- sr-users - kamailio.org

Diameter message AAR/AAA
by Ramjee Yadav 10 May '15

10 May '15

Dear, With changes incorporate in kamalio.cfg, I am able to REGISTER 200 ok with Rx interface. But I want to send AAR message only after receiving 200 ok at pcscf. Currently message sequences are CER -------> <-------CEA AAR ---------> <------AAA REGISTER---> <-------401 REGISTER----> <----200 ok Screen shot is attached for reference. But I want to Send AAR/AAA only after 200 ok. Below is the Sequence CER -------> <-------CEA REGISTER---> <-------401 REGISTER----> <----200 ok AAR ---------> <------AAA I using PCSCF as Diameter client and Seagull as a Diameter server. With changes in Kamailio.cfg Richard Good on Dec 17, 2014 Updated P-CSCF example with additional Rx_AAR params<https://github.com/kamailio/kamailio/commit/b450a769cff3b5c8ea0962299f6b087…> I am able to REGISTER but not able to send AAR/AAA after 200 ok.(Refer Attachment) Kindly let me know any other changes required in kamailio.cfg or seagull scenario.xml or some where else. and What changes. Thanks in advance. Br, Ramjee Y. ============================================================================================================================ Disclaimer: This message and the information contained herein is proprietary and confidential and subject to the Tech Mahindra policy statement, you may review the policy at http://www.techmahindra.com/Disclaimer.html externally http://tim.techmahindra.com/tim/disclaimer.html internally within TechMahindra. ============================================================================================================================

1 0

Recommended configuration for TCP support
by Mickael Marrache 08 May '15

08 May '15

Hi, We sometimes get issues caused by truncated SDP bodies. Therefore, we are looking at using TCP instead of UDP. What is the recommended configuration when TCP is used? Currently, I only have the following: children=16 listen=udp:<ip>:5060 listen=tcp:<ip>:5060 disable_tcp=no tcp_connection_lifetime=605 (registration expiry is 600) Also, is there a difference with UDP regarding NAT handling? Thanks, Mickael

3 7

db_text error
by jaflong jaflong 08 May '15

08 May '15

db_text error I am getting the following error on using db_text 0(6638) INFO: db_text [dbt_base.c:99]: dbt_init(): using database at: /usr/local/etc/kamailio/ 0(6638) ERROR: db_text [dbt_base.c:207]: dbt_query(): table version not loaded! (too few columns) 0(6638) ERROR: db_text [dbt_base.c:306]: dbt_query(): failed to query the table! I have thisin my kamailio.cfg #!define DBURL "text:///usr/local/etc/kamailio/dbtext" I created the following files cat > /usr/local/etc/kamailio/dbtext/version table_name(str) table_version(int) subscriber:6 location:6 aliases:6 cat > /usr/local/etc/kamailio/dbtext/subscriber username(str) password(str) ha1(str) domain(str) ha1b(str) someuser:someuser:xxx:10.1.1.5:xxx can some advise how to get db_text work for me Regards

2 3

Timer Process versus Slow Timer Process
by Mack Hendricks 08 May '15

08 May '15

Hi All, I'm looking to have a better understanding of when to use the Timer process versus the "Slow Timer" Process. This is what I came up with, but please correct me: - The “*Timer*” process is used to handle core processing within Kamailio. Therefore, you don’t want to put long running logic into the routing handler - The “*Slow Timer*” process can be used for long running processes

2 1

kamailio V 4.2.4 IMS_Charging ro_session_timeout question
by Yasin CANER 08 May '15

08 May '15

Hello; i am trying to understand how is ims_charging working . Kamailio is a S-CSCF here. kamailio and call dont have a problem but ro_session. i always get ro_session_ontimeout() ERROR. when look diameter trace , ims_charging module dont request for 30 more sec for granted and kill dialog. Thanks for helps. 14:22:51 S-CSCF ----->CC-Time 15 .. Sec CCR ---> Java-Diamaeter Server (INITIAL REQ) 14:22:51 S-CSCF <-----CC-Time 15 .. Sec CCA <--- Java-Diamaeter Server (INITIAL REQ) 14:23:01 S-CSCF ----->CC-Time 10 .. Sec CCR ---> Java-Diamaeter Server (UPDATE ) 14:23:01 S-CSCF <-----CC-Time 30 .. Sec CCA <--- Java-Diamaeter Server (UPDATE ) 14:23:31 S-CSCF ----->CC-Time 30 .. Sec CCR ---> Java-Diamaeter Server (TERMINATE ) 14:23:31 S-CSCF <-----CC-Time 30 .. Sec CCA <--- Java-Diamaeter Server (TERMINATE ) ------ LOG ----- May 7 14:23:31 /usr/local/sbin/kamailio[7602]: ERROR: ims_charging [ro_timer.c:449]: ro_session_ontimeout(): Diameter call session - event [2] May 7 14:23:31 /usr/local/sbin/kamailio[7602]: INFO: ims_charging [ro_timer.c:452]: ro_session_ontimeout(): Call/session must be ended - no more funds. May 7 14:23:31 /usr/local/sbin/kamailio[7600]: WARNING: ims_charging [dialog.c:167]: dlg_terminated(): inconsistent ro timer data on ro_session 0x7ff1eb4543b8 [] ---- CFG ------ $var(cc_ret) = Ro_CCR("CHARGING_CCR_REPLY", "orig", "SCUR", "", "15", "location"); ---- CFG ------

3 2

Make deb
by jay binks 07 May '15

07 May '15

is there any way to make deb packages for only 1 of the modules ?? currently it takes a long time to run "make deb" im simply trying to build and package db_cassandra ( need the package for my environment ) As a side note, is there any chance db_cassandra could be added to the kamailio deb packages ? -- Sincerely Jay

4 12

permissions module, regular expression matching Request URI
by Vik Killa 07 May '15

07 May '15

Hello, We would like to match specific calls based on two factors: 1. the source IP address 2. the Request URI I see in the permissions module, it is possible to match based on the Source IP + the From URI. We need two match both in order to return the appropriate 'tag' attribute from table. Would anyone know of a way to accomplish this? Also, I noticed that matching source IP + from uri, you cannot control which result is returned by permissions module based on regular expression. In other words if more than one match is made, permissions module seems to return the first result. Thank you and I look forward to your response. V

3 3

db_mongodb get columns error
by Mickael Marrache 07 May '15

07 May '15

Hi, After restarting Kamailio, I got the following errors during the startup: kamailio[12614]: ERROR: db_mongodb [mongodb_dbase.c:381]: db_mongodb_get_columns(): field [caller_sock] not found in result iterator kamailio[12614]: ERROR: db_mongodb [mongodb_dbase.c:739]: db_mongodb_store_result(): failed to set the columns kamailio[12614]: ERROR: db_mongodb [mongodb_dbase.c:918]: db_mongodb_query(): failed to store result kamailio[12614]: ERROR: db_mongodb [mongodb_dbase.c:927]: db_mongodb_query(): failed to do the query kamailio[12614]: ERROR: dialog [dlg_db_handler.c:257]: select_entire_dialog_table(): Error while querying database What may be causing the issue? I should always have a caller_sock in every dialog document. Thanks, Mickael

2 4

TCP connection reset by peer
by Mickael Marrache 07 May '15

07 May '15

Hi, I sometimes see the following in the logs: kamailio[14433]: ERROR: <core> [tcp_read.c:271]: tcp_read_data(): error reading: Connection reset by peer (104) kamailio[14433]: ERROR: <core> [tcp_read.c:1296]: tcp_read_req(): ERROR: tcp_read_req: error reading It looks like Kamailio tries to read a request from a TCP socket but the connection has been reset (not properly ended) by the remote peer. Is it an error? Does it prevent Kamailio from working properly? If yes, what is the solution? Thanks, Mickael

2 1

RFC: Config files installation
by Daniel-Constantin Mierla 06 May '15

06 May '15

Hello, cross posting to get heads up of everyone that wants to join the discussion, however, do reply to sr-dev(a)lists.sip-router.org, being more related to development and project files structure. At this moment we install three kamailio configuration files: - kamailio.cfg - the default one to be used when starting kamailio - kamailio-basic.cfg - a more simple version of configuration - kamailio-advanced.cfg - a more complex version (equivalent of kamailio-oob.cfg in etc/ source tree) Somehow it was suggested that apart of kamailio.cfg, the other files belong more to the docs (shared) folder (where we put INSTALL, README, etc), because they are not actually used as configuration files by kamailio installation (one has to change either init.d script or copy over kamailio.cfg). Also, some are confused not knowing if they are used or not at runtime. Shall we relocate where those files are deployed by installation? Another point: we deploy the radius dictionary with the core. I think it should be deployed by a radius module and be part of radius modules package. The question is which module should deploy it? Is auth_radius fine for everyone? Alternative would be acc_radius. To complete the cleanup of etc/ folder in the source code, I think the sip-router*.cfg files should me moved to examples/, at quick look they are invalid, using paths to old ser modules, no longer available at this moment (therefore I guess nobody is using them). Opinions and other suggestions are very welcome! Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

1 0

Re: [SR-Users] issue with TLS and 2 NIC interfaces
by Slava Bendersky 06 May '15

06 May '15

Hello Vik, I don't have same problem it works fine for me, but I have same setup on public side where multiply aliases on interface is involved. Slava. From: "Vik Killa" <vipkilla(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Monday, May 4, 2015 12:54:21 PM Subject: Re: [SR-Users] issue with TLS and 2 NIC interfaces Are you having the same problem as myself? Thanks, V On Fri, May 1, 2015 at 8:20 PM, Slava Bendersky < volga629(a)networklab.ca > wrote: Hello Vik, I have similar setup, try define you server configuration in tls.cfg [server:ip address of alias interface:port of alias interface] method = TLSv1 verify_certificate = no require_certificate = no private_key = certificate = ca_list = crl = Slava From: "Vik Killa" < vipkilla(a)gmail.com > To: "sr-users" < sr-users(a)lists.sip-router.org > Sent: Wednesday, April 29, 2015 4:32:21 PM Subject: [SR-Users] issue with TLS and 2 NIC interfaces Hello, if i have two interfaces (eth0 and eth0:0) I set kamailio to listen to the IP on eth0:0 This works great except when I try using TLS, kamailio routes traffic from eth0:0 to eth0 then to correct destination It should be just doing eth0:0 -> correct destination If I use UDP/TCP I am not seeing this behavior. If I set kamailio to listen to eth0 and remove eth0;0, TLS works fine. I'm running kamailio compiled from source git master a few days ago. Is anyone else having this problem? I'm wondering if it's a bug. Thanks, V _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

3 3

Strange behavior for TCP and UDP SIP Routing
by Grant Bagdasarian 06 May '15

06 May '15

Hello, I have a Kamailio machine configured with two interfaces. One public, one private. Public Address: 10.10.10.10 Private Address: 192.168.1.10 Kamailio forwards the SIP packet to another local endpoint: 192.168.1.20 When a client sends a SIP packet using TCP to the public address of Kamailio, Kamailio relays the packet using its private interface (private address) to the other endpoint. TCP: Client -> 10.10.10.10 -> 192.168.1.10 -> 192.18.1.20 When a client sends a SIP packet using UDP to the public address of Kamailio, Kamailio (or the OS) relays the packet using its public interface (public address) to the other endpoint. UDP: Client -> 10.10.10.10 -> 10.10.10.10 -> 192.18.1.20 In the case of TCP, the source address is 192.168.1.10. In the case of UDP, the source address is 10.10.10.10. Everything is the same in the routing script for both UDP and TCP, except in route[relay] (see below) Is this normal? What is causing this, the OS or Kamailio? Route[RELAY] { ... if($pr =~ "TCP") { if (!t_relay_to_tcp()) { sl_reply_error(); } } else { if (!t_relay_to_udp()) { sl_reply_error(); } } }

2 2

rtpengine, turn fallback
by Isuru Wijesinghe 06 May '15

06 May '15

Hi All I'm using rtpengine to in my implementation of webrtc. In my client side browser I have used following code snippet to configure stun/turn servers to identify ice candidates and turn fall back. >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> var pcConfig = {"iceServers": [ {'url': 'turn:52.6.57.126:3478?transport=tcp'}]}; peerConn = new PeerConnection(pcConfig); >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> Am using rfc 5766 turn server http://code.google.com/p/rfc5766-turn-server/ As long as stun is working my webrtc flow works fine. I wanted to know how rtpengine support the turn fallback, is there any configuration required in this case. Regards Isuru

2 1

Double record routes
by Mickael Marrache 06 May '15

06 May '15

Hi, We are adding TCP support to our load balancer and for some reason it adds two record route headers. The instance have two IP addresses on which it binds: one if the VIP address and the second is the non-VIP address. I explicitly set the tcp_source_ipv4 parameter with the VIP address so that it is used as source address for outbound TCP connection. So, we get the following INVITE going from the load balancer to a proxy: T 2015/05/05 12:08:49.715822 VIP:54667 -> PROXY:5060 [AP] INVITE sip:123@mycompany.com SIP/2.0. Record-Route: <sip:NONVIP;transport=tcp;r2=on;lr>. Record-Route: <sip:VIP;transport=tcp;r2=on;lr>. Via: SIP/2.0/TCP NONVIP;branch=z9hG4bK6f4.688efa90a17e02181ef7a11fecf8bb72.0;i=3. Via: SIP/2.0/TCP 1.1.1.1:4598;received=2.2.2.2;branch=z9hG4bKmqFaCxNo6m3f5LW4;rport=40020. You can see the INVITE is sent from the VIP address (as specified using the tcp_source_ipv4 parameter). However, the added Via corresponds to the non-VIP address. Also, you can see the two record route headers added for both addresses. Any idea? Thanks, Mickael

3 5

sht_rm_name_re() question
by Yufei Tao 05 May '15

05 May '15

Hi I've hit a problem with sht_rm_name_re() in htable module. I was calling it like this: sht_rm_name_re("Dlg=>$var(callid)::tenant"); But when I used sipp to generate 600 concurrent calls for example, I called this function when receiving BYE. But it removed more entries than it should. Seems when removing removing some entry with callid 1-3798(a)192.168.60.80, it also removed entries for [1-9]1-3798(a)192.168.60.80. How do I add '^' in front and '$' at the end of the regexp string when calling this function please? Here I don't really need regexp in fact, just want to do an exact match. But I didn't find any other functions for deleting entries. Thank you! Yufei

2 1

UAC not answer with new register after 401 reply from porvider
by Yuriy Gorlichenko 05 May '15

05 May '15

Hello. We have an issue with REGISTER to Provider. When Provider answers 401 Kamailio don't send any REGISTER with digest auth IP ourservice.com.5068 > provider.dev.5060: UDP, length 468 E...U...@.3' ..............AREGISTER sip:provider.dev SIP/2.0 Via: SIP/2.0/UDP ourservice.com:5068 ;branch=z9hG4bK4238.45a4a626000000000000000000000000.0 To: <sip:login@provider.dev> From: <sip:login@provider.dev>;tag=6a39af2190f8b2c99cc63bfeb3a3959c-1ff8 CSeq: 10 REGISTER Call-ID: 1b3f9f5d5822eca2-16804(a)ourservice.com Max-Forwards: 70 Content-Length: 0 User-Agent: kamailio (4.3.0-dev5 (x86_64/linux)) Contact: <sip:login@ourservice.com:5068> Expires: 360 IP provider.dev.5060 > ourservice.com.5068: UDP, length 572 E..XC...-.W..... ........DkJSIP/2.0 401 Unauthorized Via: SIP/2.0/UDP ourservice.com:5068 ;branch=z9hG4bK4238.45a4a626000000000000000000000000.0;received=23.100.24.250 From: <sip:login@provider.dev>;tag=6a39af2190f8b2c99cc63bfeb3a3959c-1ff8 To: <sip:login@provider.dev>;tag=as6ad61d4a Call-ID: 1b3f9f5d5822eca2-16804(a)ourservice.com CSeq: 10 REGISTER Server: FastTel SoftSwitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="provider.dev", nonce="63fc6951" Content-Length: 0

2 3

Kamailio proxy for far-end nat traversal
by rahul.ultimate 05 May '15

05 May '15

Hello I need a small guidance on creating a light weight proxy which only forwards the msgs to my sip server and also does supports symmetrical nated clients. The way I have created the configuration is a slight modification of : https://github.com/xlab1/sipfe_kamailio/blob/master/kamailio.cfg Problem is unless the sip clients use STUN media packets are not routed. And sometimes even the signalling does not pass through. Should i not use fix_nated_register and switch to fix _ nated_contact always ? Because in all those cases where signalling does not pass through, I see RTO. My best guess is kamailio trying to communicate with private ip. Anything which I can try that you could suggest from the top of yourmind woyld be muc appreciated. Thanks Rahul Sent from Samsung Mobile

2 1

Re: [SR-Users] issue with TLS and 2 NIC interfaces
by Slava Bendersky 04 May '15

04 May '15

Hello Vik, I have similar setup, try define you server configuration in tls.cfg [server:ip address of alias interface:port of alias interface] method = TLSv1 verify_certificate = no require_certificate = no private_key = certificate = ca_list = crl = Slava From: "Vik Killa" <vipkilla(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, April 29, 2015 4:32:21 PM Subject: [SR-Users] issue with TLS and 2 NIC interfaces Hello, if i have two interfaces (eth0 and eth0:0) I set kamailio to listen to the IP on eth0:0 This works great except when I try using TLS, kamailio routes traffic from eth0:0 to eth0 then to correct destination It should be just doing eth0:0 -> correct destination If I use UDP/TCP I am not seeing this behavior. If I set kamailio to listen to eth0 and remove eth0;0, TLS works fine. I'm running kamailio compiled from source git master a few days ago. Is anyone else having this problem? I'm wondering if it's a bug. Thanks, V _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

kamcli - kamailio command line control tool
by Daniel-Constantin Mierla 04 May '15

04 May '15

Hello, we just published a new command line tool for Kamailio management, named now kamcli, available at: * https://github.com/asipto/kamcli It is written in Python and adapted from a tool provided originally by a subcontractor for a specific deployment. While not being Python developers here, the benefits and flexibility proved extremely good. At this stage it is more an attempt to see if the community finds it useful and worth to get it developed further as part of Kamailio project. Short about the benefits/flexibility: - python is by default installed on OSes (including Mac OS X) (as opposite to Lua, for example) - became rather fast since Google invested in it - lot more knowledge base and developers out there (as opposite to Lua or Perl), along with tons of extensions - implementation is based on a CLI framework (Click) that makes it extremely easy to add new commands in a plugin-like fashion, therefore is easy to have custom commands when having specific needs (e.g., managing a custom database table used in kamailio.cfg via sqlops => add a new cmd_xyz.py in commands subfolder, without touching other files, not even the config) - help message is embedded in the command code file, making it easier to document commands - easy to validate parameters as well as format the output - handling kamcli configuration file options is also easy (ini format) One of my reasons of pushing this out was in the perspective of deprecating MI and working with JSONRPC fifo from command line -- I couldn't find easy ways to manage json docs with shell and common tools (e.g., cat, echo, grep, awk) only. Kamcli has implemented the JSONRPC over fifo command (see also jsonrpc-s kamailio module). Still, given the old initial implementation, at this moment, kamcli is relying on MI via fifo (mi_fifo) for some of the commands (e.g., dumping usrloc records from memory) -- (Just as side note: perhaps here we need to do a mapping, because name of commands are different, if we want to make a 'ctl transport agnostic' tool.) Here are few things to decide: - does it sound useful and are people finding it worth to develop it further? - what about the name, is it ok? Another one? If worth developing and deciding on the name, then the plan is to host the project on kamailio organization on github so each kamailio developer can code on it (it won't be inside kamailio source code tree, but a separate project inside same organization). So far, couple of commands were implemented when comparing with kamctl: - subscriber management (kamctl add/rm/passwd...) - user location management (kamctl ul ...) - database querying (kamctl db ...) - statistics (kamctl stats ...) - mi commands (kamctl mi ...) - jsonrpc commands (not in kamctl) While cross posted for the sake of getting the info to everyone in the community, let's have further discussions on sr-dev. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

2 1

TLSv1.2 and weak ciphers
by Attila Megyeri 04 May '15

04 May '15

Hi Daniel, Kamailio folks, We are trying to make our server more secure, but we have some issues. Right now, we have set the TLS method to method = TLSv1+ and cipher_list = HIGH The problem is, that there are still cipher suites offered which are not secure. E.g. If I check with the SSLLabs analizer, I see: This server supports anonymous (insecure) suites (see below for details). Grade set to F. Cipher Suites (SSL 3+ suites in server-preferred order; deprecated and SSL 2 suites always at the end) TLS_ECDHE_RSA_WITH_AES_256_GCM_SHA384 (0xc030) ECDH 256 bits (eq. 3072 bits RSA) FS 256 TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA384 (0xc028) ECDH 256 bits (eq. 3072 bits RSA) FS 256 TLS_ECDHE_RSA_WITH_AES_256_CBC_SHA (0xc014) ECDH 256 bits (eq. 3072 bits RSA) FS 256 TLS_DHE_RSA_WITH_AES_256_GCM_SHA384 (0x9f) DH 3072 bits (p: 384, g: 1, Ys: 384) FS 256 TLS_DHE_RSA_WITH_AES_256_CBC_SHA256 (0x6b) DH 3072 bits (p: 384, g: 1, Ys: 384) FS 256 TLS_DHE_RSA_WITH_AES_256_CBC_SHA (0x39) DH 3072 bits (p: 384, g: 1, Ys: 384) FS 256 TLS_DHE_RSA_WITH_CAMELLIA_256_CBC_SHA (0x88) DH 3072 bits (p: 384, g: 1, Ys: 384) FS 256 TLS_ECDH_anon_WITH_AES_256_CBC_SHA (0xc019) INSECURE 256 TLS_DH_anon_WITH_AES_256_GCM_SHA384 (0xa7) INSECURE 256 TLS_DH_anon_WITH_AES_256_CBC_SHA256 (0x6d) INSECURE 256 TLS_DH_anon_WITH_AES_256_CBC_SHA (0x3a) INSECURE 256 TLS_DH_anon_WITH_CAMELLIA_256_CBC_SHA (0x89) INSECURE 256 TLS_RSA_WITH_AES_256_GCM_SHA384 (0x9d) 256 TLS_RSA_WITH_AES_256_CBC_SHA256 (0x3d) 256 TLS_RSA_WITH_AES_256_CBC_SHA (0x35) 256 TLS_RSA_WITH_CAMELLIA_256_CBC_SHA (0x84) 256 TLS_ECDHE_RSA_WITH_3DES_EDE_CBC_SHA (0xc012) ECDH 256 bits (eq. 3072 bits RSA) FS 112 TLS_DHE_RSA_WITH_3DES_EDE_CBC_SHA (0x16) DH 3072 bits (p: 384, g: 1, Ys: 384) FS 112 TLS_ECDH_anon_WITH_3DES_EDE_CBC_SHA (0xc017) INSECURE 112 TLS_DH_anon_WITH_3DES_EDE_CBC_SHA (0x1b) INSECURE 112 TLS_RSA_WITH_3DES_EDE_CBC_SHA (0xa) 112 TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256 (0xc02f) ECDH 256 bits (eq. 3072 bits RSA) FS 128 TLS_ECDHE_RSA_WITH_AES_128_CBC_SHA256 (0xc027) ECDH 256 bits (eq. 3072 bits RSA) FS 128 TLS_ECDHE_RSA_WITH_AES_128_CBC_SHA (0xc013) ECDH 256 bits (eq. 3072 bits RSA) FS 128 TLS_DHE_RSA_WITH_AES_128_GCM_SHA256 (0x9e) DH 3072 bits (p: 384, g: 1, Ys: 384) FS 128 TLS_DHE_RSA_WITH_AES_128_CBC_SHA256 (0x67) DH 3072 bits (p: 384, g: 1, Ys: 384) FS 128 TLS_DHE_RSA_WITH_AES_128_CBC_SHA (0x33) DH 3072 bits (p: 384, g: 1, Ys: 384) FS 128 TLS_DHE_RSA_WITH_CAMELLIA_128_CBC_SHA (0x45) DH 3072 bits (p: 384, g: 1, Ys: 384) FS 128 TLS_ECDH_anon_WITH_AES_128_CBC_SHA (0xc018) INSECURE 128 TLS_DH_anon_WITH_AES_128_GCM_SHA256 (0xa6) INSECURE 128 TLS_DH_anon_WITH_AES_128_CBC_SHA256 (0x6c) INSECURE 128 TLS_DH_anon_WITH_AES_128_CBC_SHA (0x34) INSECURE 128 TLS_DH_anon_WITH_CAMELLIA_128_CBC_SHA (0x46) INSECURE 128 TLS_RSA_WITH_AES_128_GCM_SHA256 (0x9c) 128 TLS_RSA_WITH_AES_128_CBC_SHA256 (0x3c) 128 TLS_RSA_WITH_AES_128_CBC_SHA (0x2f) 128 TLS_RSA_WITH_CAMELLIA_128_CBC_SHA (0x41) 128 How can we get rid of these _anon_ cipher suites? Thanks Attila

1 1

Auth User & is_subscriber
by jay binks 04 May '15

04 May '15

Hey, from the examples it looks like the typical usage of is_subscriber is with $fu ( From User ), however im not 100% sure I want this. in-fact, I want it to be the Auth User that is used. After all if a call was Authed, then we authoritatively know what user it is, and I think this should be used over and above the from user ( which could even be set to caller Id in some cases ). Im having some trouble getting is_subscriber to work with the Auth user, does anyone have an example of this working somewhere ? -- Sincerely Jay

2 1

WebSocket data in TCP connection
by mayamatakeshi 04 May '15

04 May '15

Hello, about module websocket, is it possible to associate a value with the TCP connection and have this value readable when handling SIP requests on that connection?

1 4

Regd: Message Routing
by suryakiiran vanam 04 May '15

04 May '15

Hi Team, I am using Jitsi client with Kamailio. I was able to IM between the clients. How can I control the Message routing in Kamailio script. Please help me with any link. Regards, Surya.

2 1

WebRTC video calls
by Ivan Vujisic 04 May '15

04 May '15

I made successful audio calls from browser to browser using Asterisk 13.1 and SIPML5 browser phone. Asterisk can't manage WebRTC video calls due to lack of codec negotiation module, but I also faced RTP ports NAT traversal issue. To my understanding Kamailio is capable to resolve this. Can anybody confirm that he made successful browser to browser video calls with Kamailio sip proxy / registrar in front of Asterisk PBX. Also, any link to good tutorial or doc pages will be appreciated. Best Regards, Ivan Vujisic

2 1

UAC Module
by Ali Jibran 02 May '15

02 May '15

Hi all. I have this setup. Trunk--->Kamailio---->FreeSWITCH I have a trunk from a sip provided and registered successfully with the UAC module. Incoming is working fine. I need to make out going through kamailio too. I have it in the dialplan to forward the invite to kamailio from FreeSWITCH. I can see it the logs that it reaches kamailio. Now how do I make the call via the trunk? Basically this is what I'm trying to workout FS---->kamailio---->trunk. Any help will be much appreciated. Thanks. AJ

3 7

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users