vn ] - [ http://azsoft.com/ | AzSoft.com ] - [ http://serverhot.net/ | ServerHot.Net ]

4 years, 5 months

Reconnect call

by Gaurav Bmotra

hi how we can implement , that if call got disconnected due to low network and will reconnect .. pl help me -- *Regards:* Gaurav Kumar

4 years, 5 months

Does DMQ auto-discover cluster members + how?

by George Diamantopoulos

Hello all (again), When running dmq.list_nodes on one of my production kamailio servers, I get several entries corresponding to staging instances of kamailio. These staging instances are located on separate broadcast domains (different VLANs), and none of them are configured in the notification_address parameter for the dmq module. How do they end up there, and how do I get rid of them? Thanks! George

4 years, 5 months

RTPProxy Traversal for all Calls

by Mike B

All, Hopefully a quick answer on this one. I am looking for away to have all traffic traverse the RTP relay, not only NAT. Thinking there should be an easy way to do this with RTPPROXY but haven't been able to figure it out. Using Kamailio 5.3 with RTPProxy 2.2 Thanks, Mike

4 years, 5 months

Homer 7 and Kamailio

by Social Boh

Hello, Does Kamailio works with homer sip capture server 7 and HEPv3? I'm trying to test without success. Regards -- --- I'm SoCIaL, MayBe

4 years, 5 months

Re: [SR-Users] Issue of calls drop 32s

by Rex Lin (林昱頡)

Hi Alex, Thanks for your reply. ICE/STUN haven’t been activated by our clients. All transmission traffic pass through a proxy server. Below is the screenshot of a normal call process. [cid:image001.png@01D62AC5.FBF723D0] Best Regards, Rex -----Original Message----- From: sr-users <sr-users-bounces(a)lists.kamailio.org> On Behalf Of Alex Balashov Sent: Thursday, May 14, 2020 12:18 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] Issue of calls drop 32s It's a tell-tale sign of an end-to-end ACK from the caller not being forwarded properly to the callee. The server should not be forwarding such an ACK "to itself"; this suggests that its Request URI may be improperly constructed (it should be equivalent to the remote Contact URI of the callee in the 200 OK message), or that its Route headers are somehow incorrect (they should mirror the Record-Route set in the 200 OK). -- Alex On Wed, May 13, 2020 at 04:07:16PM +0000, Rex Lin (林昱頡) wrote: > Hi, > > > It’s weird that the calls always drop after 32s while the callee is using public ip. Also, Server forwards ACK to itself with UDP, instead of forwarding to callee with TLS. > > > Moreover, I cannot see “Received” in AOR as the callee finish the registration. > > > I recognized it’s marked by nat_uac_test(“19”) and set with fix_nated_register(), the problem can be resolved. > > > My scenario is as below, for your reference. > > 1.Kamailio version is 4.4.7, listening public ip. > > 2.Caller it’s behind NAT. > > 3.Clients of Linphone. > > > Do you know the reason why it happened? > > > > Best Regards, > > Rex > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

4 years, 5 months

kamailio 522 S-CSCF failing to start

by Martin W Woscek

Hello, There are times that our S-CSCF module just wont start. I’ve tried correlating the issue with some operation such as not de-registering an IMS client (Boghe or imsdroid) prior to shutting down the Kamailio server. But this isn’t always the case. Being that we are running this an a Ubuntu Virtual Machine, simply overwriting the image with a backed up VM is the temporary solution. I’ve dropped and created the db for S (kamdbctl) and rearranged the order of loading the presence module with same result. Status of the S-CSCF daemon on failure to start: # systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: inactive (dead) Docs: man:systemd-sysv-generator(8) Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:22:55 ubuntu /usr/local/sbin/kamailio[16511]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: * already running Jun 08 08:22:55 ubuntu kamailio_scscf[16495]: ...done. Jun 08 08:22:55 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Jun 08 08:24:03 ubuntu systemd[1]: Stopping LSB: Start the Kamailio SIP proxy server... Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: * Stopping Kamailio IMS S-CSCF: kamailio_scscf Jun 08 08:24:03 ubuntu kamailio_scscf[16517]: ...done. Jun 08 08:24:03 ubuntu systemd[1]: Stopped LSB: Start the Kamailio SIP proxy server. root@ubuntu:~/IMS_manual_scripts# systemctl start kamailio_scscf root@ubuntu:~/IMS_manual_scripts# systemctl status kamailio_scscf ● kamailio_scscf.service - LSB: Start the Kamailio SIP proxy server Loaded: loaded (/etc/init.d/kamailio_scscf; generated) Active: active (exited) since Mon 2020-06-08 08:27:23 EDT; 58s ago Docs: man:systemd-sysv-generator(8) Process: 16543 ExecStart=/etc/init.d/kamailio_scscf start (code=exited, status=0/SUCCESS) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [db.c:457]: db_check_table_version(): invalid version 0 for table presentity found, expected 5 (check table structure and ta Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: presence [presence.c:367]: mod_init(): error during table version check Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: ERROR: <core> [core/sr_module.c:898]: init_mod(): Error while initializing module presence (/usr/local/lib64/kamailio/modules/presence.so) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: WARNING: <core> [core/mem/q_malloc.c:480]: qm_free(): WARNING: free(0) called from cdp_avp: cdp_avp_mod.c: cdp_avp_destroy(226) Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:255]: cdp_exit(): CDiameterPeer child stopping ... Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: cdp [cdp_mod.c:257]: cdp_exit(): ... CDiameterPeer child stopped Jun 08 08:27:23 ubuntu /usr/local/sbin/kamailio[16559]: INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: * already running Jun 08 08:27:23 ubuntu kamailio_scscf[16543]: ...done. Jun 08 08:27:23 ubuntu systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Any insight is appreciated. Thanks, _Martin

4 years, 5 months

RTP port range in kuberenetes for kamailio

by Gaurav Bmotra

hi i have issue audio issue with kamailio in kubernetse cluster anybody , idea how to fix it in kubernetses thank you -- *Regards:* Gaurav Kumar

4 years, 5 months

How to change contact in SIP INVITE

by Trần Đức Thành

Hi everyone, I use kamailio and have some problem, need support. My scenario: - Kamailio ( x.x.x.x ) --> SIP provider (y.y.y.y) When i callout, SIP INVITE have 2 Record Route (1 x.x.x.x , 1 127.0.0.1) and 2 Via (1 x.x.x.x , 1 127.0.0.1), SIP provider reject it. I try to enable topos, SIP INVITE now have 1 Record Route x.x.x.x , 1 Via x.x.x.x. Everything look ok but Contact be masked like <sip:abcdefghijk@x.x.x.x> and SIP Provider reject it. Any one know how to have Contact as <sip:0123456789@x.x.x.x> where 0123456789 is our SIP number. Sincerely, ---- TRAN DUC THANH IT Manager ZLINK TECHNOLOGY CO., Ltd Add: 4th Floor, 14 Phao Dai Lang Str, Dong Da Dist., Hanoi, Vietnam 186 Bis Tran Quang Khai, Dist 1, HCM, Vietnam Tel: (84-24) 7 309 99 68 / (84-28) 7309 99 68 Mobile: (84)988612049 - (84)904706808 - Skype ID: thanhtd07 Email: [ mailto:tranthanh@voip.com.vn | tranthanh(a)voip.com.vn ] ZLINK - VoIP Solution - Software Licensing - Server&Network Device [ http://voip.com.vn/ | VoIP.com.vn ] - [ http://azsoft.com/ | AzSoft.com ] - [ http://serverhot.net/ | ServerHot.Net ]

4 years, 5 months

Re%3A RPC Command tm.clean&In-Reply-To=<9c051ed0-20b5-cf64-3e9a-cc60111a7e7c%40gmail.com>

by Marat Gareev

Hi, Thanks for your reply! What do you mean by "freezing operation system"? Can I manually reproduce stalled transactions? I tried to send SIGSTOP to a group of Kamailio processes while the transaction is live. The idea is to freeze processes for a while, then continue to work, so that by this time the transaction deletion time was missed. But after a few seconds, the transaction is still cleared.

4 years, 5 months

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sr-users June 2020