sr-users March 2014

sr-users@lists.kamailio.org

101 participants
165 discussions

by GRIBANOV, Oleg

Dear sirs, I would like to make some tests of the SIP connection using your server. Could you please confirm me the following details: - IP of the server? Is it 212.79.11.155 ? - How could I register and use the STUN server ? What is the address of STUN server ? Thank you, Oleg Gribanov ________________________________ Important Notice: THIS EMAIL (INCLUDING ANY ATTACHMENTS) IS CONFIDENTIAL AND MAY BE PRIVILEGED. IF YOU ARE NOT THE INTENDED RECIPIENT, PLEASE CONTACT THE SENDER, AND REFRAIN FROM COPYING, FORWARDING, DISCLOSING, DISTRIBUTING OR OTHERWISE USING THIS EMAIL OR ANY PART OF IT. DOING OTHERWISE MAY CONSTITUTE AN OFFENCE OR CONTRACTUAL BREACH. ALL DISCUSSIONS AND DRAFTS CONTAINED IN OR ATTACHED TO THIS EMAIL ARE WITHOUT COMMITMENT AND ARE SUBJECT TO CONTRACT, UNLESS EXPRESSLY STATED OTHERWISE. CET EMAIL (AINSI QUE TOUTES SES PIECES JOINTES) EST CONFIDENTIEL ET PEUT ETRE COUVERT PAR LE SECRET PROFESSIONNEL. SI VOUS N'ETES PAS LE DESTINATAIRE D'ORIGINE, VEUILLEZ CONTACTER L'EXPEDITEUR, ET ABSTENEZ-VOUS DE LE COPIER, TRANSFERER, REVELER, DIFFUSER OU FAIRE UN QUELCONQUE USAGE DE CET EMAIL, EN TOUT OU PARTIE. ENTREPRENDRE UNE TELLE ACTION PEUT CONSTITUER UN DELIT OU UNE FAUTE CONTRACTUELLE. TOUTES DISCUSSIONS ET PROJETS QUI SONT INCLUS OU JOINTS A CET EMAIL SONT SANS ENGAGEMENT DES SOCIETES DU GROUPE EUROSPORT ET DOIVENT FAIRE L'OBJET D'UN CONTRAT, SAUF ACCORD EXPRES CONTRAIRE. ________________________________

10 years, 1 month

Rx interface

by Dees

Does Kamailio support full Rx interface?

10 years, 1 month

Integrating Kamailio and Freeswitch

by mark li

Hi there. I'm a noobie to Kamailio and Freeswitch... but I'm trying to follow the article located here: http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc I've tried to add all the sections marked with WITH_FREESWITCH in the sample config in the article into my own kamailio-advanced.cfg file. Here's what my cfg file looks like: http://pastebin.com/KsvrYVN7 I've restarted kamailio after making these changes. Then I tried to dial 41 to listen to vmail or 44999 to leave a message for user 999 but both return a busy tone. Any suggestions would be appreciated. Thanks.

10 years, 1 month

Need to confirm the changes for the Kamailio.cfg

by varun pratapsingh

HI All, This is my first post to this community and I am new user of the Kmailio. I want to use the Kamailio and RTPProxy on the same machine. I have installed and rtpproxy succesfully and I have started both the kamailio and RtpProxy. My RtpProxy is listening on localhost:7722 port of my local machine. Now this machine is on the public IP and I want that all sip and RTP passes though this machine from SIP Softphones before going to the SIp Server. This means the machine on which I am running the Kamailio and RTPProxy will act like a SIP proxy from my SIP Softphones through which the SIP and RTP relays. SIP is relaying fine with defult kamailio.cfg but can anyone let me know the changes in Kamailio.cfg so that my RTP also passes or rleays through this Kamailio + RTPProxy machine. Thanks in advance. Thanks and Regards Varun

10 years, 1 month

Preventing information about my sip network

by Arya Farzan

I'm concerned about others reverse engineering their way into my project's sip network. Is there anyway to prevent others from finding out that the SIP protocol is being used and prevent others to reverse engineer their way into my sip network?

10 years, 1 month

Rewrite SIP INVITE

by Mickael MONSIEUR

Hello, I installed kamailio and asterisk with the tutorial of asipto. For alias numbers I configured the module alias_db. Everything works because Asterisk outgoing call is directed at Kamailio, then Kamailio is sending to the alias registered (table location) Only the number alias appears in the "To" (02XXXXXX) and not in the INVITE URI. It only shows "s" ... is it possible to force writing 02XXXXXX instead of "s" ? Example: INVITE sip:s@10.1.0.191:5060 SIP/2.0 Max-Forwards: 69 From: "0475XXXXXX" <sip:1053212@sip.domain.com>;tag=as7df9ab18 To: <sip:02XXXXXX@kamailioIP:5060> Contact: <sip:1053212@asteriskIP:5060> Call-ID: 344d42bd16975a54141d11f635bdfc71(a)sip.domain.com CSeq: 102 INVITE Date: Wed, 26 Mar 2014 15:06:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 252 Because without it, Asterisk servers behind Kamailio not will route the call to the correct extension but to the "s". Asterisk ignores the "To" apparently this is strange ... Thank you, Mickael

10 years, 1 month

mediaproxy-ng, invalid header version

by Mihai Marin

Hellor Sirs, Sir Richard, I saw some updates in the last 2 weeks that were working with Firefox - I also did some tests as it was working. Now, I tried to get the latest version and I'm getting the following error: mediaproxy-ng[2747]: Got valid command from 127.0.0.1:35127: answer - { "sdp": "v=0#015#012o=Mozilla-SIPUA-29.0a2 13431 0 IN IP4 0.0.0.0#015#012s=SIP Call#015#012t=0 0#015#012a=ice-ufrag:8a9a3649#015#012a=ice-pwd:80b3b796ecdd0addb079fc531c7e0afa#015#012a=fingerprint:sha-256 A3:D7:1F:F6:60:1D:91:27:64:89:2E:09:CE:42:19:DB:7E:5F:02:06:A3:DA:E0:26:0F:F7:30:4C:1E:65:86:2B#015#012m=audio 50990 RTP/SAVPF 109 101#015#012c=IN IP4 188.215.94.132#015#012a=rtpmap:109 opus/48000/2#015#012a=ptime:20#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=recvonly#015#012a=setup:active#015#012a=candidate:0 1 UDP 2128609535 192.168.0.5 50990 typ host#015#012a=candidate:1 1 UDP 1692467199 188.215.94.132 50990 typ srflx raddr 192.168.0.5 rport 50990#015#012a=candidate:0 2 UDP 2128609534 192.168.0.5 50991 typ host#015#012a=candidate:1 2 UDP 1692467198 188.215.94.132 50991 typ srflx raddr 192.168.0.5 rport 50991#015#012a=rtcp-mux#015#012m=video 50992 RTP/SAVPF 120#015#012c=IN IP4 188.215.94.132#015#012a=rtpmap:120 VP8/90000#015#012a=sendrecv#015#012a=rtcp-fb:120 nack#015#012a=rtcp-fb:120 nack pli#015#012a=rtcp-fb:120 ccm fir#015#012a=setup:active#015#012a=candidate:0 1 UDP 2128609535 192.168.0.5 50992 typ host#015#012a=candidate:1 1 UDP 1692467199 188.215.94.132 50992 typ srflx raddr 192.168.0.5 rport 50992#015#012a=candidate:0 2 UDP 2128609534 192.168.0.5 50993 typ host#015#012a=candidate:1 2 UDP 1692467198 188.215.94.132 50993 typ srflx raddr 192.168.0.5 rport 50993#015#012a=rtcp-mux#015#012", "flags": [ "force", "trust-address" ], "replace": [ "origin", "session-connection" ], "call-id": "ufdo7gas8lt6jpco60b1", "received-from": [ "IP4", "188.215.94.132" ], "from-tag": "nj7tuhg5hj", "to-tag": "7lpqrsa1eb", "command": "answer" } mediaproxy-ng[2747]: [ufdo7gas8lt6jpco60b1] Returning to SIP proxy: d3:sdp1463:v=0#015#012o=Mozilla-SIPUA-29.0a2 13431 0 IN IP4 0.0.0.0#015#012s=SIP Call#015#012t=0 0#015#012a=ice-ufrag:8a9a3649#015#012a=ice-pwd:80b3b796ecdd0addb079fc531c7e0afa#015#012m=audio 30002 RTP/SAVPF 109 101#015#012c=IN IP4 93.187.138.212#015#012a=rtpmap:109 opus/48000/2#015#012a=ptime:20#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=candidate:0 1 UDP 2128609535 192.168.0.5 50990 typ host#015#012a=candidate:1 1 UDP 1692467199 188.215.94.132 50990 typ srflx raddr 192.168.0.5 rport 50990#015#012a=candidate:0 2 UDP 2128609534 192.168.0.5 50991 typ host#015#012a=candidate:1 2 UDP 1692467198 188.215.94.132 50991 typ srflx raddr 192.168.0.5 rport 50991#015#012a=recvonly#015#012a=rtcp:30002#015#012a=rtcp-mux#015#012a=setup:active#015#012a=fingerprint:sha-1 CA:7D:5B:F1:C8:1A:05:8E:9B:88:0F:36:9B:C4:7A:60:A3:B5:02:A4#015#012a=candidate:kgDINMdmuvprUMd6 1 UDP 2130706432 93.187.138.212 30002 typ host#015#012m=video 30006 RTP/SAVPF 120#015#012c=IN IP4 93.187.138.212#015#012a=rtpmap:120 VP8/90000#015#012a=rtcp-fb:120 nack#015#012a=rtcp-fb:120 nack pli#015#012a=rtcp-fb:120 ccm fir#015#012a=candidate:0 1 UDP 2128609535 192.168.0.5 50992 typ host#015#012a=candidate:1 1 UDP 1692467199 188.215.94.132 50992 typ srflx raddr 192.168.0.5 rport 50992#015#012a=candidate:0 2 UDP 2128609534 192.168.0.5 50993 typ host#015#012a=candidate:1 2 UDP 1692467198 188.215.94.132 50993 typ srflx raddr 192.168.0.5 rport 50993#015#012a=sendrecv#015#012a=rtcp:30006#015#012a=rtcp-mux#015#012a=setup:active#015#012a=fingerprint:sha-1 CA:7D:5B:F1:C8:1A:05:8E:9B:88:0F:36:9B:C4:7A:60:A3:B5:02:A4#015#012a=candidate:kgDINMdmuvprUMd6 1 UDP 2130706432 93.187.138.212 30006 typ host#015#0126:result2:oke mediaproxy-ng[2747]: [ufdo7gas8lt6jpco60b1 port 30000] Error parsing RTP header: invalid header version mediaproxy-ng[2747]: [ufdo7gas8lt6jpco60b1 port 30004] Error parsing RTP header: invalid header version mediaproxy-ng[2747]: [ufdo7gas8lt6jpco60b1 port 30002] Error parsing RTP header: invalid header version mediaproxy-ng[2747]: [ufdo7gas8lt6jpco60b1 port 30006] Error parsing RTP header: invalid header version I was dreaming one week ago when I saw Firefox working with chrome :)? Do you know something about this error? Thank you. Best regards, Mihai M

10 years, 1 month

Using Kamailio with a SIP Trunk

by Rizwan Khan

I want the following setup: 1 Kamailio server to handle internal calls (A/V), IM and Presence. 1 Asterisk or any other way to communicate with an NGN where I will create the SIP Trunk to route calls outside of the network. Is this the right approach or there is a way to directly communicate with the NGN to make a SIP trunk by using some external modules. Any guidelines will be highly appreciated. Rizwan Khan

10 years, 1 month

Memory leak when using db_unixodbc, help tracking it down (try 2)

by Alex Villacís Lasso

I have a CentOS 6 installation with the following packages installed from the RPM build service from Kamailio: kamailio-unixodbc-4.1.2-1.1.x86_64 kamailio-4.1.2-1.1.x86_64 kamailio-presence-4.1.2-1.1.x86_64 kamailio-utils-4.1.2-1.1.x86_64 I am also using db_unixodbc for all database accesses (with MySQL backend). The problem is that after a few hours, a single Kamailio process (never more than one) starts complaining that it has run out of memory: 9(11479) ERROR: uac [uac_reg.c:638]: uac_reg_tm_callback(): got sip response 408 while registering [admin_pbx.elastix.com] 9(11479) ERROR: db_unixodbc [dbase.c:335]: db_unixodbc_fetch_result(): no memory left 9(11479) ERROR: <core> [db_query.c:502]: db_fetch_next(): unable to fetch next rows 9(11479) ERROR: db_unixodbc [dbase.c:224]: db_unixodbc_free_result(): invalid parameter value 9(11479) ERROR: db_unixodbc [dbase.c:327]: db_unixodbc_fetch_result(): no private memory left 9(11479) ERROR: <core> [db_query.c:434]: db_fetch_query_internal(): unable to fetch the db result 9(11479) ERROR: presence [publish.c:108]: msg_presentity_clean(): failed to query database for expired messages 9(11479) ERROR: db_unixodbc [dbase.c:327]: db_unixodbc_fetch_result(): no private memory left 9(11479) ERROR: <core> [db_query.c:434]: db_fetch_query_internal(): unable to fetch the db result 9(11479) ERROR: presence [publish.c:108]: msg_presentity_clean(): failed to query database for expired messages 9(11479) ERROR: db_unixodbc [dbase.c:327]: db_unixodbc_fetch_result(): no private memory left 9(11479) ERROR: <core> [db_query.c:434]: db_fetch_query_internal(): unable to fetch the db result I have attached the stderr output when I kill the offending process. From what I can see, the memory leak somehow involves database allocations that fail to be freed, consuming more than 3 MB out of 4 MB allocated: 9(11479) NOTICE: <core> [main.c:857]: sig_usr(): Memory still-in-use summary (pkg): 9(11479) NOTICE: qm_sums: summarizing all alloc'ed. fragments: 9(11479) NOTICE: qm_sums: count= 1 size= 10240 bytes from mi_fifo: mi_writer.c: mi_writer_init(57) 9(11479) NOTICE: qm_sums: count= 378 size= 3029384 bytes from db_unixodbc: dbase.c: db_unixodbc_fetch_result(333) 9(11479) NOTICE: qm_sums: count= 1 size= 4096 bytes from db_unixodbc: dbase.c: db_unixodbc_fetch_result(325) 9(11479) NOTICE: qm_sums: count= 1 size= 16 bytes from <core>: sr_module.c: init_modules(1002) 9(11479) NOTICE: qm_sums: count= 1 size= 56 bytes from textops: textops.c: hname_fixup(1634) 9(11479) NOTICE: qm_sums: count= 1 size= 32 bytes from rr: rr_cb.c: register_rrcb(63) I have the setup to compile my own RPMS, and I intend to track down the memory leak on my own. However, I would like your help in focusing my search, on whether this is a known issue, or where in db_unixodbc should I look.

10 years, 1 month

Regarding SRV lookup

by Swetha Raj

Hi All, I am trying to communicate between two sip servers.The communication from external sip server user to kamailio user works but message from kamailio to external sip user is not reaching. I get the following error in Jitsi "The above message could not be delivered A network problem occurred. Please check your network configuration and try again. Error was: 408 Request Timeout" Does kamailio do SRV lookup ? How do i force kamailio to use DNS SRV lookup. Any clues ??

10 years, 1 month

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sr-users March 2014