sr-users April 2014

sr-users@lists.kamailio.org

100 participants
170 discussions

by aawaise

Hello, I want to ask some basic questions. Anyone please guide me to links with answers. I am sure such basics must be explained online. 1. What is exactly achieved by "route(RELAY)" ?? Routing calls from local users to other domains ?? 2. If user 1 is caller and user 2 is being called. Which pseudo variable will be used to catch value of called user i.e user 2 and his domain ?? I want to pick domain name from location table in database and not from the authorization module. Thanks, Cheers, Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Basic-question-regarding-Routing-tp… Sent from the Users mailing list archive at Nabble.com.

10 years, 1 month

Routing decision based on domain entry in Location Table

by aawaise

Hello, I have deployed two kamailio servers, both sharing the same database. I want my kamailio servers to make a decision on the routing of incoming call by seeing the domain of "called user" from location table of online users. If the "called user" is connected to same server, the call should be routed to user directly. In other case if the user is connected to other server, the call should be forwarded to other user for further processing. Will someone please guide me how can this be achieved ?? P.S : I am using kamailio-3.1.1. Thanks, Regards, Aawaise. -- View this message in context: http://sip-router.1086192.n5.nabble.com/Routing-decision-based-on-domain-en… Sent from the Users mailing list archive at Nabble.com.

10 years, 1 month

rls multipart body does not contain watcherinfo+xml

by Mihai Marin

Hello Sirs, I'm trying to accomplish a rls subscribe but I have some miss-understandings regarding the multipart publish response that should come. I'm using the integrated xcap from the kamailio with all the xml files added. My problem is that event if I request a multipart containing application/pidf+xml, application/rlmi+xml, application/watcherinfo+xml I only get the first two. The request contains the following headers: Event: presence Supported: eventlist Accept: multipart/related, application/pidf+xml, application/rlmi+xml, application/watcherinfo+xml The response only sends the first two: --4kSPpRGyJQct4DIhyGnyobqO Content-Transfer-Encoding: binary Content-ID: <1397508890.sip:bob@93.187.138.214.681064155> Content-Type: application/rlmi+xml;charset="UTF-8" <?xml version="1.0"?> <list uri="sip:bob@93.187.138.214" xmlns="urn:ietf:params:xml:ns:rlmi" version="4" fullState="false"><resource uri="sip:alice@93.187.138.214"><instance id="Scf8UhwQ" state="active" cid="1397508890.sip:alice@93.187.138.214.1552580371"/></resource></list> --4kSPpRGyJQct4DIhyGnyobqO Content-Transfer-Encoding: binary Content-ID: <1397508890.sip:alice@93.187.138.214.1552580371> Content-Type: application/pidf+xml <?xml version="1.0" encoding="UTF-8" standalone="no"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" entity="sip:alice@93.187.138.214"> <dm:person id="p6850"> <rpid:activities/> <rpid:status-icon>http://93.187.138.214:5060/xcap-root/oma_status-icon/users/sip:alice@93.187…</rpid:status-icon> </dm:person> <tuple id="t7673"> <status> <basic>open</basic> </status> <contact>sip:alice@93.187.138.214</contact> <note>Online</note> </tuple> </presence> --4kSPpRGyJQct4DIhyGnyobqO-- What do I missing? I need the body part with the watchers info in oder to see the state of the watcher (active, pending) - to accomplish the authorization for users. Thank you in advance for you help. Best regards, Mihai M

10 years, 1 month

by betul.cerit＠netgsm.com.tr

Hi to all We integrated kamailio and asterisk as explained here http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb When kamailio receives a register request it authenticates and forwards the request to asterisk.Why do we need to do this?What happens if we dont pass registration requests to asterisk? Thanks in Advance Betül

10 years, 1 month

Security

by Keith

Hi All, I am looking to apply some security on invites so I only accept from IP addresses in the address table or authenticated users. I have the address table bit working but I am struggling with the authenticated users bit. Wondering if anyone has done it? I was thinking of using the location table in some way as those users have authenticated? Any ideas would be appreciated. Keith

10 years, 1 month

uac_auth problem

by neumann

Hi all sorry for my english! Iam trying to authenticate as uac on 3rd party server(Asterisk). Kamailio get 401, send next invite with DIGEST but Asterisk send 401 again((( Asterisk don’t write in log what auth is failed. 115.600045 xxx.xxx.xxx.xxx -> yyy.yyy.yyy.yyy SIP/SDP 1093 Request: INVITE sip:321321231@xxx.xxx.xxx.xxx, with session description 115.601818 yyy.yyy.yyy.yyy -> xxx.xxx.xxx.xxx SIP 674 Status: 401 Unauthorized 115.602470 xxx.xxx.xxx.xxx -> yyy.yyy.yyy.yyy SIP 389 Request: ACK sip:321321231@xxx.xxx.xxx.xxx 115.604310 xxx.xxx.xxx.xxx -> yyy.yyy.yyy.yyy SIP/SDP 1267 Request: INVITE sip:321321231@xxx.xxx.xxx.xxx, with session description 115.605537 yyy.yyy.yyy.yyy -> xxx.xxx.xxx.xxx SIP 674 Status: 401 Unauthorized 115.606431 xxx.xxx.xxx.xxx -> yyy.yyy.yyy.yyy SIP 389 Request: ACK sip:321321231@xxx.xxx.xxx.xxx My aster peer: [kamailio] type=peer defaultuser=kamailio secret=1234 host=dynamic disallow=all allow=alaw nat=no qualify=yes My kamailio config: modparam("uac","auth_username_avp","$avp(s:uac_user)") modparam("uac","auth_password_avp","$avp(s:uac_pass)") modparam("uac","auth_realm_avp","$avp(s:uac_realm)") route[DISPATCH_OUT] { if(!ds_select_dst( 4, 8)){ send_reply("404", "No destination"); exit; } t_set_fr(0,2000); t_on_failure("RTF_DISPATCH_OUT"); t_relay(); exit; } failure_route[RTF_DISPATCH_OUT] { if (t_is_canceled()) { exit; } if ( t_check_status("401|407") ) { $avp(s:uac_user) = "kamailio"; $avp(s:uac_pass) = "1234"; $avp(s:uac_realm) = "asterisk"; if (isflagset(7)) { t_reply("513","Authentication failed"); xlog("L_INFO", "[%ci]: Remote authentication failed\n"); break; } if (uac_auth()) { xlog("L_INFO", "401/407 message recived"); # mark that auth was performed setflag(7); # trigger again the failure route t_on_failure("RTF_DISPATCH_OUT"); # repeat the request with auth response this time append_branch(); #t_relay_to_udp("94.25.100.198","5060"); t_relay(); break; } else { xlog("L_INFO", "uac_auth filed!!!!!!!!!!!"); } t_reply("514","Unknow authentication peer"); xlog("L_INFO", "[%ci]: Unknown authentication peer.\n"); break; } # next DST - only for 500 or local timeout if (t_check_status("500") or t_check_status("480") or (t_branch_timeout() and !t_branch_replied())){ ds_mark_dst("p"); if(ds_next_dst()){ t_set_fr(0,2000); t_on_failure("RTF_DISPATCH_OUT"); route(RELAY); exit; } } } ———————————— Timofeev Dmitry VoIP Engineer Linux, Asterisk, Freeswitch, Cisco solutions Skype: itsroot icq: 227227933

10 years, 1 month

client provisioning.

by Slava Bendersky

Hello Everyone, Is possible do phone provisioning with kamailio ? Slava.

10 years, 1 month

PERMISSIONS module issue

by Samuel Ware

I having issue updating my allow list for the PERMISSIONS module. I added an address to the ADDRESS table. I have tried to do a service restart, kamctl address reload, and kamcmd permissions.addressReload. The kamctl address show displays the new address; however kamcmd permissions.addressDump does not neither does kamcmd permissions.subnetDump. The messages from this new address return a false to the !allow_source_address("1”) command in my routing logic. I am wondering if this is a bug or I am doing something wrong. I am on the most recent GIT version to the best of my knowledge. Sam

10 years, 1 month

SDP body rewritten issue at RTPproxy

by Wingsravi R

Dear All, I am working with Kamailio (4.1.2) and RTPproxy (1.2.1) servers and Jitsi as SIP clients. I have a problem in Audio/Video calling with Kamailio+RTPproxy server combination that Video is not streaming to other side through RTPproxy(also no audio both sides), I am suspecting something is rewritten in SDP body at RTPproxy, concerning to audio/video parameters (I am unable to rectify what is causing problem there) . Please find the Tcpdump based SIP capture of video call with this Kamailio+RTPproxy set-up. And with only kamailio server running (without RTPproxy instance running), audio/video call is fine on both sides. (find SIP capture for the same call). Also find my kamailio.cfg file for any reference. Please anybody help me in rectifying problems at RTPproxy server. Awaiting somebody's comments on this issue. Regards, Ravi

10 years, 1 month

Re: [SR-Users] SDP c= IP doesn't get rewritten

by Ciprus, Daniel

Yes I do Sent via the Samsung GALAXY S®4, an AT&T 4G LTE smartphone -------- Original message -------- From: Daniel-Constantin Mierla Date:04/11/2014 6:13 AM (GMT-05:00) To: "Ciprus, Daniel" ,"Kamailio (SER) - Users Mailing List" Subject: Re: [SR-Users] SDP c= IP doesn't get rewritten Hello, are you using a rtp relay (e.g., rtpproxy)? Cheers, Daniel On 11/04/14 02:46, Daniel Ciprus wrote: All, I am facing problem with rewriting SDP part of the INVITE, in particular following is in my configuration: #!subst "/RTP_INT_ADDR/10.96.173.15/" #!subst "/RTP_EXT_ADDR/168.7.24.17/" #!subst "/BOT_RTP_ADDR/10.96.173.52/" subst_body('/RTP_INT_ADDR/RTP_EXT_ADDR/g'); subst_body('/BOT_RTP_ADDR/RTP_EXT_ADDR/g'); and this is what I ended up with: c=IN IP4 10.96.173.15168.7.24.17 Apparently 2nd rule doesn't work rewrite, it's doing append instead. What I want to achieve is : if SDP part contains BOT_RTP_ADDR -> rewrite it. In case SDP part contains RTP_INT_ADDR-> rewrite it, so 2x if. I have tried to use sdpops module, it's coring in my case and I have a reason to not upgrade, I just can't. Any other ideas how to handle my problem ? Thanks Dan Server:: kamailio (4.2.0-dev0 (x86_64/linux)) Build:: mi_core.c compiled on 17:42:57 Dec 18 2013 with gcc 4.4.7 Flags:: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES GIT:: unknown -- Daniel Ciprus Integration engineer http://www.acision.com 9954 Mayland Dr Suite 3100 Richmond, VA 23233 USA T: +1 804 762 5601 E: daniel.ciprus(a)acision.com<mailto:daniel.ciprus@acision.com> ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org<mailto:sr-users@lists.sip-router.org> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda ________________________________ This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you for understanding.

10 years, 1 month

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sr-users April 2014