- sr-users - kamailio.org

WSS configuration sample needed
by Sunil More 09 May '16

09 May '16

Hello All, I am trying to connect Kamailio over WSS and the error I get is "tls_accept: client did not present a certificate" . It works fine for WS, however Please guide me to connect over WSS. the following are the logs i See.. DEBUG: tls [tls_domain.c:703]: sr_ssl_ctx_info_callback(): SSL handshake started DEBUG: <core> [tcp_main.c:2430]: tcpconn_do_send(): sending... ay 9 11:07:01 P172 /usr/local/kamailio/sbin/kamailio[32025]: DEBUG: tls [tls_domain.c:715]: sr_ssl_ctx_info_callback(): SSL handshake done May 9 11:07:01 P172 /usr/local/kamailio/sbin/kamailio[32025]: DEBUG: tls [tls_domain.c:718]: sr_ssl_ctx_info_callback(): SSL disable renegotiation May 9 11:07:01 P172 /usr/local/kamailio/sbin/kamailio[32025]: DEBUG: tls [tls_server.c:411]: tls_accept(): TLS accept successful May 9 11:07:01 P172 /usr/local/kamailio/sbin/kamailio[32025]: DEBUG: tls [tls_server.c:418]: tls_accept(): tls_accept: new connection from sunilmore.in:49703 using TLSv1/SSLv3 AES256-SHA 256 May 9 11:07:01 P172 /usr/local/kamailio/sbin/kamailio[32025]: DEBUG: tls [tls_server.c:421]: tls_accept(): tls_accept: local socket: sunilmore.in:10082 May 9 11:07:01 P172 /usr/local/kamailio/sbin/kamailio[32025]: DEBUG: tls [tls_server.c:432]: tls_accept(): tls_accept: client did not present a certificate And the websocket closes. Please help. Thanking You, Sunil More Ph : 9503338275

2 5

sdp_remove_line_by_prefix
by Alberto Sagredo 09 May '16

09 May '16

Hi Im trying to remove my Asterisk s= line on SDP but when doing: sdp_remove_line_by_prefix("s=Asterisk"); Nothing happens. Loaded sdpops.so module as well. Is there any additional needed? BR

4 10

Getting sip.instance in INVITE
by Cibin Paul 09 May '16

09 May '16

Hi, How can I get sip.instance value in the sip header of INVITE? I do get this value during REGISTER Thanks & Regards Cibin

2 2

Third party registration help
by Code Developer 09 May '16

09 May '16

Hello, I'm after some help on third-party SIP registration in Kamailio - specifically, how to perform registration of user agents that don't register themselves, but only send out occasional messages. The UAs not registering themselves are devices on a GSM network that try to save band-width by not sending REGISTER requests for themselves. They only ever send out SIP messages when necessary, to which they expect a reply message from our processing server. (Note, details like external IP addresses have been changed for security.) The configuration is as follows. We have a SIP server running Kamailio (4.3, location db enabled, nathelper enabled) on a static external IP address (50.170.158.149), a processing server on an internal network registered with the SIP server (let's say with a User agent name of '1000'), and a monitoring device in another part of the country connected to the internet via a GSM network (let's say User agent name of '2000'). The device is effectively NAT'd on the GSM provider's private network. At some point the device (UA 2000) sends a SIP message to the processing server via Kamailio (to sip:1000@50.170.158.149). The processing server is registered, so the SIP server knows where it is and forwards the message accordingly. 200/OK replies propagate back successfully to the device. The message flow appear as below. Only relevant header fields are shown: Message received by the SIP server from remote device: ======== IP Src:60.250.149.198, Dst:50.170.158.149 UDP Src Port: 59177, Dst Port: 5060 -- Request-Line: MESSAGE sip:1000@50.170.158.149 Via: SIP/2.0/UDP 10.92.240.131:5060;branch=z9hG4bK... To: <sip:1000@50.170.158.149> From: <sip:2000@10.92.240.131>;tag=7a82c48a ======== Message forwarded from SIP server, received by processing server as shown: ======== IP Src:50.170.158.149, Dst: 192.168.0.235 UDP Src Port:5060, Dst Port:5060 -- Request-Line: MESSAGE sip:1000@50.170.158.146:5060; ob SIP/2.0 Via: SIP/2.0/UDP 50.170.158.149;branch=z9hG... Via: SIP/2.0/UDP 10.92.240.131:5060;rport=59177;received=60.250.149.198;branch=z9hG... To: <sip:1000@50.170.158.149> From: <sip:2000@10.92.240.131>;tag=7a82c48a ======== The processing server now needs to send back a reply message to the remote, unregistered device which is NATd on the GSM internal network. We can do this by sending back a reply to the external IP address and rport values found in the second Via header of the message received by the processing server (60.250.149.198, port 59177). The reply can only be sent from the SIP server or the GSM network's router will ignore it - it's only a temporary reply address/port open for a minute or so as it's effectively just a NAT route to the internet. In the first instance, when the processing server attempts to send back a message to the remote device via the SIP server, the SIP server replies with a 404 as the remote device hasn't registered and the SIP server has no idea where to send it. After looking at RFC3261 my conclusion was that we must get the processing server to perform a third-party registration on behalf of the remote device before we can send a reply message back to that device. My understanding is that to perform the third party registration I should send a REGISTER request from our processing server to the SIP server: 1) Set the 'From' field to the AOR of the UA performing the registration (which is the processing server, 1000). 2) Set the 'To' field to the AOR of the UA being registered (i.e. the remote device, 2000). 3) Add a 'Contact' header field set to the IP address and port of the remote device (from Via = 60.250.149.198 port 59177). The message flow for the register is as follows (only relevant header fields shown): Processing server sends a register request to the SIP Server: ======== Request-Line: REGISTER sip:50.170.158.149 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.235:5060;rport;branch=z9hG... From: <sip:1000@50.170.158.149>;tag=e6e2d70... To: <sip:2000@50.170.158.149> Contact: <sip:2000@60.250.149.198:59177> ======== SIP server responds to the processing server with a 401 unauthorized response containing the auth header field (nonce challenge): ======== WWW-Authenticate: Digest realm="50.170.158.149", nonce="VytkJ1crIvu4ekGp/M4mnRlYmJRciF3N" ======== Processing server sends a revised register request to the SIP Server, augmented with the authorisation header for the supplied nonce: ======== Request-Line: REGISTER sip:50.170.158.149 SIP/2.0 Via: SIP/2.0/UDP 50.170.158.146:1096; rport;branch=z9hG4... From: <sip:1000@50.170.158.149>;tag=dae68a... To: <sip:2000@50.170.158.149> Contact: <sip:2000@60.250.149.198:59177> Authorization: Digest username="1000", realm="50.170.158.149", nonce="VytkJ1crIvu4ekGp/M4mnRlYmJRciF3N" response="d4d7ffba8..." ======== However, the SIP Server again responds with a 401 and won't perform the third-party registration. I haven't been able to determine why. I *think* the response hash the processing server is sending back is correct. I'm using PJSIP on our processing server to perform all SIP functionality. The processing server does obviously register itself, the difference being that the To and From fields are the same (<sip:1000@50.170.158.149>) and there is no Contact header. I'm at the point now of being unable to find any information or examples on how third-party registration requests should be performed - apart from RFC3261, which doesn't really describe the process very well. Can anyone tell me why the third-party registration would fail, despite replying with the authorisation header? Does Kamailio first need to be configured to allow third-party reg somehow? Alternatively, should I be attempting to send back SIP messages to unregistered devices in some other fashion? I hope someone can help or point me in the right direction. As an aside, when the device (UA2000) is plugged into a network via ethernet, it does then perform its own registration and everything works - messages go back and forth without issue.

2 1

LCR Module Question
by Alberto Sagredo 09 May '16

09 May '16

Hi I have it working but i have re-read documentation and do not see how to do what i need. I explain it :) Now i have only one LCR provider and i need to add a backup one. I do not know if its enough to add under same lcr_id or its better to add with different one and add several lcr_rule and lcr_rule_target Acordding to next_gw() function, what is better to be used? In my case. a primary providers is used and only a backup one for the momment. Thanks for your help Alberto

3 3

Media-Proxy & NAT
by cao tao 09 May '16

09 May '16

Hello: I use kamailio as my SIP signal proxy, and media-proxy as RTP relayer. It works well at many situations. But when it comes to two layer of NAT(a user behind two layer of NAT), I got a single way media(the sip signal layer was ok). I can’t hear the voice of user who behinds two layer of NAT, but he can hear the voice of mine. Most part of my kamailio config is default, and simply enable media-proxy by engage_media_proxy()： if (is_method("INVITE") && !has_totag()) { engage_media_proxy(); } And in my UAC/UAS side, I use PJSIP. Can anybody give me some help? Any suggestion will be appreciated!

2 1

User offline
by 王欣鑫 06 May '16

06 May '16

Hello, I use kamailio set up an IMS on a cloud server in a public network, and have alice and bob registered on the server using ImsDroid in a private network. When I use alice to call bob, I got "User offline" error. Below are .pcap file captured from 4060 port and pcscf.log/icscf.log/scscf.log and all config files(in kamailio.zip) 4060.pcap <http://sip-router.1086192.n5.nabble.com/file/n148166/4060.pcap> pcscf.log <http://sip-router.1086192.n5.nabble.com/file/n148166/pcscf.log> icscf.log <http://sip-router.1086192.n5.nabble.com/file/n148166/icscf.log> scscf.log <http://sip-router.1086192.n5.nabble.com/file/n148166/scscf.log> kamailio.zip <http://sip-router.1086192.n5.nabble.com/file/n148166/kamailio.zip> My kamailio version and debian operating system version is like below: root@iZ94lg22es7Z:~# kamailio -version version: kamailio 4.4.0 (x86_64/linux) d4f23c flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: d4f23c compiled on 13:02:46 May 5 2016 with gcc 4.9.2 root@iZ94lg22es7Z:~# root@iZ94lg22es7Z:~# cat /etc/issue Debian GNU/Linux 8 \n \l root@iZ94lg22es7Z:~# Can anybody give me some help? Any suggestion will be appreciated! -- View this message in context: http://sip-router.1086192.n5.nabble.com/User-offline-tp148166.html Sent from the Users mailing list archive at Nabble.com.

2 14

Clients are busy
by kai.ohnacker＠cbc.de 06 May '16

06 May '16

Hello Community, in my scenario works multiple user with the kamailio server and uses different clients. They work in a mobile scenario and setup the clients and break the clients in a high period. If they break the clients, sometimes they don't send a bye, but they take off the cable connection during a call. If these devices will connected again and other clients try to call them, they are busy. That is my problem :). Does the Kamailio server provides any function to send a bye message if a clients don't register at a required time? Or is there another solution available? Thanks for help. Cheers, Kai

2 1

making rtpengine gives error on centos
by Yasin CANER 05 May '16

05 May '16

4 5

Kamailio - Sip_capture Module
by Ali Taher 05 May '16

05 May '16

Hello, We are currently using kamailio sip_capture module, and then we need to create a Wireshark file (pcap format)from fields in the database. As we noticed in sip_capture table there is msg fields that contains only the sip content. And in order to create our file we need to have the entire packet and not part of it. We are hopefully asking if you can give us a hand with that , beside the Homer project, what we actually wanted is to know if there is another module in kamailio that store the entire packet captured and not only the part of it. Regards, Ali Taher

2 3

Pause a Transaction for a short while
by Sirvan Paraste 05 May '16

05 May '16

Hello all, is there anyway to pause a transaction for a short while (1 second) ? I have implemented the push notification, when the callee get register I need a short time to be sure that the app is completely start and can do its duty. otherwise it missed the call. so I want to pause the transaction. I have tested async_sleep("1"); it worked. is there any better solution ? $var(hjoin) = 0; lock("$tU"); async_sleep("1");

2 1

ACK not forwarded by Proxy when LOCATION lookup
by Al S 05 May '16

05 May '16

Hi, I am trying to simulate Inbound calls from UAC to Kamailio to UAS. Kamailio performs location lookup within route[LOCATION] and updates URI with the address of UAS. However, it does not add a new Record-Route. Therefore, ACK message sent by UAC is not routed by the proxy: UAC ---> (send INVITE) ---> Kamailio ---> (send INVITE) ---> UASUAS ---> (send 180) ---> Kamailio ---> (send 180) ---> UACUAS ---> (send 200) ---> Kamailio ---> (send 200) ---> UAC UAC ---> (send ACK) ---> Kamailio ???? ACK message has "Route" header with the address of Proxy. Thanks,AI S

2 1

You must register first with a S-CSCF/Server error on LIR select next S-CSCF
by 王欣鑫 05 May '16

05 May '16

Hello, I'm using kamailio and OpenImsCore HSS module to set up an IMS enviroment(P/I/S-CSCF and HSS). All these four modules are set up in one cloud server in public network with different ports(4060/4061 5060 6060 3869 3870) to commonicate with each other. Now I have two users Alice and Bob, they are in private network. I can register Alice and Bob with ImsDroid on this IMS, however when I use Alice to call Bob, I can see "You must register first with a S-CSCF" error. As I found this was caused by kamailio.cfg in pcscf module, I comment out these lines below. Then I got another error "Server error on LIR select next S-CSCF". And I captured packets on port 5060, I can see several "Request-Line: OPTIONS sip:sbc-2.ng-voice.com SIP/2.0" and "Request-Line: OPTIONS sip:sbc-1.ng-voice.com SIP/2.0". #if (!pcscf_is_registered("location")) { # send_reply("403","Forbidden - You must register first with a S-CSCF"); # break; #} What's wrong with my IMS? Could any body give me some help? Thank you! Here are my IMS configure files. kamailio.zip <http://sip-router.1086192.n5.nabble.com/file/n148131/kamailio.zip> Here are my hss configure files. hss_config.zip <http://sip-router.1086192.n5.nabble.com/file/n148131/hss_config.zip> -- View this message in context: http://sip-router.1086192.n5.nabble.com/You-must-register-first-with-a-S-CS… Sent from the Users mailing list archive at Nabble.com.

1 0

Using websocket server for events
by Yuriy Gorlichenko 04 May '16

04 May '16

Hi is there is a valy to send websocket events (not only SIP) on remote subscribed client from config file? I just want make shure that i need to use redis for this staff insead of native websocket events

2 3

Using Kamailio as UAC on any PBX
by Jöran Vinzens 04 May '16

04 May '16

Hi All, we are trying to use the Kamailio as an UAC on a PBX. It should behave just like a normal SIP UAC (e.g. an SNOM phone). The reason why we want that, is connecting a service to any other PBX with Client features and security (DA). Our setup looks like: Kamailio UAC w/ register <-> Kamailio OBP/SBC <-> PBX The register is working fine. We insert the credentials into DB and trigger kamailio to reload the config (kamctl uac.reload). Then the register is sent out including the unauth (401) handling. After than we are registered on the PBX. The Register is sent though the OBP as configured in DB. We are trying to make calls: Calls from PBX to Kamailio are working fine but we struggle with calls from kamailio torwards PBX. We have the l_uuid in a variable and we try to get the PV ready for the invite. First we need to do is rewrite the "from" URI. We thought we can get a proper FROM URI with the uuid we have. But the uac_replace_from just sets the $fu as given and does not make any logic with UAC credentials. To handle the RURI, we trigger uac_reg_lookup with the uuid and it replaces the $ru with user-name and domain from the data stored in DB. In this case that makes no sense. The $rU should be what the user had dialed, the $rd should be the server given in uacreg DB table. The OBP given in DB is ignored completely. After we managed to rewrite everything in SIP as needed we try to catch the 407 as it's shown in the module documentation with failure route. Here we have not found any way to get the credentials for the uuid. Now we wonder if we have missed any global setting which sets everything right. If anybody has a similar setup and a working config, we appreciate any help we can get. our config looks like: ... loadmodule "uac.so" .... modparam("uac","auth_realm_avp","$avp(arealm)") modparam("uac","auth_username_avp","$avp(auser)") modparam("uac","auth_password_avp","$avp(apass)") modparam("uac","restore_from_avp","$avp(original_uri_from)") modparam("uac","restore_to_avp","$avp(original_uri_to)") modparam("uac","credential","username:domain:password") modparam("uac", "reg_db_url","DBURI") modparam("uac", "reg_timer_interval", 60) modparam("uac", "reg_retry_interval", 300) modparam("uac", "reg_contact_addr", "1.2.3.4:5060") ... route { xlog("L_NOTICE", "INVITE: F=$fu T=$tu R=$ru\n"); route("QUERY_UUID"); # <- sets $var(UUID) to l_uuid of the DB $var(request_user) = $rU; uac_reg_lookup("$var(UUID)", "$ru"); $var(request_from_user) = $rU; $rU = $var(request_user); uac_replace_from("$var(request_from_user)","sip:$var(request_from_user)@ 1.2.3.4"); t_on_failure("TRUNKAUTH"); xlog("L_NOTICE", "INVITE: F=$fu T=$tu R=$ru $avp(auser) $avp(apass)\n"); # avp "auser and apass are "<null>" if (!t_relay()) { xlog("L_ERR", "error to <$tu> from <$fu>\n"); sl_reply_error(); }; } failure_route[TRUNKAUTH] { if (t_is_canceled()) { exit; } if(t_check_status("401|407")) { # $avp(auser) = "test"; <-should be set from the l_uuid profile anyhow # $avp(apass) = "test"; $var(request_user) = $rU; if (uac_auth()) { # We do never get here since uac_auth never returns true. $rU = $var(request_user); route("FORWARD_MESSAGE"); exit; } xlog("L_NOTICE", "DAMN!!!"); } } -- Jöran Vinzens - vinzens[at]sipgate.de <vinzens(a)sipgate.de> sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391 www.sipgate.de - www.sipgate.co.uk

2 1

rtpengine default set
by Alex Balashov 04 May '16

04 May '16

Hi, After upgrading to Kamailio v4.4, I receive the following error on boot: May 03 17:18:22 allegro-4.evaristesys.com /usr/local/sbin/kamailio[24972]: ERROR: rtpengine [rtpengine.c:2691]: select_rtpp_set(): script error-invalid id_set to be selected May 03 17:18:22 allegro-4.evaristesys.com /usr/local/sbin/kamailio[24972]: ERROR: rtpengine [rtpengine.c:2691]: select_rtpp_set(): script error-invalid id_set to be selected May 03 17:18:22 allegro-4.evaristesys.com /usr/local/sbin/kamailio[24972]: NOTICE: rtpengine [rtpengine.c:1788]: mod_init(): Default rtpp set 0 NOT found My rtpengine module config is: --- #!ifdef WITH_RTPENGINE #!subst "/RTPENGINE_SOCK/udp:127.0.0.1:5050/" modparam("rtpengine", "rtpengine_sock", "1 == RTPENGINE_SOCK") modparam("rtpengine", "rtpengine_disable_tout", 30) modparam("rtpengine", "rtpengine_tout_ms", 2000) modparam("rtpengine", "rtpengine_retr", 2) #!endif --- Furthermore, I have set_rtpengine_set("1") before every call to the rtpengine_* module function(s) in my route script. Subsequently to that, Kamailio starts normally and connects to rtpengine: May 03 17:18:22 allegro-4.evaristesys.com /usr/local/sbin/kamailio[24972]: INFO: rtpengine [rtpengine.c:2542]: rtpp_test(): rtp proxy <udp:127.0.0.1:5050> found, support for it enabled May 03 17:18:22 allegro-4.evaristesys.com /usr/local/sbin/kamailio[24975]: INFO: rtpengine [rtpengine.c:2542]: rtpp_test(): rtp proxy <udp:127.0.0.1:5050> found, support for it enabled May 03 17:18:22 allegro-4.evaristesys.com /usr/local/sbin/kamailio[24974]: INFO: rtpengine [rtpengine.c:2542]: rtpp_test(): rtp proxy <udp:127.0.0.1:5050> found, support for it enabled [...] It might be something on my end, but my guess is that in v4.4 there is a fixup that runs beforehand and looks for a default set. Strictly speaking, I don't need an explicitly identified set, since I'm only using one rtpengine instance. However, I explicitly specify it because I'd like the code to be open in the future to selecting a given rtpengine relay based on some criteria. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 2

create packet
by zeinab 04 May '16

04 May '16

Hello, My name is Zeinab and I am using kamailio to capture sip packets. What I really want to do is to create a .pcap file so that I can open it in Wireshark. As I saw, that you are only saving the sip content in database. And to create the packet I need the entire frame. So I think there is something missing from my side. Can you help me with that? You can see in the attachment the field needed from wireshark And in sip_capture I found only this: If there is some manual were I can find the entire packets please let me know Thank you for your time and looking forward to hear from you. Best Regards, Zeinab

2 1

create PCAP file
by zeinab 04 May '16

04 May '16

Hello, My name is Zeinab and I am using kamailio to capture sip packets. What I really want to do is to create a .pcap file so that I can open it in Wireshark. As I saw, that you are only saving the sip content in database. And to create the packet I need the entire frame. So I think there is something missing from my side. Can you help me with that? You can see in the attachment the field needed from wireshark And in sip_capture I found only this: If there is some manual were I can find the entire packets please let me know Thank you for your time and looking forward to hear from you. Best Regards, Zeinab

1 0

Kamailio 5-0 - Executing event routes in embedded languages
by Daniel-Constantin Mierla 04 May '16

04 May '16

Hello, as announced earlier this morning, most of the routing blocks can be written directly in an embedded language (right now supported being Lua and Python). The missing blocks are (most of) the event routes. The event route from tm for branch failure route is already supported, as the name has to be set via t_on_branch_failure(...). In the native configuration, event routes are executed if they are defined. Their names (as in event_route[name]) are not valid function names for the other programming languages to be used directly corresponding callback function name. I was thinking of the alternative, next are the ones I came up with, more suggestions are welcome. 1) add parameters for event route callback names, like: modparam("htable", "evrtcb_mod_init", "ksr_htable_mod_init") would define the name of the function for the alternative to event_route[htable:mod-init]. This has a benefit in speed, if the parameter is not defined, then there is no overhead in trying to execute a callback function in an embedded language. The drawback is that there is need to code a bit of C to add those parameters to the components executing event routes. 2) do a direct mapping using some translation rules, like: - prefix with ksr_ - replace ':' and '-' with underscore '_' So event_route[htable:mod-init] will have automatically the correspondent function ksr_htable_mod_init(). The drawback is that each time the even route should be called, a lookup on embedded language symbols table needs to be done to see if it exists, then execute it. There is still some C code to do, but less than adding parameters like for 1). 3) like 2), but instead of an automatic translation, keep a mapping table: struct mapping { str kname; str ename; } table[] = { str_init("htable:mod-init"), str_init("ksr_htable_mod_init"), } Adding a new event route will require adding a new record in the mapping table. The benefit is that the function name in embedded language can be selected to be whatever, with or without ties to the event_route name. Opinions? Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 0

A question about the code
by ngregory＠reprivata.com 04 May '16

04 May '16

3 4

Kamailio 5.0 - Routing blocks in Lua or Python
by Daniel-Constantin Mierla 04 May '16

04 May '16

Hello, I want to announce that the framework (named kemi) which allows writing the routing blocks in an embedded language is already in place. Lua or Python can be use as alternative to the native scripting language to write the routing blocks. More details can be found at: - https://www.kamailio.org/wiki/devel/config-engines Example of scripts that offer more or less same features as etc/kamailio-basic.cfg are available in source tree at examples/kemi. For convenience of reading, here are the links to git repository: - the common file setting global and modules parameters as well as specifying the config engine: https://github.com/kamailio/kamailio/blob/master/examples/kemi/kamailio-bas… - the part of config with routing blocks in native language: https://github.com/kamailio/kamailio/blob/master/examples/kemi/kamailio-bas… - the part of config with routing blocks in Lua: https://github.com/kamailio/kamailio/blob/master/examples/kemi/kamailio-bas… - the part of config with routing blocks in Python: https://github.com/kamailio/kamailio/blob/master/examples/kemi/kamailio-bas… You can compare the last three above and see how different or similar they look for providing same SIP routing features using different languages. A lot of modules need to export now functions to embedded config interface, which is quite straightforward: https://www.kamailio.org/wiki/devel/config-engines#exporting_functions_to_k… The main missing part is dealing with event_route blocks -- I will send a separate email to discuss that. Interesting that some incipient performance tests to compare the interpreters show good results for using especially Lua but also Python: - https://www.kamailio.org/wiki/devel/config-engines#interpreters_performances Anyone willing to help can start exporting functions from modules to embedded interface or test some more complex configs in Lua or Python. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

1 0

t_on_branch_failure never called AND t_relay never fails in a, forked scenario
by Tristan Mahé 03 May '16

03 May '16

Hi fellow kamailio users, I'm playing actually with t_on_branch_failure, in a simultaneous forking scenario ( lookup("location") returns multiple contacts ). Any idea on how to get those "on_branch_failure" working ? Thanks in Advance ! * Test by putting failure_exec_mode: - at 1 => no branch failure being called, never. - at 0 => t_relay does not fail if a single branch answers the transaction, even if all other branches are failing. * Test by moving the t_on_branch_failure("manage_branch_failure"); line before or after the lookup("location") or on a _on_branch route to set it per branch. - The on_route branch are executed, but never the failure one. - tried also with failure_exec_mode = 0 or 1. * Using set_forward_no_connect, I can see in the logs a "Sending to branch X failed" but no failure route being executed. Without it, nothing in the logs indicating that the branch has timeout. * Config extract: # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # 3 by default # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) modparam("tm", "reparse_on_dns_failover", 0) # 1: Allow branches to have a per branch failure route # 0: t_relay should fail if a branch fail modparam("tm", "failure_exec_mode", 1) # Allow locally generated 408 to be forgotten in case a reply is received. #modparam("tm", "faked_reply_prio", 2000) # USER location service route[LOCATION] { t_on_branch_failure("manage_branch_failure"); if (!lookup("location")) { .... } # Let's get back after 2.5s if no 100/Trying received. t_set_fr(0, 2500); # set_forward_no_connect(); #t_on_branch("SET_BRANCH_FAILURE"); route(RELAY); exit; } route[RELAY] { .... if (!t_relay()) { xlog("L_WARN","$rm|$ci|$fu|$tu|RELAY_FAILED"); sl_reply_error(); } exit(); } # set failure route per branch branch_route[SET_BRANCH_FAILURE] { xlog("L_WARN","$rm|$ci|$fu|$tu|SETTING_BrANCH_FAILURE_ROUTE"); t_on_branch_failure("manage_branch_failure"); } # manage failure routing cases per branch event_route[tm:branch-failure:manage_branch_failure] { xlog("L_WARN","$rm|$ci|$fu|$tu[$avp(dest_to)]|MANAGE_BRANCH_FAILURE"); }

3 2

set_body_multipart and memory leak
by Sergey Basov 03 May '16

03 May '16

Hi, All. I have some problems with pkg_memory leak. After each call used private memory of the "udp receiver" process increases at 2500 bytes. On test environment I hav set it to 2Mbyte, but it is full after 700 calls. I need to modify SDP in invite and other packets. When I do ==================================================== if(has_body("application/sdp")) set_body_multipart(); if (msg_apply_changes()) { xlog("L_INFO", "ISUP 1 Changes Applied Succesfully"); } } record_route(); dlg_manage route(RELAY); ==================================================== Debug from kamailio: Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: textops [textops.c:2627]: has_body_f(): content type is 196611 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: textops [textops.c:1577]: set_multibody_helper(): delimiter<17>:[unique-boundary-1] Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: textops [textops.c:1486]: generate_boundary(): adding final CRLF+CRLF Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: textops [textops.c:1714]: set_multibody_helper(): content-type<44>:[multipart/mixed;boundary="unique-boundary-1"] Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: textops [textops.c:1766]: set_multibody_helper(): set flag FL_BODY_MULTIPART Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: ERROR: <core> [msg_translator.c:1691]: get_boundary(): Content-Type hdr has no params Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: WARNING: <core> [msg_translator.c:1954]: build_req_buf_from_sip_req(): check_boundaries error Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [msg_translator.c:422]: clen_builder(): content-length: 251 (251) Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: textopsx [textopsx.c:231]: msg_apply_changes_f(): SIP message content updated - reparsing Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): SIP Request: Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): method: <INVITE> Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): uri: <sip:0637304494@10.56.42.27:5070;user=phone> Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:614]: parse_msg(): version: <SIP/2.0> Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK09e6.2654d2382f7c4e4fb8bb62fe85f62dfa.0>; state=6 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 236, <i> = <1>; state=16 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: INFO: <script>: ISUP 1 Changes Applied Succesfully Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKsr-s7wTDLa0zUfsTql5zpl0zpl0z.lRD.zok.TLz.lSk2Fw1RY9PJYMxpdED.dED.aEG38LkpcsWhqMQr1v1bFAyJGCxrCoQdB71Mtceh52GpZKDpX3zRFAzVFJzhBLGpeJe.TczW7cDuT8GLs7O37vBbn6BbTfGhM2Gpl*>; state=16 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=ffffffffffffffff Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:509]: parse_headers(): parse_headers: this is the second via Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=10 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [46]; uri=[sip:0637304494@10.56.42.27:5070;user=phone] Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [<sip:0637304494@10.56.42.27:5070;user=phone>#015#012] Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <2> <INVITE> Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=251 Apr 21 15:38:20 sip1 /usr/sbin/kamailio-sipt[17775]: DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

2 16

xmlrpc crash
by Juha Heinanen 03 May '16

03 May '16

i was sending some xmlrpc requests to kamailio and got crash. -- juha (gdb) where #0 0x00007f9c465da634 in __GI___libc_free (mem=0x17828c0) at malloc.c:2945 #1 0x00007f9c39e0b797 in collect_garbage () at xmlrpc.c:741 #2 0x00007f9c39e12669 in dispatch_rpc (msg=0x0, s1=0x17828b0 "ams", s2=0x7f9c46903628 <main_arena+8> "`\352w\001") at xmlrpc.c:2440 #3 0x000000000053c2c0 in do_action (h=0x7ffd17c562d0, a=0x7f9c46355c28, msg=0x7ffd17c563b0) at action.c:1060 #4 0x000000000053ad85 in run_actions (h=0x7f9c46903620 <main_arena>, a=0x17828b0, msg=0x7f9c46903628 <main_arena+8>) at action.c:1549 #5 0x00007f9c39e0d04b in em_receive_request (orig_msg=0x7f9c45cc43e8, new_buf=0x0, new_len=24651952) at xmlrpc.c:2303 #6 0x00007f9c39e14034 in process_xmlrpc (msg=0x7f9c45cc43e8) at xmlrpc.c:2385 #7 0x00000000004fdce3 in nonsip_msg_run_hooks (msg=0x7f9c45cc43e8) at nonsip_hooks.c:111 #8 0x0000000000549610 in receive_msg (buf=0x0, len=1171014632, rcv_info=0x7f9c3e4b0110) at receive.c:188 #9 0x00000000005cb97a in tcp_read_req (con=0x7f9c3e4b00f8, bytes_read=0x7ffd17c56e64, read_flags=0x7ffd17c56e68) at tcp_read.c:1418 #10 0x00000000005d0293 in handle_io (fm=0x7f9c3e4b00f8, events=10416, idx=-1) at tcp_read.c:1611 #11 0x00000000005d6bce in io_wait_loop_epoll (h=<optimized out>, t=<optimized out>, repeat=<optimized out>) at io_wait.h:1061 #12 tcp_receive_loop (unix_sock=1183856160) at tcp_read.c:1781 #13 0x00000000004d9df7 in tcp_init_children () at tcp_main.c:4788 #14 0x000000000050736f in main_loop () at main.c:1690 #15 0x000000000041cdbc in main (argc=0, argv=0x0) at main.c:2616

2 15

Kamailio Client-Server
by Shiv Patidar 02 May '16

02 May '16

hiii 1.We are writing our own SIP Client to test my Client i am using Kamailio server 2. So i am trying to map the identities used between Client and Server 3. If can possible can u please help me to the steps has to be follower to set identities common between Client and Server. 4. Or if u have some test code for client can u share the SIP message dump for test code so i can re-use the same for configuration my client code. Thank you

3 2

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