- sr-users - kamailio.org

Re: [SR-Users] help with NOTIFY
by Slava Bendersky 20 Oct '16

20 Oct '16

Hello Everyone, This message as continue conversation from http://lists.sip-router.org/pipermail/sr-users/2015-March/087557.html. That my previous post about it. I never was be be able forward NOTIFY from asterisk to client through kamailio. Right now in use asterisk 14 pjsip. Any help thank you. volga629 From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 19 October, 2016 18:36:52 Subject: [SR-Users] help with NOTIFY Hello Everyone, I am trying forward NOTIFY to client. My setup is PBX Server ---- local lan ---kamailio ---- internet ---- client The problem with whole setup that kamailio is not forwarding NOTIFY to client. Here are log. U 2016/10/19 14:23:29.581609 10.18.130.50:5160 -> 10.18.130.46:5060 NOTIFY sip:10101@10.18.130.46:5060 SIP/2.0. Via: SIP/2.0/UDP 10.18.130.50:5160;rport;branch=z9hG4bKPjONocd4FvI5LkOmSR6ZCA5Z1f7SYDb3Ty. From: <sip:8101@networklab.ca>;tag=ziApds23FQsPL2zd6q8pCUlwW.jwa6mK. To: "Slava" <sip:10101@networklab.ca>;tag=734f813f. Contact: <sip:10.18.130.50:5160>. Call-ID: A0bJWg8v0nGmtUmiHzUixQ... CSeq: 18547 NOTIFY. Route: <sip:kamailio_local_ip:5060;lr;r2=on;ftag=734f813f>. Route: <sip:kamailio_public_ip:5081;transport=tls;lr;r2=on;ftag=734f813f>. Event: presence. Subscription-State: active;expires=413. Allow-Events: message-summary, presence, dialog, refer. Max-Forwards: 70. User-Agent: FPBX-13.0.188.9(14.0.2). Content-Type: application/pidf+xml. Content-Length: 579. . <?xml version="1.0" encoding="UTF-8"?> <presence entity="sip:8101@10.18.130.50:5160" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:pp="urn:ietf:params:xml:ns:pidf:person" xmlns:es="urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status" xmlns:ep="urn:ietf:params:xml:ns:pidf:rpid:rpid-person"> <note>On the phone</note> <tuple id="8101"> <status> <basic>closed</basic> </status> <contact priority="1">"Slava" <sip:10101@networklab.ca></contact> </tuple> <pp:person> <status> <ep:activities>ep:busy</ep:activities> </status> </pp:person> </presence> U 2016/10/19 14:23:32.045388 10.18.130.46:5060 -> 10.18.130.50:5160 SIP/2.0 408 Request Timeout. Via: SIP/2.0/UDP 10.18.130.50:5160;rport=5160;branch=z9hG4bKPju05rMNlzhfukLvaq4ELmq.QI4GBqDN-m;received=10.18.130.50. From: <sip:8101@networklab.ca>;tag=Yzy2m6O2AFT6rGSDBmo6PjcGaCWT1i5H. To: "Slava" <sip:10101@networklab.ca>;tag=ad5a945e. Call-ID: 2pq5iebmtLdB_OsMUs3EOg... CSeq: 14623 NOTIFY. Server: kamailio (4.3.4 (x86_64/linux)). Content-Length: 0. Any help thank you. volga629 _______________________________________________ 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

1 0

sipml5 fail invite kamailio ims web sockets
by Serhat Guler 20 Oct '16

20 Oct '16

Dear all, I can register the sipml5 client to my kamailio IMS setup successfully, but when I try to make a call the invite fails with "478 Unresolvable destination". From the wireshark trace file I think the PCSCF on the terminating side cannot resolve the destination. After the invite request fails the caller re-registers to the networkj automatically. The wireshark file is attached. The registration process happens successfully but all of the nodes give some errors: 10(3773) INFO: <script>: [REGISTER] from [sip:alice@net1.test] to [sip:alice@net1.test] 10(3773) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=425 a=25 n=xlog 10(3773) ERROR: <script>: REGISTER (sip:alice@net1.test (192.168.0.10:14115) to sip:alice@net1.test, 25c80750-d52d-30c8-d373-c0136401a927) 10(3773) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=437 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=437 a=5 n=route 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=627 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=622 a=25 n=mf_process_maxfwd_header 10(3773) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <59840> <REGISTER> 10(3773) DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 10(3773) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=633 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=627 a=26 n=sanity_check 10(3773) DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header 10(3773) DEBUG: sanity [mod_sanity.c:248]: w_sanity_check(): sanity checks result: 1 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=639 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=633 a=24 n=has_totag 10(3773) DEBUG: siputils [checks.c:97]: has_totag(): no totag 10(3773) DEBUG: <core> [select.c:412]: run_select(): Calling SELECT 0xb6e5fe40 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=645 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=639 a=25 n=is_method 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=649 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=645 a=24 n=t_lookup_request 10(3773) DEBUG: tm [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=1 T start=0xffffffff 10(3773) DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=21728, isACK=0 10(3773) DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed 10(3773) DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found 10(3773) DEBUG: tm [t_lookup.c:1080]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=2 T end=(nil) 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=652 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=649 a=25 n=is_method 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=650 a=26 n=send_reply 10(3773) DEBUG: sl [sl.c:280]: send_reply(): reply in stateless mode (sl) 10(3773) DEBUG: websocket [ws_conn.c:429]: wsconn_get(): wsconn_get for id [2] 10(3773) DEBUG: websocket [ws_conn.c:437]: wsconn_get(): wsconn_get returns wsc [0xb2329228] refcnt [2] 10(3773) DEBUG: websocket [ws_frame.c:751]: ws_frame_transmit(): Tx message: The SCSCF says (while inviting): 0(3582) ERROR: dialog_ng [dlg_handlers.c:196]: populate_leg_info(): bad sip message or missing Contact hdr 0(3582) ERROR: dialog_ng [dlg_handlers.c:453]: dlg_onreply(): could not add further info to the dlg out Making calls between 2 IMS clients(boghe) is possible. Would appreciate any help (I can share the cfg files if needed). Cheers, Serhat

2 3

Kamailio script config - Generate Documentation
by José Seabra 20 Oct '16

20 Oct '16

Hello there, I'm sending this email in order to know if anyone had work or has any idea how we can generate Documentation from kamailio script using Comments? This would be great because we could write comments explaining the Route method during script development and use it to generate documentation about our kamailio script, saving time and leave our deployment documented. Thank you -- Best Regards José Seabra

3 5

Regarding pua_reginfo, server_address
by Ram Anji 20 Oct '16

20 Oct '16

Hi All, I am configured kamailio to handle reg info event subscriptions, My server is running if i configure the following parameter. #kamailio local ip is 192.168.20.20 *modparam("pua_reginfo", "server_address", "sip:reginfo@192.168.20.20 <sip%3Areginfo(a)192.168.20.20>")* If i am not configured the above server address, i am getting bad configuration Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: auth [auth_mod.c:333]: mod_init(): auth: qop set, but nonce-count (nc_enabled) support disabled Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /etc/kamailio/permissions.allow Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: permissions [permissions.c:607]: mod_init(): default allow file (/etc/kamailio/permissions.al... rule set Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /etc/kamailio/permissions.deny Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: permissions [permissions.c:616]: mod_init(): default deny file (/etc/kamailio/permissions.den... rule set *Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: ERROR: pua_reginfo [pua_reginfo.c:105]: mod_init(): server_address parameter not set* *Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: ERROR: <core> [sr_module.c:944]: init_mod(): Error while initializing module pua_reginfo (/usr/lib/...ginfo.so)* Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Oct 18 11:31:36 arm kamailio[2771]: already running ... failed! Oct 18 11:31:36 arm kamailio[2771]: . Oct 18 11:31:36 arm systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Hint: Some lines were ellipsized, use -l to show in full. I want to know what is the significance of this parameter and find no where this value is used. Thanks in advance. Regards, Raman

2 1

RTPProxy
by Rodrigo Moreira 20 Oct '16

20 Oct '16

Hi, What is difference between modules rtpproxy and rtpengine? Best regards. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

5 12

memory is not freed in system, while kamailio's statistic shows OK
by Vasiliy Ganchev 20 Oct '16

20 Oct '16

Hi folk! Have a production server - used as webrtc to sip proxy - it is built on kamailio v 4.4.2 - Server handles connections from ~4000 incoming SIP trunks (transport=TLS). Kamailio provides auth for this trunks. - Kamailio receives also connections from Webrtc app. Calls from Webrtc routed to one of the 4000 sip trunks. Server is running on Debian GNU/Linux 7 The number of calls on the system is small. Most of the load is generated by registration/re-registrations of SIP trunks The problem is following: - output of kamcmd core.shmmem and kamcmd pkg.stats shows constant using of memory - output of htop shows constantly increasing memory usage, e.g. after one week working: PID USER PRI NI VIRT RES SHR S CPU% MEM% TIME+ Command ..... 6013 kamailio 20 0 744M 134M 130M S 0.0 3.6 2:28.80 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 32 -u kamailio -g kamailio 6002 kamailio 20 0 744M 134M 130M S 0.0 3.6 2:30.93 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 32 -u kamailio -g kamailio 6012 kamailio 20 0 744M 134M 130M S 0.0 3.5 2:35.00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 32 -u kamailio -g kamailio 5991 kamailio 20 0 744M 133M 129M S 0.0 3.5 2:05.88 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 32 -u kamailio -g kamailio 6018 kamailio 20 0 744M 133M 129M S 0.0 3.5 2:31.90 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 32 -u kamailio -g kamailio ..... Despite running with: PKG_MEMORY=32, Amount of used memory per process is higher. Finishes all with monit killing kamailio, because of memory usage more than allowed studied this tutorial: https://www.kamailio.org/wiki/tutorials/troubleshooting/memory but did not find a way how to dig up the problem's root. Tried also to reproduce the issue in lab environment (having clone of production server). Generated load with incoming registration using PJSIP, but with no success - memory usage is constant. kamailio -v version: kamailio 4.4.2 (x86_64/linux) 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: unknown compiled with gcc 4.7.2 Any suggestion/idea how to troubleshoot deeper would be highly apprecieted Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/memory-is-not-freed-in-system-while… Sent from the Users mailing list archive at Nabble.com.

2 3

help with NOTIFY
by Slava Bendersky 19 Oct '16

19 Oct '16

Hello Everyone, I am trying forward NOTIFY to client. My setup is PBX Server ---- local lan ---kamailio ---- internet ---- client The problem with whole setup that kamailio is not forwarding NOTIFY to client. Here are log. U 2016/10/19 14:23:29.581609 10.18.130.50:5160 -> 10.18.130.46:5060 NOTIFY sip:10101@10.18.130.46:5060 SIP/2.0. Via: SIP/2.0/UDP 10.18.130.50:5160;rport;branch=z9hG4bKPjONocd4FvI5LkOmSR6ZCA5Z1f7SYDb3Ty. From: <sip:8101@networklab.ca>;tag=ziApds23FQsPL2zd6q8pCUlwW.jwa6mK. To: "Slava" <sip:10101@networklab.ca>;tag=734f813f. Contact: <sip:10.18.130.50:5160>. Call-ID: A0bJWg8v0nGmtUmiHzUixQ... CSeq: 18547 NOTIFY. Route: <sip:kamailio_local_ip:5060;lr;r2=on;ftag=734f813f>. Route: <sip:kamailio_public_ip:5081;transport=tls;lr;r2=on;ftag=734f813f>. Event: presence. Subscription-State: active;expires=413. Allow-Events: message-summary, presence, dialog, refer. Max-Forwards: 70. User-Agent: FPBX-13.0.188.9(14.0.2). Content-Type: application/pidf+xml. Content-Length: 579. . <?xml version="1.0" encoding="UTF-8"?> <presence entity="sip:8101@10.18.130.50:5160" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:pp="urn:ietf:params:xml:ns:pidf:person" xmlns:es="urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status" xmlns:ep="urn:ietf:params:xml:ns:pidf:rpid:rpid-person"> <note>On the phone</note> <tuple id="8101"> <status> <basic>closed</basic> </status> <contact priority="1">"Slava" <sip:10101@networklab.ca></contact> </tuple> <pp:person> <status> <ep:activities>ep:busy</ep:activities> </status> </pp:person> </presence> U 2016/10/19 14:23:32.045388 10.18.130.46:5060 -> 10.18.130.50:5160 SIP/2.0 408 Request Timeout. Via: SIP/2.0/UDP 10.18.130.50:5160;rport=5160;branch=z9hG4bKPju05rMNlzhfukLvaq4ELmq.QI4GBqDN-m;received=10.18.130.50. From: <sip:8101@networklab.ca>;tag=Yzy2m6O2AFT6rGSDBmo6PjcGaCWT1i5H. To: "Slava" <sip:10101@networklab.ca>;tag=ad5a945e. Call-ID: 2pq5iebmtLdB_OsMUs3EOg... CSeq: 14623 NOTIFY. Server: kamailio (4.3.4 (x86_64/linux)). Content-Length: 0. Any help thank you. volga629

1 0

Active calls limit
by Ivan Dudko 19 Oct '16

19 Oct '16

Hello! I am trying to implement limit of active calls for subscribers. And for each subscriber this limit must be personal. I find example route in presentation of Daniel-Constantin Mierla. And i need to set $xavp(caller=>active_calls) = 1; for each dialog. But i can't understand how to load this number of active calls from some-thing similar of user profile or subscriber table? route[DIALOG] { if (is_method("CANCEL") || (has_totag() && is_method("INVITE|BYE|ACK"))) { dlg_manage(); return; } if (is_method("INVITE") && !has_totag() && !isflagset(WITH_ACTIVE_CALLS_LIMIT)) { if( $xavp(caller[0]=>active_calls) != $null && $xavp(caller[0]=>active_calls) > 0 ) { if(!get_profile_size("caller", "$fU@$fd", "$var(acsize)")) { send_reply("500", "No more active calls"); exit; } if($var(acsize)>=$xavp(caller[0]=>active_calls)) { send_reply("403", "No more active calls"); exit; } set_dlg_profile("caller", "$fU@$fd"); } setflag(WITH_ACTIVE_CALLS_LIMIT); dlg_manage(); } }

2 5

Kamailio Bench
by Rodrigo Moreira 19 Oct '16

19 Oct '16

Hello, i need your help, I'm having problems with the reverse response of an INVITE passing through a rtpproxy (NAT). Does anyone have any idea what might be going on? UE is informed of the result INVITE, but the UAS does not receive the test bench detail. Greetings. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

2 1

Problems with CDRs
by Diogenes Marcano 19 Oct '16

19 Oct '16

I am using kamailio with ubuntu, but I can't obtain CDRs. In my conf. file related to CDR I have: modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_start_id", "start_time") modparam("acc", "cdr_end_id", "end_time") modparam("acc", "cdr_duration_id", "duration") But I don't know where is the problem. Regards Diogenes Marcano +58 414 117 2011

2 1

BYE issue
by Nelson Migliaro 18 Oct '16

18 Oct '16

Hello everybody, I am having issues with one SIP vendor. I have a Kamailio in bridge mode (private IP / Public IP) and some Asterisk and Media Gateways. Calls get established and I have two way audio but when the remote party hangs up the call, the BYE arrives to the Kamailio and does not move forward. I think the problem is SIP vendor rewrite the BYE header and change the asterisk IP with the public IP of the kamailio. The IP that appears in the header of the BYE have to be the same that appears in the contact (UAC that send the call, in my case the Asterisk). Vendor should not change that IP. ¿Am I correct? Thank you ----------------------------------------------------------------------------------------------------- INVITE ---------------------------------------------------------------------------------------------------- 2016/10/17 18:50:49.110967 PUBLIC-KAMAILIO-IP:5060 -> VENDOR-IP:6060 INVITE sip:DESTINATION-NUMBER@VENDOR-IP:6060 SIP/2.0 Record-Route: <sip:PUBLIC-KAMAILIO-IP;r2=on;lr=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1MMDIudm9pY2U G9jYWw-;did=09b.9572;nat=yes> Record-Route: <sip:PRIVATE-KAMAILIO-IP;r2=on;lr=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1MMDIudm9pY2U G9jYWw-;did=09b.9572;nat=yes> Via: SIP/2.0/UDP PUBLIC-KAMAILIO-IP;branch=z9hG4bK06a.07540d0e2f32a811ecf9c0a5235dc77a.1 Via: SIP/2.0/UDP PRIVATE-ASTERISK-IP:5060;received=PRIVATE-ASTERISK-IP;branch=z9hG4bK6bb5a7b3;rport=5060 Max-Forwards: 69 From: "SOURCE-NUMBER" <sip:SOURCE-NUMBER@MY-COMPANY>;tag=as5e87b96c To: <sip:DESTINATION-NUMBER@VENDOR-IP> Contact: <sip:SOURCE-NUMBER@PRIVATE-ASTERISK-IP:5060> Call-ID: 025cc3717ba59faa000cf4db6f8be588@PRIVATE-ASTERISK-IP:5060 CSeq: 102 INVITE User-Agent: UAC Date: Mon, 17 Oct 2016 16:53:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 269 v=0 o=root 292850421 292850421 IN IP4 PUBLIC-KAMAILIO-IP s=Asterisk PBX c=IN IP4 PUBLIC-KAMAILIO-IP t=0 0 m=audio 23456 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes ----------------------------------------------------------------------------------------------------- BYE ----------------------------------------------------------------------------------------------------- 2016/10/17 18:50:58.241666 VENDOR-IP:6060 -> PUBLIC-KAMAILIO-IP:5060 BYE sip:SOURCE-NUMBER@PUBLIC-KAMAILIO-IP:5060 SIP/2.0 Via: SIP/2.0/UDP VENDOR-IP:6060;branch=z9hG4bKeff4.48943e76.0 Via: SIP/2.0/UDP VENDOR-IP:5060;branch=z9hG4bK1d4e605e4ll19f74fBYE421ce8658050206 Max-Forwards: 34 Route: <sip:PUBLIC-KAMAILIO-IP;lr;r2=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1> Route: <sip:PRIVATE-KAMAILIO-IP;lr;r2=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1> To: "SOURCE-NUMBER"<sip:SOURCE-NUMBER@YO>;tag=as5e87b96c From: <sip:DESTINATION-NUMBER@PUBLIC-KAMAILIO-IP>;tag=421ce86-co1547-INS001 Call-ID: 025cc3717ba59faa000cf4db6f8be588@PRIVATE-ASTERISK-IP:5060 CSeq: 154701 BYE User-Agent: VENDOR Content-Length: 0 -----------------------------------------------------------------------------------------------------

3 5

420 Bad Extension error
by Infinicalls Infinicalls 18 Oct '16

18 Oct '16

Hi all, I am testing Edge Proxy setup with the example given here : http://kamailio.org/docs/modules/4.2.x/modules/outbound.html and I get this error : U 2016/10/06 09:18:48.331580 10.0.0.4:5060 -> 52.233.25.164:5060 SIP/2.0 420 Bad Extension. Via: SIP/2.0/UDP 10.1.0.4;branch=z9hG4bK162a.e9f73fe5e52ded1f30801aa00ee2a771.0; received=52.233.25.164. Via: SIP/2.0/UDP 10.114.70.81:1228;received=113.193.149.49;rport=1228;branch=z9h G4bKPj3ec7c87e09d6452b9e68e60052cb2651. From: "User1" <sip:user1@infinicalls-proxy1.myserver.com>; tag=ac1e36aec8bf445bbec8e43cf28e9f48. To: "User1" <sip:user1@infinicalls-proxy1.myserver.com>;ta g=b27e1a1d33761e85846fc98f5f3a7e58.b172. Call-ID: a270d285de6145c6b8670b909b104e74. CSeq: 20149 REGISTER. Contact: <sip:user1@10.114.70.81:1228;ob>;expires=300. Unsupported: path. Path: <sip:10.1.0.4;lr>. Supported: outbound. P-Registrar-Error: No support for found Path indicated. Server: kamailio (4.4.3 (x86_64/linux)). Content-Length: 0. Both my proxy and registrar are behind NAT. rtpproxy is installed and started. I already have modparam("registrar", "use_path", 1) in my registrar. Any idea what is missing? Thanks. regards Ganesh Kumar

2 17

Sip route can be as IMS PCSCF node
by Surender Singh 17 Oct '16

17 Oct '16

Hi, Can we use sip router as IMS PCSCF to connectivity with IMS. Does sip router supports the diameter RX interface to connectivity with PCRF . Regards Surender Singh ::DISCLAIMER:: ---------------------------------------------------------------------------------------------------------------------------------------------------- The contents of this e-mail and any attachment(s) are confidential and intended for the named recipient(s) only. E-mail transmission is not guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or may contain viruses in transmission. The e mail and its contents (with or without referred errors) shall therefore not attach any liability on the originator or HCL or its affiliates. Views or opinions, if any, presented in this email are solely those of the author and may not necessarily reflect the views or opinions of HCL or its affiliates. Any form of reproduction, dissemination, copying, disclosure, modification, distribution and / or publication of this message without the prior written consent of authorized representative of HCL is strictly prohibited. If you have received this email in error please delete it and notify the sender immediately. Before opening any email and/or attachments, please check them for viruses and other defects. ----------------------------------------------------------------------------------------------------------------------------------------------------

2 1

sipml5 register to kamailio IMS (Unauthorized - Challenging the UE)
by Serhat Guler 17 Oct '16

17 Oct '16

Hi, I am trying to register sipml5 webrtc client to my kamailio IMS setup. I have tried to register the client both with ws and wss, but it seems to be that the sipml5 doesn't calculate the authentication digest right. The authentication mechanism i set to AKAv1-MD5 as default in the hss. A simple wireshark file is attached. .10 being the host, .11 being the kamailio server. >From the output of scscf we can see that the digests do not match. [REGISTER] from [sip:bob@net1.test] to [sip:bob@net1.test] ims_auth [authorize.c:824]: authenticate(): uri=sip:net1.test nonce=rB2iDuerHwoy+LUStSOsYojAESfWmAAApFZ3XNB8FdA= response=61cbdbeb47c9880ededfca51c3801800 qop=auth-int nc=00000001 cnonce=2d4545cf4c935c8094c8b1da3d4a2976 hbody=d41d8cd98f00b204e9800998ecf8427e ims_auth [authorize.c:872]: authenticate(): UE said: 61cbdbeb47c9880ededfca51c3801800 and we expect 7e27ef414cf37d96cd1c849bb7e59415 ha1 a53988ba0b257941bc747b1026225c77 (REGISTER) tm [tm.c:1265]: w_t_reply(): ERROR: t_reply: cannot send a t_reply to a message for which no T-state has been established Thanks a lot, Serhat

2 2

How to use sql transformations?
by Daniel Tryba 17 Oct '16

17 Oct '16

How to use the sql transformation? https://www.kamailio.org/wiki/cookbooks/4.4.x/transformations#sql_transform… has the following example: xlog("$$rm = $rm = $(rm{s.sql})"); But adding this to the request_route and starting kamailio will fail: ERROR: pv [pv_trans.c:2351]: tr_parse_string(): unknown transformation: sql}/sql/3! ERROR: <core> [pvapi.c:1629]: tr_lookup(): error parsing [{s.sql}] ERROR: <core> [pvapi.c:1010]: pv_parse_spec2(): bad tr in pvar name "rm" ERROR: <core> [pvapi.c:1036]: pv_parse_spec2(): invalid parsing in [$(rm{s.sql})] at (4) ERROR: xlog [xlog.c:512]: xdbg_fixup_helper(): wrong format[$$rm = $rm = $(rm{s.sql})] ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:372 line 372 is the above xlog and sqlops.so is loaded (and works). Anybody got a working example of this? Or an other hint to prevent sql injections when using user supplied variables in sql queries?

1 0

503 Service unavailable to 500
by Dhruvin Desai 17 Oct '16

17 Oct '16

Hi 503 Service unavailable received at kamailio SIP server is forwarded as 500 server internal error to the UAC. Which seems fine based on the fact that there is no retry-after header. Would like to know if there is any way to to control the SIP Headers like adding Proprietary SIP Headers to the 500 "Server Interrnal Errror" created by kamalio stack for 503 service unavailable (without retry-after Header) It seems that Kamalio does not retain the proprietary sip headers received in 503 to 500 response forwarded. Any insight into the issue would be helpful -- Warm Regards Dhruvin Desai.

2 1

RTPEngine & Homer
by Grant Bagdasarian 17 Oct '16

17 Oct '16

Hello, I'm trying to get rtpengine to send its stats to Homer, but I'm unsure to which IP I should configure RTPengine to send the stats? We have a Kamailio server configured with the sipcapture module, which works perfectly for SIP traffic, but RTP stats are not visible in Homer. Do I need to point RTPengine to the sip capture server or directly to Homer? Running RTPEngine 4.5.2 and Kamailio 4.4.1. Regards, Grant

1 0

HOMER / HEP support in SEMS?
by Daniel Pocock 14 Oct '16

14 Oct '16

Does anybody know if SEMS has HOMER / HEP support like Kamailio, or if it is under development? It would be useful to extract RTCP stats from SRTP streams, the standalone HOMER captagent can't see inside them when they are encrypted (SRTCP) Regards, Daniel

2 1

Accounting CDR generation
by Ivan Dudko 14 Oct '16

14 Oct '16

Hello! I am setting this in kamailio.cfg #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_user=$rU;dst_ouser=$tU;dst_domain=$rd;sip_code=$rs;sip_reason=$rr;callid=$dlg(callid);dst_ip=$Ri") modparam("acc", "cdr_start_id", "start_time") modparam("acc", "cdr_end_id", "end_time") modparam("acc", "cdr_duration_id", "duration") #!endif But in CDRs table ('acc_cdrs') field dst_user=$rU is always empty if call is not completed with remote side answer. But in accounting table ('acc') this field isn't empty. Is this behavior is correct? I think it is not correct.

2 7

Websocket with kamailio IMS
by Serhat Guler 13 Oct '16

13 Oct '16

Dear all, I am trying to configure websockets for kamailio IMS. My basic kamailio(without IMS) setup works properly with websockets and without websockets. The IMS functionality also works, but only with IMS clients and now I want to use webrtc clients and make calls. At the moment I am trying to add the websockets support to the pcscf. I have did some changes to the pcscf cfg file. My problem is not really related to the cfg of pcscf as the component is listening for tcp/tls connections, but I cannot connect through the assigned ports. The output of the pcscf component and the the wireshark pcap file for the Websocket are attached. (I basically tried to browse http://server.ip.address:tcp.port or https://server.ip:tls.port without using a webrtc client yet). 192.168.0.10 -> host 192.168.0.11 -> virtual machine running kamailio Thanks in advance, Serhat

2 1

Reload PIPILIMIT table from DB
by Grant Bagdasarian 12 Oct '16

12 Oct '16

Hello, Is it possible to live reload the pipelimit module so it gets new values from the DB without a restart of Kamailio? I couldn't find a command to do this. Regards, Grant Bagdasarian CM

2 3

prefix_route and db_cluster not working
by Martin Nyström 12 Oct '16

12 Oct '16

Hi, I have an issue when I am unable to get prefix_route to work together with db_cluster. If I set the db_url manually it works great, pointing it to a specific database. However we need the redundancy for this module. Dispatcher uses the same cluster without any issues. The following are my configuration sets: # DB CLUSTER (db_cluster.inc) loadmodule "db_cluster.so" modparam("db_cluster", "connection", "KAMDB01=>mysql://kamailio:password123@192.168.55.10/kamailio") modparam("db_cluster", "connection", "KAMDB02=>mysql://kamailio:password123@192.168.55.11/kamailio") modparam("db_cluster", "cluster", "AccCluster=>KAMDB02=9s9s;KAMDB01=4s4s") modparam("db_cluster", "cluster", "Settings=>KAMDB02=9s9s;KAMDB01=4s4s") # KAMAILIO.CFG (kamailio.cfg) # PREFIX ROUTE MODPARAM loadmodule "prefix_route.so" include_file "db_cluster.inc" modparam("prefix_route", "db_url", "cluster://Settings") modparam("prefix_route", "db_table", "prefix_route") modparam("prefix_route", "exit", 1) Syslog output: Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: ERROR: <core> [db_cmd.c:122]: db_cmd(): DB driver cluster does not implement mandatory db_cmd function Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: ERROR: <core> [db_cmd.c:168]: db_cmd(): db_cmd: Cannot create db_cmd structure Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: ERROR: prefix_route [prefix_route.c:126]: pr_db_load(): prefix_route: db_load(): db_cmd() failed Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: : prefix_route [prefix_route.c:214]: mod_init(): prefix_route: db_load() failed Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: ERROR: <core> [sr_module.c:974]: init_mod(): Error while initializing module prefix_route (usr/local/lib64/kamailio/modules/prefix_route.so) Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Ubuntu 14.04LTS version: kamailio 4.3.6 (x86_64/linux) 105a17 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, DBG_QM_MALLOC, 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: 105a17 compiled on 10:06:12 Oct 4 2016 with gcc 4.8.4 Thank you.

2 3

Enabling accounting module
by Diogenes Marcano 12 Oct '16

12 Oct '16

Can I enable accounting modules acc and acc_cdrs with out diameter? Best regard

2 1

Permissions module question
by Ivan Dudko 11 Oct '16

11 Oct '16

Hello! I am new to kamailio and trying to use it vanilla config. Now main question is how to use ip based auth. I found recent post of Daniel-Constantin Mierla: http://lists.sip-router.org/pipermail/sr-users/2011-December/071147.html Here he recommends to use 'address' table from permissions module, I try yo use advice and add this lines at config begin: #!define WITH_MYSQL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!define WITH_AUTH #!define WITH_IPAUTH #!define WITH_USRLOCDB Database is created and kamailio can access it. I am add user 1000 to kamailio via kamctl and successful register it with soft-phone. I create trunk without registration to kamailio on asterisk server. And trying to call from asterisk to user 1000. Call is successful. I try to create file /etc/kamailio/permissions.deny with content 'ALL : ALL'. And retry previous call. It still sucessful. I try to add record with asterisk address to 'address' table with group 1. And retry previous call. It still sucessful. I am confused. I do not now how to disable any address for ip_auth except if it in the 'address' table. And allow any address with if it request kamailio with registration. -- Best Regards, Ivan Dudko

3 4

Kamailio - DMQ dns srv Question
by José Seabra 11 Oct '16

11 Oct '16

Hello, I have a doubt related with DMQ dns behavior, I noticed that when kamailio starts, it tries to resolve DMQ name configured on parameter notification_address as the following sequence: 1. SRV 2. A 3. AAAA Isn't supposed kamailio try first resolve the NAPTR DMQ name, and then SRV? I'm asking this because kamailio is trying resolve the SRV record without any transport protocol specified on query, as my dns server only accepts queries on the format "_Service._Proto.Name" the SRV will never be resolved. Thank you. BR -- José Seabra

4 13

transaction is not suspended ERROR
by david 11 Oct '16

11 Oct '16

Hello all i'm having some errors sometimes and i'm not able to know why they happen i made a lua configuration script where i create some ESL sockets with several freeswitches (10) (around 500-600 sockets in total, depending on the number os children (8) and listen ips we have (5)), and i use the lua script to send some API to the FSW by a function. No matter what the FSW replies. i dont know why, but i think that having that conn:bgapi($command) in the lua script, makes the issue to happen. theorically, the lua script waits to receive the response from the fsw through the tcp socket, and after that the script continues (with an async_sleep command), but after the API sent, i see (only sometimes, few times) an async_sleep seem to fail when doing the t_continue with the following error WARNING: tm [t_suspend.c:186]: t_continue(): transaction is not suspended [20748:225229907] could it be posible that having manye ESL sockets opened in the kamailio server makes this to happen? how many async workers are needed? i have 8 right now is there a way where i can know or correlate the log to a $ci or something known? best regards david

2 3

Kamailio on Redhat 7.2
by kamailio noob 11 Oct '16

11 Oct '16

Hi All, Can anyone please provide me the details that how to install kamailio on Redhat 7.2. I am following the below url guide for installation of Kamailio. https://www.kamailio.org/wiki/install/4.4.x/git when i compile kamailio code I am not able to find certain scripts in /usr/local/sbin Can anyone help regarding this ? Regards, Atif Ramzan.

2 1

Busy return code when a user has an ongoing transaction
by Gholamreza Sabery 10 Oct '16

10 Oct '16

I want to be able to return a Busy code foe a user that is talking to someone else and receives an INVITE for a new call. How can I implement this in Kamailio? Regards

2 2

Order of record-route
by Dirk Teurlings - Signet B.V. 10 Oct '16

10 Oct '16

Hi, Looking at an issue with SIP-Cisco and a Barracuda firewall, I've come across something I need to know more about. The setup is as follows: CLIENT -> KAMAILIO -> ASTERISK CLIENT <- KAMAILIO <- ASTERISK Client sends INVITE, Kamailio forwards to Asterisk, SIP 100 and SIP 183 are being sent and finally a 200 OK. We now expect the Client to send back an ACK packet, but that never reaches us. In this case the SIP 183 is the first packet that contains the full record route of the path, the 200 OK contain the same record-routes being: Record-Route: <sip:192.168.0.200;r2=on;lr;ftag=12D1120C-1C4;did=4c8.9203;nat=yes> Record-Route: <sip:1.2.3.4;r2=on;lr;ftag=12D1120C-1C4;did=4c8.9203;nat=yes> Record-Route: <sip:4.3.2.1;r2=on;lr;did=4c8.3f471377> Record-Route: <sip:10.0.0.101;r2=on;lr;did=4c8.3f471377> 192.168.0.200 is the private address of Kamailio 1.2.3.4 is the public address of Kamailio 4.3.2.1 is the public address of the Client 10.0.0.101 is some private addres of the Client What we're seeing is that the Client is sending his ACK message to our 192.169.200 private address, which he will never reach. So me question in this case is; is the order of the Record-route headers relevant and do I have any say in which order Kamailio outputs these headers? To be clear, the Kamailio config only contains one reference to record_route which is caught for INVITE messages, which only fires when all things like NAT and transaction checks haven't already picked it up. Kind regards, Dirk

3 5

kamctl - Error opening FIFO
by Hermann Norpois 10 Oct '16

10 Oct '16

Hello, I have problems with the behaviour of kamctl ubuntu@ip-172-31-32-10:/etc/kamailio$ kamctl ul show ERROR: Error opening Kamailio's FIFO /var/run/kamailio/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module. #my FIFO file ubuntu@ip-172-31-32-10:/etc/kamailio$ ls -l /var/run/kamailio/kamailio_fifo prw-rw---- 1 kamailio kamailio 0 Oct 7 09:01 /var/run/kamailio/kamailio_fifo #FIFO in config ubuntu@ip-172-31-32-10:/etc/kamailio$ grep fifo kamailio.cfg loadmodule "mi_fifo.so" # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo") #With sudo it works ubuntu@ip-172-31-32-10:/etc/kamailio$ sudo kamctl ul show Domain:: location table=1024 records=0 max_slot=0 Further ... ubuntu@ip-172-31-32-10:/etc/kamailio$ kamctl restart INFO: Stopping Kamailio : ERROR: No PID file found (/var/run/kamailio.pid)! Kamailio probably not running INFO: check with 'ps axw | /bin/egrep kamailio' #But kamailio is running ubuntu@ip-172-31-32-10:/etc/kamailio$ ps axw|egrep kamailio 4122 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4123 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4124 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4125 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4126 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4127 ? S 0:46 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4128 ? S 0:44 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4129 ? S 0:44 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4130 ? S 0:44 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4131 ? S 0:03 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4132 ? S 909:27 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4133 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4134 ? S 5:08 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4135 ? S 25:51 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4136 ? S 26:02 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4137 ? S 26:08 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4138 ? S 26:18 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4139 ? S 10:17 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 27344 pts/1 S+ 0:00 egrep --color=auto kamailio Whats wrong with my usage of kamctl? Thanks Hermann

3 2

Q: (markup) language for defining structures and function prototypes
by Daniel-Constantin Mierla 10 Oct '16

10 Oct '16

Hello, writing to see if anyone is aware of any (well known) markup language for defining structures and function prototypes. The goal is to use it for specifying the functions exported by KEMI (https://www.kamailio.org/wiki/devel/config-engines) so we can generate C code for the exported structure, but more important, to get at least the index with the functions available over the KEMI. I feel it's going to be a big burden to write C code and docs in two places, even worse, to keep both of them sync'ed over the time. At this moment, there are many functions exported via KEMI, but not documented anywhere (apart of the C code). It should allow defining the name of the function, the parameters with their associated types, return code and eventually the description for each of them. At the end we could invent one based on xml format (or something similar), but I would better use something that already exists there. I searched quickly, but couldn't find anything straightaway. A hint could be a format that is used to document the RPC commands. Once the format is decided, we may have to write a tool to generate C code to be compiled with the module and the xml docbook file to be included in the documentation. Any idea to solve this welcome! Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com

1 0

Kamailio stop processing SIP
by Igor Potjevlesch 07 Oct '16

07 Oct '16

Hello, Someone has an idea? Again, Kamailio stop processing SIP during more than 30s. Nothing print in the logs. The processing back on when a script detect no answer and restart the service. Regards, Igor. De : Igor Potjevlesch [mailto:igor.potjevlesch@gmail.com] Envoyé : vendredi 23 septembre 2016 11:05 À : sr-users(a)lists.sip-router.org Objet : Dimension number of child Hello, I'm wondering if there's a best practise to dimension the number of child to setup? I see, very often, Kamailio that stops to reply to SIP traffic for couple a seconds. This cause some calls drop if RE-INVITE are not processed for example. So, I'm wondering if the issue is not due to a lack of child for processing the SIP traffic. Thank you for your inputs. Regards, Igor.

3 6

Migrating from OpenSER config file question.
by Brian McCrary 07 Oct '16

07 Oct '16

Hello group, I know this may be hard to believe but I'm in the process of upgrading an old, but stable, OpenSER 1.0 group of servers to Kamilio. I'm basically going to sort of start from scratch with the database and export my old MySQL database and write some scripts to reimport the data using kamctl scripts since the database structure has changed substantially since then it looks like. I've mostly migrated the config file over fairly easily, except for AVPops. I've got a section of config file that writes into the usr_preferences database for call forwarding, but I seem to get the same error anywhere I try to write to the database using AVPops. Here's the output I get when checking the config: 0(1378) DEBUG: <core> [pvapi.c:268]: pv_cache_add(): PV cache not initialized, doing it now 0(1378) ERROR: <core> [pvapi.c:828]: pv_parse_spec2(): error searching pvar "avp" 0(1378) ERROR: <core> [pvapi.c:1032]: pv_parse_spec2(): wrong char [s/115] in [$avp(s:callfwd)] at [5 (5)] 0(1378) : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 145, column 4-18: Can't get from cache: $avp(s:callfwd) ERROR: bad config file (1 errors) Here's what I think are the revelant portions of the config file: modparam("avpops", "db_url", "mysql://user:pass@localhost/kamailio") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops", "uuid_column", "uuid") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "value") modparam("avpops", "type_column", "type") <snip> if (avp_db_load("$from/username", "$avp(s:callfwd)")) { if(!avp_db_delete("$from/username", "$avp(s:callfwd)/usr_preferences")) { sl_send_reply("500", "Server DB error"); } } # avp_write("$ruri","$avp(s:callfwd)"); # Old OpenSER 1.0 syntax... $avp(s:callfwd) = $ruri; # New Kamailio syntax ?? avp_db_store("$from/username","$avp(s:callfwd)/usr_preferences"); t_relay(); return; Basically I just want to write the ruri into the usr_preferences database with an attribute of callfwd but not sure what I'm missing. I see the error "Can't get from cache" but not sure how that pertains to what I'm trying to do. Anyone have any ideas? Thanks, Brian

2 5

Size of passwords fields in db
by Daniel-Constantin Mierla 06 Oct '16

06 Oct '16

Hello, writing here to decide on a topic opened by pull request 779: - https://github.com/kamailio/kamailio/pull/779 what would be a fair size for db column storing a password that one would like to have for proper security? I would like to make it consistent over all tables that have a password column by defining a xml entity for the size of these columns. The pull request suggests 64 chars, has anyone other opinions on making it larger or smaller? If they are defined varchar, then should not be a problem of allocated size, so we can go with 128 if someone feels it worth doing larger now so we don't have to change it again in the near future. This change is about db schema, the modules I expect to work with allocated strings (or have length checks) in this case and should not be affected. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com

2 4

Fwd: TOPOH Module in Kamialio
by Ginti Saurabh 06 Oct '16

06 Oct '16

Hi, Can you let me know what is TOPOH module do in kamailio? what does it do? and if commented out this line then will it effect the system?

2 6

Load balancing with Kamailio
by Nitesh Bansal 06 Oct '16

06 Oct '16

Hello, I'm planning on doing some smart load balancing with Kamailio. We have a distributed network, with multiple Kamailio boxes in different locations serving as Ingress SBC, these Kamailio boxes are the entry point for a SIP call and then they route the call to a pre-configured Asterisk boxes. I want to move away from this, I would like these Kamailios to be able to distribute the traffic to Asterisk boxes based on the actual load on these boxes, the goal is to be more dynamic? Is there any Kamailio module which could do that? Do I need to integrate some other tool(Homer etc) with Kamailio to achieve this? Any suggestions are welcome. -- *Nitesh Bansal* Business Development Engineer http://www.voxbone.com/

2 1

Serial Forking and t_suspend
by Kelvin Chua 06 Oct '16

06 Oct '16

we have serial forking already setup properly but need to t_suspend(). found out that it completely breaks the destination set. other than manually rebuilding the destination set after t_continue(), are there any other solution out there? Kelvin Chua

2 1

Re: [SR-Users] Identical Kamailio setup on 2 hosts
by Daniel Tryba 05 Oct '16

05 Oct '16

On Fri, Sep 30, 2016 at 10:29:16PM +0530, Infinicalls Infinicalls wrote: > Hi Daniel, > This seems to be ideal setup for my requirement. Would you mind > sharing the kamailio.cfg file with me (with all the important data > masked) ? Not much important stuff to censor. LBIPADDR is the ipadress of the machine itself. REGISTRARNET/30 is the net of the backend machines. The dispatcher.list contains the backend registrars The backend is essentially the default config (with an always on rtpengine) with the following additions: You need to enable path.so and set modparam("path", "use_received", 1) Tell the registrar module to use path (path_mode 1 fixes you 420 problem): modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 1) modparam("registrar", "path_use_received", 1) If you want to send keepalives from the registrar to the enduser: modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "udpping_from_path", 1) the loadbalancer config has a little blurb about relaying the sipping from the backend manually using the path header. But I'd advice to ping from the an external process to have better control (eg sipsak)

2 3

Addressing a distinct PSI result is: 555 AS Contacting Failed - iFC terminated dialog
by Valentin Christoph 05 Oct '16

05 Oct '16

Hi all, I'm trying to send a call to an AS via a kamailio S-CSCF (release 4.4.3). I'm addressing the AS by a distinct PSI: sip:04325432101@domain.net;user=gsmr. At the first try the S-CSCF sends a SAR to HSS and receives a successful SAA (tshark traces attached), but the S-CSCF rejects the call with 555 AS Contacting Failed - iFC terminated dialog. At the second try (a few seconds later), the S-CSCF doesn't exchange SAR/SAA with the HSS, and now the call is successfully forwarded to the AS. This problem is permanent, it is not the result of a temporary loss of connectivity between S-CSCF and AS. I checked the iFCs that I had configured at the HSS and I think, they are OK. I attached - tshark traces from the node, where the S-CSCF resides: 2016_09_19_error_555_17_scscf_filtered.pcap - WITH_DEBUG logs from kamailio: 2016_09_19_error_555_17_kamailio_1st_try.log - A tarball of /usr/local/etc/kamailio: 2016_09_19_error_555_17_kamailio.tar.gz Can anybody help? Is it a bug in kamailio or is the problem on my side? Thanks, Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

2 4

484 Address Incomplete error
by Infinicalls Infinicalls 05 Oct '16

05 Oct '16

Hi all, I am trying a simple proxy (with dispatcher) and a registrar. However, I have this issue U 2016/10/05 08:22:44.057195 10.0.0.4:5060 -> 52.233.25.164:5060 SIP/2.0 484 Address Incomplete. Via: SIP/2.0/UDP 10.1.0.4;branch=z9hG4bK8653.a9c65320000000000000000000000000.0;received=52.233.25.164. To: <sip:51.140.187.167:5060>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b79c. From: <sip:calls@infinicalls.com>;tag=533cb9e91f4b999cf76861cbb9ed54ed-d3d8. CSeq: 10 OPTIONS. Call-ID: 28fcdbee608363cf-5442(a)127.0.0.1. Server: kamailio (4.4.3 (x86_64/linux)). Content-Length: 0. I have set modparam("dispatcher", "flags", 2) Any idea? Thanks. regards Ganesh Kumar

2 1

NatHelper - Received adding on Contact header
by Yasin CANER 04 Oct '16

04 Oct '16

3 4

Sip Proxy server_kamailio Features
by Surender Singh 03 Oct '16

03 Oct '16

Hello, Thanks to all to develop the such type Sip Proxy server. Please suggest whether SIP proxy server is open source ,If yes than it can fulfil the my below requirement . 1) I having some sip clients which are registered on IMS .Can I use the kamailio Sip proxy sever between sip client and IMS. Sip client ------------------Kamailio Sip Proxy ---------------IMS 2) Second things, can I use Kamailio Sip Proxy to interconnect two IMS nodes . Can I create the SIP trunk without registration mode (username and password) between IMS nodes using Kamailio. Because the my IMS node doesn't support sip trunk in registration mode. Its support one normal trunk (SIP-T) My current IMS set up like below. IMS-1 (IBCF)---------------IMS2(IBCF). Regards Surender Singh ::DISCLAIMER:: ---------------------------------------------------------------------------------------------------------------------------------------------------- The contents of this e-mail and any attachment(s) are confidential and intended for the named recipient(s) only. E-mail transmission is not guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or may contain viruses in transmission. The e mail and its contents (with or without referred errors) shall therefore not attach any liability on the originator or HCL or its affiliates. Views or opinions, if any, presented in this email are solely those of the author and may not necessarily reflect the views or opinions of HCL or its affiliates. Any form of reproduction, dissemination, copying, disclosure, modification, distribution and / or publication of this message without the prior written consent of authorized representative of HCL is strictly prohibited. If you have received this email in error please delete it and notify the sender immediately. Before opening any email and/or attachments, please check them for viruses and other defects. ----------------------------------------------------------------------------------------------------------------------------------------------------

2 1

Kamailio number of concurrent calls
by Gholamreza Sabery 03 Oct '16

03 Oct '16

I used SIPp to stress test different scenarios in Kamailio, but how can I simulate a real call with media? SIPp has media sending capabilities but in not enough for example to simulate 1000 calls persecond! How can I test this?

5 7

How to check in Kamailio if a current sip message 200 OK is a response to the BYE.
by AFRICALLSHOP 01 Oct '16

01 Oct '16

Hello, I try to check in Kamailio if a current sip message "200 OK" is a response to a "BYE" request. Could you confirm the following code please? IF (IS_METHOD("BYE") && STATUS == 200) { } Regards Abdoul.

2 3

Proxy unable to connect - 420 Bad Extension error
by Infinicalls Infinicalls 01 Oct '16

01 Oct '16

Hi, I have setup a proxy and able to connect to the registrar. However, after a few attempts, it throws the "420:Bad Extension" error, saying "Unsupported: path. Path: <sip:10.1.0.4;lr>. Supported: outbound. P-Registrar-Error: No support for found Path indicated." Any idea, how to overcome this ? 10.1.0.4 is not my IP anywhere. I don;t know where it takes that value from. Thanks. regards Ganesh Kumar

2 2

Call forking priority
by Igor Potjevlesch 30 Sep '16

30 Sep '16

Hello list, I have a proxy that manage fix calls and a proxy that manage mobile calls. I want to implement call forking so when I call to a fix phone the call goes to the fix proxy and it forks to the mobile proxy who manage the call and viceversa. I want the fix to ring 2 or 3 times before the mobile. In order to do so I relay first to the fixed proxy, which will bounce back to mobile proxy the branch for mobile device. But now if I want to be able to choose which phone (mobile or fixe) rings before I have tried by doing the "async_route("RELAY", "7");" in the fixe proxy like in the mobile proxy but it doesn't work. My code is: Fix proxy: if($rU==123456789){ add_diversion("forking"); $var(priority)=1; append_branch("sip:987654321@proxy_mobile"); } route[INVITE]{ if($var(priority) == "1"){ async_route("RELAY", "7"); }else{ route(RELAY); exit; } Mobile proxy: if($tU==987654321){ $rU=123456789; } route[INVITE]{ if($dir == "forking"){ async_route("RELAY", "7"); }else{ route(RELAY); exit; } } Thanks.

2 2

ims_charging state machine issue
by 4fba81+8i7felu07050＠guerrillamail.com 30 Sep '16

30 Sep '16

Wanted to ask about changing rates while a call is not yet connected. I did this, $var(cc_ret1) = Ro_CCR_Stop("term", "487", "Request Terminated"); $var(cc_ret2) = Ro_CCR("RO_REPLY", "term", 30, "1", "1"); however, upon sending the new CCR, i get this ERROR: cdp [acctstatemachine.c:185]: cc_acc_client_stateful_sm_process(): Received unknown event [1] in state [1] and CRITICAL: ims_charging [ro_timer.c:122]: insert_ro_timer(): Trying to insert a bogus ro tl=0x7f484ab48728 tl->next=0x7f484ab42858 tl->prev=0x7f484aaa1090 looks like there is some statemachine issue. question is, how do we change rate? ---- Sent using GuerrillaMail.com Block or report abuse: https://www.guerrillamail.com/abuse/?a=UlM5EB4FUac1hAen5G1RYR3XANifx8NV1epN…

1 0

Identical Kamailio setup on 2 hosts
by Infinicalls Infinicalls 30 Sep '16

30 Sep '16

Hi, I have two identical hosts running Kamailio with the same set of users. They are located in different locations. Both are running behind NAT and I've enabled rtpproxy for that and also advertised the public IP. Both the MySQL DBs are similar and have the same data. Both the hosts have different sub-domains in alias and I have set auto_alias=no Now, the problem is my users are not able to communicate from one server to another. user1(a)mydomain.com from host1 is not able to communicate to user1(a)mydomain.com at host2, though inter-server communication is happening nicely. Any idea how to solve this? Thanks. regards Ganesh Kumar -- --- http://www.infinicalls.com

3 8

Load balancing traffic with Kamailio
by Nitesh Bansal 29 Sep '16

29 Sep '16

Hello, I'm planning on doing some smart load balancing with Kamailio. We have a distributed network, with multiple Kamailio boxes in different locations serving as Ingress SBC, these Kamailio boxes are the entry point for a SIP call and then they route the call to a pre-configured Asterisk boxes. I want to move away from this, I would like these Kamailios to be able to distribute the traffic to Asterisk boxes based on the performance of the Asterisk ,it would be great if we could take a look at SIP response time from Asterisk and use that as metric for load balancing the traffic) Is there any Kamailio module which could do that? Do I need to integrate some other tool with Kamailio to achieve this? Any suggestions are welcome. -- *Nitesh Bansal* Business Development Engineer http://www.voxbone.com/

3 3

Re: [SR-Users] Distributed Message Queue Module
by Charles Chance 29 Sep '16

29 Sep '16

Hello, That depends entirely on what you need to replicate between servers. DMQ is just a communication channel. How you use it is up to you. For example, if you enable dmq_usrloc, there's no need to share a location database. Htable has an option for replicating over dmq too. The number of direct module integrations like dmq_usrloc/htable is fairly limited, but you can also share arbitrary data between nodes yourself from within your config. I think you'll need to provide a little more information about what you're trying to achieve, however, in order for us to offer any further advice. Cheers, Charles On 29 Sep 2016 08:08, "Moacir Ferreira" <moacirferreira(a)hotmail.com> wrote: Hi, When using the DMQueue module what is the best approach, using one SQL server/database for each member node or having a single SQL server for all nodes? Thanks, Moacir _______________________________________________ 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 -- Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Faraday Wharf, Innovation Birmingham Campus, Holt Street, Birmingham Science Park, Birmingham B7 4BB.

1 0

Distributed Message Queue Module
by Moacir Ferreira 28 Sep '16

28 Sep '16

Hi, When using the DMQueue module what is the best approach, using one SQL server/database for each member node or having a single SQL server for all nodes? Thanks, Moacir

1 0

memory problem
by beer Ll 28 Sep '16

28 Sep '16

hi all I have kamailio 4.2.0 on Debian Jessie the parameters on startup of kamailio are : SHM_MEMORY=512 PKG_MEMORY=32 the server is dedicated to the kamailio SIP service with authentication and accounting on a separated db server the server process about 30/50 call setups per second everythings works good but after 5/6 days the processes lock and I received a lot of error log like these: <core> [rvalue.c:1710]: rval_str_add2(): rv eval out of memory <core> [msg_translator.c:362]: received_builder(): out of memory <core> [msg_translator.c:2717]: via_builder(): out of memory ... <core> [msg_translator.c:2423]: build_res_buf_from_sip_req(): out of memory; needs 514 ... <core> [data_lump.c:172]: insert_new_lump_before(): out of memory acc [acc_extra.c:267]: extra2strar(): extra2strar: out of memory. after a kamailio reboot evrethings works again for 5/6 days. what can I do to resolve?

3 3

RPC tcp limit buffer
by pars3c 28 Sep '16

28 Sep '16

Hello, i'm trying to list the dialog (dlg_profile list) in the proxy using an RPC call to jsonrpc-s (dlg_profile list). But when there are many calls , kamailio send an error because the tcp buffer is not big enough. The only solution is to increase the parameter tcp_conn_wq_max every time i've an error ? Thanks

2 2

Accessing cached database table
by Daniel W. Graham 28 Sep '16

28 Sep '16

Is there a way to access a cached database table from the config file, without using a function from a module? Example - Lookup ip_addr from 'address' table or src_ip from 'trusted' table based on the 'tag' column. No function exists in permissions module to do this, but running 'kamctl address show ' or 'kamctl trusted show ', show that the tables along with tag column are cached. Thanks! -Dan

2 2

invalid sendtoparameters
by Linux Vince 27 Sep '16

27 Sep '16

I am getting this error on every outgoing request. Below are log entries i found Sep 27 22:09:06 centos1 /usr/sbin/kamailio[22421]: ERROR: <core> [udp_server.c:550]: udp_send(): sendto(sock,0x7fb76f5da5e0,1568,0, 10.3.4.2:5060,16): Invalid argument(22) Sep 27 22:09:06 centos1 /usr/sbin/kamailio[22421]: CRITICAL: <core> [udp_server.c:555]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 27 22:09:06 centos1 /usr/sbin/kamailio[22421]: ERROR: <core> [forward.h:201]: msg_send_buffer(): udp_send failed SNDTO values are SNDTO ,10.3.4.2 PORT 5060 I have confirmed the PID and it is bound to my public IP 22417 main process - attendant 22418 udp receiver child=0 sock=10.3.4.1:5060 22419 udp receiver child=1 sock=10.3.4.1:5060 22420 udp receiver child=2 sock=10.3.4.1:5060 22421 udp receiver child=3 sock=10.3.4.1:5060 22422 udp receiver child=4 sock=10.3.4.1:5060 22423 udp receiver child=5 sock=10.3.4.1:5060 22424 udp receiver child=6 sock=10.3.4.1:5060 22425 udp receiver child=7 sock=10.3.4.1:5060 22426 udp receiver child=0 sock=127.0.0.1:5060 22427 udp receiver child=1 sock=127.0.0.1:5060 22428 udp receiver child=2 sock=127.0.0.1:5060 22429 udp receiver child=3 sock=127.0.0.1:5060 22430 udp receiver child=4 sock=127.0.0.1:5060 22431 udp receiver child=5 sock=127.0.0.1:5060 22432 udp receiver child=6 sock=127.0.0.1:5060 22433 udp receiver child=7 sock=127.0.0.1:5060 22434 slow timer 22435 timer 22436 secondary timer 22437 ctl handler 22438 MI FIFO 22439 WEBSOCKET KEEPALIVE 22440 tcp receiver (generic) child=0 22441 tcp receiver (generic) child=1 22442 tcp receiver (generic) child=2 22443 tcp receiver (generic) child=3 22444 tcp receiver (generic) child=4 22445 tcp receiver (generic) child=5 22446 tcp receiver (generic) child=6 22447 tcp receiver (generic) child=7 22448 tcp main process How do i identify the issue ?

1 1

SIREMIS MANUAL
by Carlyle Edwards 27 Sep '16

27 Sep '16

Hi, is there a manual for SIREMIS?

2 1

Kamailio IMS NAT issue: PRACK/ACK RURI changing in P-CSCF
by Owais 26 Sep '16

26 Sep '16

Hi all, My scenario is as follows: I have All the CSCFs on the same IP using different ports. P-CSCF advertises a public IP. Rest of the nodes including I-CSCF and S-CSCF listen local only. Behavior: When an ACK or PRACK goes from P-CSCF to S-CSCF, the RURI changes such that alias parameter is removed. Hence, P-CSCF ends up sending PRACK and ACK on private IP of UE. Here's what the config looks like: Any hints? -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-IMS-NAT-issue-PRACK-ACK-RU… Sent from the Users mailing list archive at Nabble.com.

2 2

REGISTER between Kamailio 4.4 and Asterisk 13
by Grant Bagdasarian 26 Sep '16

26 Sep '16

Hello, We're trying to get our Asterisk test server to REGISTER with Kamailio. The first attempts are ok, but once the nonce timer is exceeded Kamailio starts rejecting the REGISTER requests from Asterisk, due to the nonce being expired. The auth_check function is the one returning error -4. Is this an issue in Asterisk or Kamailio? Which component is responsible for generating a new nonce value? I'm a bit confused regarding the flow of SIP REGISTER. I hope someone could help me out on this. Thanks! Regards, Grant Bagdasarian

2 2

Config changes needed from 4.4.2 to 4.4.3 (and a bug)
by Sebastian Damm 26 Sep '16

26 Sep '16

Hi, Kamailio release 4.4.3 was tagged as a "bug fix release", no config changes should be needed. However, we had to change two things in our config file to get 4.4.3 running. I thought, I might share those on the list. 1) Defines must not have '-' in the names. In 4.4.2 we had a Define like this: #!define NO-FRAUDCHECK This threw an error when trying to restart with 4.4.3. Replacing it with #!define NO_FRAUDCHECK fixed the problem. 2) Module Order problems. We introduced the usage of http_async_client module a few weeks ago, and this worked perfectly. However, 4.4.3 complained that the pv module was not available before loading http_async_client. After moving up the pv module above http_async_client, everything worked again. 3) There's a bug in uac_auth function. I've opened an issue for that: https://github.com/kamailio/kamailio/issues/797 Best Regards, Sebastian

2 1

Textops replace question
by Igor Olhovskiy 26 Sep '16

26 Sep '16

Hi! I’m trying to use replace function, but got strange results. Idea - modify Contact header What I got as start packet. (202 Accepted on onreply_route) … Contact: <sip:*97@10.0.20.81:5060> … I give a command replace("Contact: <.*>», "Contact: <" + $fu + ">"); $fu = sip:11@master.rufan.at And, as a result I got Contact: <sip:11@master.rufan.at>sip:*97@10.0.20.81:5060 Why it’s adding old Contact part at the end? What can cause this? On onreply_route I have only this command. Thanks! -- Best regards, Igor

3 4

Dialog sharing between Kamailio servers
by Michael Jepson 26 Sep '16

26 Sep '16

Hi all, I am trying to limit the concurrent calls of a customer, but we have multiple Kamailio servers in our platform. I have the dialog module up and running and it is storing its data in a database, but when I use get_profile_size, it will only return the number of dialogs in that profile for the current machine, it does not add the others. How can I get Kamailio to add the others, i.e. how can I get Kamailio to consult the database to get the correct size. Now, Kamailio seems to only write to the database, but not read from it. Best regards, MICHAEL JEPSON

2 1

Load balancing with Kamailio
by NITESH BANSAL 26 Sep '16

26 Sep '16

Hello, I'm planning on doing some smart load balancing with Kamailio. We have a distributed network, with multiple Kamailio boxes in different locations serving as Ingress SBC, these Kamailio boxes are the entry point for a SIP call and then they route the call to a pre-configured Asterisk boxes. I want to move away from this, I would like these Kamailios to be able to distribute the traffic to Asterisk boxes based on the actual load on these boxes, the goal is to be more dynamic? Is there any Kamailio module which could do that? Do I need to integrate some other tool(Homer etc) with Kamailio to achieve this? Any suggestions are welcome.

3 3

LCR routing, and finaly route to media gateway if not found
by Loic Chabert 26 Sep '16

26 Sep '16

Hello List, I'm finishing my implementation of LCR with kamailio for outgoing calls but one use case does not work. Let me explain: - I have some sbc for one destination. If one destination fail, kamailio try to second one etc.. - If no SBC is available for this destination (for exemple, all are down), i want to send my call to an error server (asterisk), play an error sound. To accomplish this, i re-write the RURI: "sip:error1@errorserver.local", and launch RELAY route. But kamailio dont make a DNS query to find IP of "errorserver.local", only change RURI and try to send to the last LCR gateway. I have also try to add a new branch, but same result... Why kamailio do not make a new DNS query ? *route[RELAY] { #if (is_method("INVITE")) { # if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); #} if (!t_relay()) { sl_reply_error(); } exit;}route[NOTFOUND]{ $ru = "sip:error1@errorserver.local"; xlog("L_WARN", "[$ci]|$si|Routing this call to $ru (error server)\n"); append_branch($ru); route(RELAY);}route[LCR] { xlog("L_INFO", "[$ci]|$si|Loading gateways...\n"); if (!load_gws(1)) { xlog("L_ERR", "[$ci]|$si|Can't load LCR gateways\n"); sl_send_reply("503", "Unable to load gateways"); exit; } else { $var(i) = 0; while(is_avp_set("$(avp(i:709)[$var(i)])")) { xlog("L_INFO", "[$ci]|$si|This gateway was found: gw_uri_avp[$var(i)]=$(avp(i:709)[$var(i)]) \n"); $var(i) = $var(i) + 1; }; if(is_avp_set("$avp(i:709)")) { xlog("L_INFO", "[$ci]|$si|Trying gateway ‘$avp(i:709)’\n"); } else { xlog("L_INFO", "[$ci]|$si|No more gateways. Goto route NOTFOUND...\n"); route(NOTFOUND); }; # try the first matched gateway if (next_gw()) { xlog("L_INFO", "[$ci]|$si|New request URI built: $ru . Relaying it (and arming backup destination if needed)."); # Route to failure for failover t_on_failure("lcrfailure"); route(RELAY); } else { xlog("L_INFO", "[$ci]|$si|No more gateways (after next_gw()). Goto route NOTFOUND..."); route(NOTFOUND); }; }; exit;}failure_route[lcrfailure] { # the previous gateway is no good if (t_check_status("408|50[34]")) { xlog("L_WARN", "[$ci]|$si|Entering in entering failure_route[lcrfailure]. Error code: $T_reply_code \n"); if(is_avp_set("$avp(i:709)")) { xlog("L_INFO", "[$ci]|$si|Trying gateway ‘$avp(i:709)’\n"); } else { xlog("L_ERR", "[$ci]|$si|AVP is not set\n"); }; if (next_gw()) { xlog("L_INFO", "[$ci]|$si|New request URI built: $ru . Relaying it (and arming backup destination if needed)."); # Route to failure for failover t_on_failure("lcrfailure"); route(RELAY); } else { xlog("L_INFO", "[$ci]|$si|No more gateways (after next_gw()). Goto route NOTFOUND..."); route(NOTFOUND); }; exit; };}* Many thanks, Regards.

3 2

how to config gateway in kamailio
by Gaurav Bmotra 26 Sep '16

26 Sep '16

Hi i want to config gateway in kamailio i want to know how can i config IP Authenticate gateway and user/ pass gateway -- *Regards:* Gaurav Bmotra

2 1

Scripting Issue
by aman jeff 26 Sep '16

26 Sep '16

Hai Team, I can't able to set maxptime in SDP Please help me

1 1

Location Table In db_mode 3
by Gholamreza Sabery 25 Sep '16

25 Sep '16

I have multiple Kamailio servers in an Active/Active scenario (the database is a MySQL server) using db_mode 3. But when the number of clients goes up writes on the location table become too much and it causes database problems. I even put my location table on a separate SSD disk on the server to separate it's IO but still at times I have problems. How can I move location table into memory? Is this a good solution? What other options do I have?

4 12

how to config gateway in kamailio
by Bmotra Infotech 24 Sep '16

24 Sep '16

i want to config gateway in kamailio i want to know how can i config IP Authenticate gateway and user/ pass gateway Bmotra Infotech

2 1

Re: [SR-Users] LCR routing, and finaly route to media gateway if not found
by Loic Chabert 23 Sep '16

23 Sep '16

In fact, kamailio is out from lcr at this point, no more gateway to try (all gateways are down). I just want to send call to a static server for playing sound. To be more specific, if no gateways are not set on database, a sound must be played. Should i change $du and $ru to make it work ? Le 23 sept. 2016 4:36 PM, "Juha Heinanen" <jh(a)tutpro.com> a écrit : Loic Chabert writes: > To accomplish this, i re-write the RURI: "sip:error1@errorserver.local", > and launch RELAY route. But kamailio dont make a DNS query to find IP of > "errorserver.local", only change RURI and try to send to the last LCR > gateway. > > I have also try to add a new branch, but same result... Why kamailio do not > make a new DNS query ? check if both hostname and ip address are given for the gateway. if so, next_gw() also sets destination uri. -- juha _______________________________________________ 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

Kamailio ims nat issue: PRACK/ACK ruri changing in p-cscf
by Owais 23 Sep '16

23 Sep '16

Hi all, My scenario is as follows: I have All the CSCFs on the same IP using different ports. P-CSCF advertises a public IP. Rest of the nodes including I-CSCF and S-CSCF listen local only. Behavior: When an ACK or PRACK goes from P-CSCF to S-CSCF, the RURI changes such that alias parameter is removed. Hence, P-CSCF ends up sending PRACK and ACK on private IP of UE. Here's what the config looks like: #!define WITH_NAT Any hints? -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-ims-nat-issue-PRACK-ACK-ru… Sent from the Users mailing list archive at Nabble.com.

1 0

604 Does not exist anywhere - HSS User Unknown
by Rodrigo Moreira 23 Sep '16

23 Sep '16

Hi, After I am registred on Kamailio a try to make a call (INVITE), and S-CSCF returns the messege: 604 Does not exist anywhere - HSS User Unknown What I am doing wrong? Is the contact entry (Alice) wrong? How should the contact entry on the list? sip:alice@mydomain.com Best Regards., -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

3 3

Kamailio Not Starting
by Linux Vince 23 Sep '16

23 Sep '16

I am getting below error when i start kamailio Not starting : invalid configuration file! 0(2360) ERROR: <core> [mem/f_malloc.c:415]: fm_search_defrag(): fm_search_defrag(0x7f732c5ff010, 232); Free fragment not found! 0(2360) ERROR: <core> [mem/f_malloc.c:498]: fm_malloc(): fm_malloc(0x7f732c5ff010, 232) called from core: route_struct.c: mk_action(118), module: core; Free fragment not found! 0(2360) CRITICAL: <core> [route_struct.c:137]: mk_action(): memory allocation failure 0(2360) ERROR: <core> [mem/f_malloc.c:415]: fm_search_defrag(): fm_search_defrag(0x7f732c5ff010, 40); Free fragment not found! 0(2360) ERROR: <core> [mem/f_malloc.c:498]: fm_malloc(): fm_malloc(0x7f732c5ff010, 40) called from core: cfg.y: yyparse(2636), module: core; Free fragment not found! 0(2360) : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/proxy.cfg, line 5372, column 3-28: Not enough memory ERROR: bad config file (1 errors) I tried increasing SHM_MEMORY and PKG_MEMORY under /etc/default/kamailio I doubled memory but still getting same error. Is there any other place i should look at ?

2 3

Kamailio ims nat issue: PRACK/ACK ruri changing in p-cscf
by Owais Ahmad 23 Sep '16

23 Sep '16

Hi all, My scenario is as follows: I have All the CSCFs on the same IP using different ports. P-CSCF advertises a public IP. Rest of the nodes including I-CSCF and S-CSCF listen local only. Behavior: When an ACK or PRACK goes from P-CSCF to S-CSCF, the RURI changes such that alias parameter is removed. Hence, P-CSCF ends up sending PRACK and ACK on private IP of UE. Here's what the config looks like: listen=udp:lo.ca.l.ip:6000 advertise pu.bl.ic.ip:6000 alias=pcscf-ims.domain.net #!define WITH_NAT Any hints?

1 0

Problem reseting TMX statistics
by Átila de Moura Tavano Moretto 23 Sep '16

23 Sep '16

I'm currently using version 4.4.2 and am having trouble trying to reset the statistics in the TMX module. According to its documentation, all statistics except "inuse_transactions" can be reset. Checking the tmx_mod.c files for the statistics definition, I saw that the STAT_NO_RESET flag wasn't used for any statistic listed there (which seems odd because I think it should have been set for the inuse_transactions statistic): static stat_export_t mod_stats[] = { {"UAS_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_uas_trans }, {"UAC_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_uac_trans }, {"2xx_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_2xx }, {"3xx_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_3xx }, {"4xx_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_4xx }, {"5xx_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_5xx }, {"6xx_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_6xx }, {"inuse_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_inuse }, {"active_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_active}, {"rpl_received", STAT_IS_FUNC, (stat_var**)tmx_stats_rcv_rpls }, {"rpl_absorbed", STAT_IS_FUNC, (stat_var**)tmx_stats_abs_rpls }, {"rpl_generated", STAT_IS_FUNC, (stat_var**)tmx_stats_rld_loc_rpls}, {"rpl_relayed", STAT_IS_FUNC, (stat_var**)tmx_stats_rld_rcv_rpls}, {"rpl_sent", STAT_IS_FUNC, (stat_var**)tmx_stats_rld_tot_rpls}, {0, 0, 0} }; All seems ok, but when I try to reset the stats, the command simply doesn't work. The values remain unaltered. The clear command doesn't work either, it just returns the current values. Can anyone help me trying to debug this problem, I'm very new to Kamailio and its code, and don't have a good understanding of how it the statistics gathering/reset works. Thanks. Atila Moretto

1 0

Location Table In db_mode 3
by Gholamreza Sabery 23 Sep '16

23 Sep '16

I have multiple Kamailio servers in an Active/Active scenario (the database is a MySQL server) using db_mode 3. But when the number of clients goes up writes on the location table become too much and it causes database problems. I even put my location table on a separate SSD disk on the server to separate it's IO but still at times I have problems. How can I move location table into memory? Is this a good solution? What other options do I have?

1 0

Dimension number of child
by Igor Potjevlesch 23 Sep '16

23 Sep '16

Hello, I'm wondering if there's a best practise to dimension the number of child to setup? I see, very often, Kamailio that stops to reply to SIP traffic for couple a seconds. This cause some calls drop if RE-INVITE are not processed for example. So, I'm wondering if the issue is not due to a lack of child for processing the SIP traffic. Thank you for your inputs. Regards, Igor.

1 0

failed to find command http_async_query - Even though module loaded
by Jonathan Hunter 23 Sep '16

23 Sep '16

Hi guys, I am using the http_async_client module to POST some requests to a Web Interface, and on server 1 it all works well. I have done a fresh install from github, version 4.4 on Centos 6.8 on server 2, and kamailio wont start. Its the same configuration file, however it states; failed to find command http_async_query (params 3) However the module is loaded, and when installing no errors were written when creating the module, and in terms of its links they are below and look ok to me. Is there anywhere else I should be looking? Thanks Jon ldd http_async_client.so linux-vdso.so.1 => (0x00007ffecedf1000) libcurl.so.4 => /usr/lib64/libcurl.so.4 (0x00007fdc5bd63000) libevent-2.0.so.5 => /usr/lib64/libevent-2.0.so.5 (0x00007fdc5bb1c000) libsrutils.so.1 => /usr/local/kamailio-4.4/lib64/kamailio/libsrutils.so.1 (0x00007fdc5b90c000) libkcore.so.1 => /usr/local/kamailio-4.4/lib64/kamailio/libkcore.so.1 (0x00007fdc5b6ff000) libc.so.6 => /lib64/libc.so.6 (0x00007fdc5b36a000) libidn.so.11 => /lib64/libidn.so.11 (0x00007fdc5b138000) libldap-2.4.so.2 => /lib64/libldap-2.4.so.2 (0x00007fdc5aee7000) librt.so.1 => /lib64/librt.so.1 (0x00007fdc5acde000) libgssapi_krb5.so.2 => /lib64/libgssapi_krb5.so.2 (0x00007fdc5aa9a000) libkrb5.so.3 => /lib64/libkrb5.so.3 (0x00007fdc5a7b3000) libk5crypto.so.3 => /lib64/libk5crypto.so.3 (0x00007fdc5a586000) libcom_err.so.2 => /lib64/libcom_err.so.2 (0x00007fdc5a382000) libz.so.1 => /lib64/libz.so.1 (0x00007fdc5a16c000) libssl3.so => /usr/lib64/libssl3.so (0x00007fdc59f28000) libsmime3.so => /usr/lib64/libsmime3.so (0x00007fdc59cfb000) libnss3.so => /usr/lib64/libnss3.so (0x00007fdc599bb000) libnssutil3.so => /usr/lib64/libnssutil3.so (0x00007fdc5978e000) libplds4.so => /lib64/libplds4.so (0x00007fdc5958a000) libplc4.so => /lib64/libplc4.so (0x00007fdc59385000) libnspr4.so => /lib64/libnspr4.so (0x00007fdc59146000) libpthread.so.0 => /lib64/libpthread.so.0 (0x00007fdc58f29000) libdl.so.2 => /lib64/libdl.so.2 (0x00007fdc58d25000) libssh2.so.1 => /usr/lib64/libssh2.so.1 (0x00007fdc58afc000) libm.so.6 => /lib64/libm.so.6 (0x00007fdc58878000) /lib64/ld-linux-x86-64.so.2 (0x00007fdc5c1f8000) liblber-2.4.so.2 => /lib64/liblber-2.4.so.2 (0x00007fdc58668000) libresolv.so.2 => /lib64/libresolv.so.2 (0x00007fdc5844e000) libsasl2.so.2 => /usr/lib64/libsasl2.so.2 (0x00007fdc58234000) libkrb5support.so.0 => /lib64/libkrb5support.so.0 (0x00007fdc58028000) libkeyutils.so.1 => /lib64/libkeyutils.so.1 (0x00007fdc57e25000) libssl.so.10 => /usr/lib64/libssl.so.10 (0x00007fdc57bb8000) libcrypto.so.10 => /usr/lib64/libcrypto.so.10 (0x00007fdc577d4000) libcrypt.so.1 => /lib64/libcrypt.so.1 (0x00007fdc5759d000) libselinux.so.1 => /lib64/libselinux.so.1 (0x00007fdc5737d000) libfreebl3.so => /lib64/libfreebl3.so (0x00007fdc5717a000)

2 2

Selective rport behaviour
by Alex Balashov 23 Sep '16

23 Sep '16

Hi, In some scenarios, it happens that a request is received with the following characteristics: IP xxx.xxx.xxx.xxx:14000 -> server:5060 Via: SIP/2.0/xxx.xxx.xxx.xxx:5060;branch=9hG4bKblah;rport Normally, we just force_rport() on all incoming requests so that we reply to the real source port of the request, since most endpoints on this installation are NAT'd. However, occasionally we run into a scenario where an ALG or misconfigured client incorrectly inserts an rport attribute into its topmost Via, and really expects to receive a response at the address/port indicated in the Via (i.e. 5060). Does Kamailio offer a means of dealing with these on a selective basis? Would refraining from calling force_rport() be enough? Or would it be necessary to set reply_to_via=1 as well, thus breaking symmetrical behaviour for the vast majority of the NAT'd endpoints? In other words, I'm not 100% clear on the following: 1) What impact does force_rport() have if an 'rport' attribute is sent by the client? In this case, there should be nothing to "force"; I assume that if the 'rport' attribute is placed by the client, then the proxy will return replies to the source port of the request even if force_rport() is not called, because that's the RFC 3581-compatible thing to do. Right? 2) Does reply_to_via=1 override the behaviour hypothesised in #1? 3) Does reply_to_via=1 override force_rport()? Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 3

Killing dialog using kamcmd
by Laura 22 Sep '16

22 Sep '16

Dear list, i have developed a tool that automatically stop the call after some trigger i use.. For do that i use the kamcmd dlg.end_dlg command. It work perfectly but i have noticed a problem over the kamailio acc table. When i kill a specific call using this command.. the acc table don't write the BYE record for the dialog.. Is that correct ? how can i fix.. I'm using ngrep for verify the SIP dialog and i see the BYE command correctly on the A and B leg of the call.. Thanks for any idea. Laura

2 2

ndb_mongodb -Sending Query to Mongo
by Jonathan Hunter 22 Sep '16

22 Sep '16

Hi Guys, I am using the ndb_mongodb module, and Im just trying to return values from the dispatcher table, in particular the setid of a particular entry depending on its attrs value. If I query using mongodb shell command it works fine, here I enter test.com and it returns only the setid value, or 2; db.dispatcher.find( { attrs: "test.com" }, { setid: 1, _id: 0 } ){ "setid" : 2 } However I have been having issues getting this to work using mongodb_find_one, as I would of thought the following route return the same; if(mongodb_find_one("ml-mongodb", "kamailiosbc", "dispatcher", "{ \"attrs\" : \"test.com\" },{ \"setid\": \"1\", \"_id\": \"0\" }", "mgr1")) However it returns all the entries relating to that attrs columm from dispatcher table, instead of just setid. Is it possible to do this? I am stuggling to find the json format to acheive the same thing. Also in terms of the mgr1 response, I see its contained within the $mongodb(key), can I drill down to get specific fields from it or would I need to use transformations or jansson to do this? Many thanks Jon

2 2

STUN server - stun_allow_stun
by Hermann Norpois 22 Sep '16

22 Sep '16

Hello, I have some STUN questions. My plan is to build a P2P connection with ICE (psua2 or libnice). 1) According to the cookbook ( http://www.kamailio.org/wiki/cookbooks/4.0.x/core) one can activate an internal STUN server: stun_allow_stun = 0 | 1 (off | on); default 1 This is followed by: "Note: STUN support is only available if sip-router/Kamailio was compiled with STUN support (“make STUN=1 cfg”)" I did an installation with sudo apt-get ... Is STUN support availbable? How do I check for STUN support? 2) I ICE it seems to be usual that the body of the SIP request contains an SDP message that includes the IP addresses and ports to be used. So the results of the STUN request (and TURN request possibily) are already included. How does the kamailio STUN server work? Is it an independent STUN server (caller asks the kamailio STUN server - gets an answer - encodes an SDP with the external IP adress an port information - starts the SIP INVITE) ? Or does Kamailio change SDP lines after contacting the internal STUN server? 3) Is the kanailio STUN server prefered to other STUN servers, for instance stun.stunprotocol.org? Thanks Hermann

2 1

Problem using ds_is_from_list in failure_route
by Joel Serrano | VOZELIA 21 Sep '16

21 Sep '16

Hi all, I'm facing a strange problem using ds_is_from_list in the failure_route block. I have 4 servers in a dispatcher group 8201. Scenario: A call comes in, gets routed to a backend server from 8201 dispatcher group, the backend replies with a 480. Then we have: onreply_route[MANAGE_REPLY] { [...] if (ds_is_from_list("8201")) && t_check_status("480") { <<<<<< This gets executed, because the node is in 8201 and the reply was a 480 if (is_present_hf("X-UserID")) { $var(UserId) = $hdr(X-UserID); } } [...] } So far so good, next is: failure_route[MANAGE_FAILURE] { [...] if (ds_is_from_list("8201")) && (t_check_status("480")) { <<<<<< This does NOT get executed. append_hf("X-UserID: $var(UserId)\r\n"); if (!ds_select_domain("8205","4")) { xlog("L_ERR", "[$ci] - No additional gateways\n"); t_reply("502","Bad Gateway"); exit; } t_set_fr(0,2000); route(RELAY); } [...] } To find the problem, I added some logging in the failure_route before entering the IF statement: if (ds_is_from_list("8201")) xlog("L_NOTICE", "DS is in group 8201"); <<<<< No log, so did NOT match. if (t_check_status("480")) xlog("L_NOTICE", "Status is 480"); <<<<< Yes log, so DID match. According to the docs, ds_is_from_list can be used in any route block, so I think I am definitely doing something wrong... or maybe it's a bug?? The idea is to achieve this: Call comes in from softphone into proxy, proxy sends it to backend, if backend replies with a 480, send it to another dispatcher set (8205). If the 480 comes from somewhere that is not the backend (8201), then do nothing. Any suggestions? Thanks in advanced. Best regards, Joel. Este mensaje y, en su caso, los ficheros anexos son confidenciales, especialmente en lo que respecta a los datos personales, y se dirigen exclusivamente al destinatario referenciado.Les informamos que los datos contenidos en el presente mail, han sido facilitados por usted o recogidos de Fuentes Accesibles al Público. Dicha información personal, será incluida en un fichero bajo la responsabilidad de VOZELIA TELECOM S.L.Todos los ficheros que trata nuestra empresa se encuentran debidamente inscritos ante la Agencia Española de Protección de Datos.Vd. Como propio interesado podrá ejercitar sus derechos de Acceso, Cancelación, Rectificación u Oposición, en el mail: administracion(a)vozelia.com o en la dirección Avenida de Manoteras 8 escalera 3 2ºB, 28050 Madrid.Tal y como establece la Ley Orgánica de Protección de Datos 15/1999 de 13 de Diciembre.Para ejercer sus derechos desde Panama podrá hacerlo en la dirección Vía Argentina, Edificio Mediteranean Loft. Panama.En caso de no ser el destinatario y haya recibido este mensaje por error, agradeceremos que nos lo comunique inmediatamente al remitente sin difundir, almacenar o copiar su contenido.La divulgación o el suministro, en todo o en parte, a cualquier tercero, no podrá ser realizada sin el previo, expreso y escrito consentimiento de VOZELIA TELECOM S.L. .

2 3

Error 2002 (HY000) creating database kamailio
by Diogenes Marcano 21 Sep '16

21 Sep '16

Iam new in kamailio. When I try to create the data base with kamdbctl y have the following error root@diogenes-Inspiron-1525:/usr/local/sbin# kamdbctl create MySQL password for root: INFO: test server charset ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2 "No such file or directory") ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2 "No such file or directory") Usage: grep [OPTION]... PATTERN [FILE]... Try 'grep --help' for more information. /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 111: [: =: unary operator expected INFO: creating database kamailio ... ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2 "No such file or directory") ERROR: Creating database kamailio failed! root@diogenes-Inspiron-1525:/usr/local/sbin# mysql ERROR 2002 (HY000): Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2 "No such file or directory") Thank you for your hrlp Diogenes Marcano

2 1

parameter not found in module dialog
by Michael Jepson 21 Sep '16

21 Sep '16

Hi all, I am trying to limit concurrent calls, using dialog, or rather, I want to try it. I have now only added loadmodule "dialog.so" and a series of modparam commands to my config file and already I am running into problems: ERROR: <core> [modparam.c:139]: set_mod_param_regex(): parameter <toroute_column> of type <1> not found in module <dialog> It fails on the last modparam line, but if I remove it (by commenting it), it fails on whatever the latest modparam line is after that. I have tried removing multiple calls, but every time it fails on whatever is the last line. I have the required modules already loaded (tm.so, rr.so and pv.so), so that's not it I guess. Any idea what might be causing this issue? Also, I noticed (when calling locate dialog.so on the Linux machine) that there is also an ims_dialog.so, if I replace dialog with ims_dialog, I run into the exact same issues. Any help is greatly appreciated! Best regards, Michael Jepson CM Telecom

2 1

Kamailio invite fails with 403 at cscf
by Owais Ahmad 20 Sep '16

20 Sep '16

I am using kamailio 4.4.2 IMS Registration completes successfully, but S-CSCF gives the following error when P-CSCF relays INVITE to S-CSCF: 32(1349) ERROR: ims_dialog [dlg_handlers.c:190]: populate_leg_info(): bad sip message or missing Contact hdr 32(1349) ERROR: ims_dialog [dlg_handlers.c:1352]: dlg_onreply(): could not add further info to the dlg out Resultantly, P-CSCF returns a 403 with: "Forbidden - You must register first with a S-CSCF" It appears that pcscf_is_registered("location") is unable to lookup the location although the database seems to have valid entries. Any hints on how I can get the INVITE processed without this issue? Regards, Owais

3 4

.NET Core & Kamailio 5.0
by Grant Bagdasarian 20 Sep '16

20 Sep '16

Hi, I was reading some docs on Kamailio 5 and saw in the ToDo list an item about implementing the dynamic object mechanism for Mono. Would it be possible to implement it directly for .NET Core, since the official .NET support has been added to Linux some time ago? Regards, Grant

1 0

event_route[uac:reply] question
by Julia Boudniatsky 20 Sep '16

20 Sep '16

I send INVITE via uac_send_req() and receive "183" and "200"responses. Only "200" appears in the log of event_route[uac:reply]. Is event_route[uac:reply] executed only for final responses? Thank you, Julia

3 5

OOM Errors
by Phil Lavin 20 Sep '16

20 Sep '16

Hi all, Kamailio stopped performing some functions earlier today, citing OOM errors. An excerpt from the log is below. SHM usage, at the time, looked fine. Also see below for that output. A restart of Kamailio resolved the issue. Is this SHM related or OS memory manager related? root@ua-proxy-01:~# kamctl stats shmem shmem:fragments = 910 shmem:free_size = 104131224 shmem:max_used_size = 51590800 shmem:real_used_size = 30086504 shmem:total_size = 134217728 shmem:used_size = 19558800 Sep 19 11:35:56 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [data_lump.c:161]: insert_new_lump_before(): out of memory Sep 19 11:40:10 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [msg_translator.c:2390]: build_res_buf_from_sip_req(): out of memory; needs 423 Sep 19 11:40:16 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [msg_translator.c:2390]: build_res_buf_from_sip_req(): out of memory; needs 453 Sep 19 11:42:46 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [data_lump.c:136]: insert_new_lump_after(): out of memory Sep 19 11:42:46 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [ut.c:163]: as_asciiz(): Out of memory Sep 19 11:42:57 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [ut.c:163]: as_asciiz(): Out of memory Sep 19 11:43:12 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [data_lump.c:361]: anchor_lump(): out of memory Sep 19 11:43:12 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [data_lump.c:361]: anchor_lump(): out of memory Sep 19 11:43:12 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [data_lump.c:306]: del_lump(): out of memory Sep 19 11:43:12 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [data_lump.c:306]: del_lump(): out of memory Sep 19 11:43:12 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [msg_translator.c:326]: received_builder(): out of memory Sep 19 11:43:12 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [msg_translator.c:326]: received_builder(): out of memory Sep 19 11:43:12 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [msg_translator.c:326]: received_builder(): out of memory Sep 19 11:43:16 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [data_lump.c:136]: insert_new_lump_after(): out of memory Sep 19 11:43:16 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [ut.c:163]: as_asciiz(): Out of memory Sep 19 11:44:18 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [data_lump.c:161]: insert_new_lump_before(): out of memory Sep 19 11:45:07 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [ut.c:163]: as_asciiz(): Out of memory Sep 19 11:45:30 ua-proxy-01 /sbin/kamailio[20218]: ERROR: acc [acc_extra.c:267]: extra2strar(): extra2strar: out of memory. Sep 19 11:45:43 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [data_lump.c:186]: insert_subst_lump_after(): out of memory Sep 19 11:45:52 ua-proxy-01 /sbin/kamailio[20218]: ERROR: <core> [data_lump.c:161]: insert_new_lump_before(): out of memory

2 10

Presence: Empty reginfo NOTIFY bodies
by Stefan Kohlhauser 19 Sep '16

19 Sep '16

Hey everyone! Problem: At startup the presence module for reginfos can get into a state where it is not adding bodies to reginfo NOTIFYs. Setup: We are using the pua_reginfo to let the Kamailio (4.2.6) PUBLISH the current registration state to itself. The body is correctly added to the presentity-table in the Postgres. Analysis: However, we have identified two cases where after startup the Kamailio can reach a state where he won't add bodies to the NOTIFYs. And this does not fix itself over time. We are pretty confident that the problem lies in an inconsistency between the presentity table in the DB and the hashtable of the presence module. Case 1: The entry for a URI in the DB is expired. On startup of the Kamailio the hashtable remains empty for that URI. Maybe due to another bug the pua_reginfo occasionally sends PUBLISHes with an expired "SIP-If-Match"-header. In module/presence/presentity.c::update_presentity() the DB is updated. However, the hashtable is only updated if it is a new request (i.e. no "SIP-If-Match"-header and thus new_t not set). The hash table is not filled on startup, not on the first PUBLISH as it contains a (wrong) "SIP-If-Match"-header, and not on any subsequent PUBLISH as they contain correct "SIP-If-Match"-headers. Case 2: The presentity table contains at least one expired and one non-expired reginfo entry for a URI. The Kamailio seems to send a NOTIFY with a body within about a minute (probably the default clean period of 100 secs?). But after that short interval only NOTIFYs with empty bodies are sent. To us it looks like the cleanup function (module/presence/publish.c::msg_presentity_clean()) is cleaning up the expired DB entries correctly. But despite one still valid entry in the DB the entry in the hash table gets removed as well. And it wont get filled again (see case 1). This only seems to apply to the "reg" event, the "presence" event does not seem to be affected. We are aware aware of the publ_cach parameter, but we don't know if this has any performance impact on our system. If needed we can try to provide more insight on this problem. Some help on this or a fix would be much appreciated! BR, Stefan

1 0

Kamailio P-CSCF error on INVITE
by Owais 19 Sep '16

19 Sep '16

Hi guys, Kamailio P-CSCF is unable to return true for: pcscf_is_registered("location") on INVITE and responds to the INVITE with: 403 Forbidden - You must register first with a S-CSCF Registration completes fine and I can see database with valid location entries though... Any ideas? Regards, Owais -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-P-CSCF-error-on-INVITE-tp1… Sent from the Users mailing list archive at Nabble.com.

2 3

how to access reply headers
by mjbemail2000-seti＠yahoo.co.uk 19 Sep '16

19 Sep '16

Hi, I have a registered handset which replies to an INVITE message from Kamailio with a "302 Moved Temporarily" message. The reason for this is because the user has set their handset to forward the call to another telephone number. In a wireshark trace I can see the forwarded telephone number in the Contact header of the SIP message but I can't find a way to access this header via any variables. In the config file I have tried this: # Manage incoming repliesonreply_route[MANAGE_REPLY] { ... if(t_check_status("3[0-9][0-9]")) { xlog("L_INFO", "test1: $hdr(Contact)"); xlog("L_INFO", "test2: $ct"); } ... } , but in both cases the value returned is the Contact header of the original INVITE message and not the Contact shown in the "302 Moved Temporarily" reply message. How do I access the Contact header in the reply message? Regards, Mark.

2 1

Debian 4.4.3 reproducable crashes in rtpengine.so
by Daniel Tryba 19 Sep '16

19 Sep '16

With kamctl fifo nh_enable_rtpp udp:rptipaddr:7723 1 kamailio will begin to crash: Sep 19 16:00:08 sipcluster-backend2 kernel: [1658593.859081] kamailio[22992]: segfault at 4 ip 00007f5a4d44e11f sp 0000 7ffcac5293e0 error 4 in rtpengine.so[7f5a4d446000+41000] Sep 19 16:00:12 sipcluster-backend2 /usr/sbin/kamailio[22999]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 2 8 Sep 19 16:00:12 sipcluster-backend2 /usr/sbin/kamailio[22976]: ALERT: <core> [main.c:740]: handle_sigs(): child process 22992 exited by a signal 11 Sep 19 16:00:12 sipcluster-backend2 /usr/sbin/kamailio[22976]: ALERT: <core> [main.c:743]: handle_sigs(): core was gene rated Sep 19 16:00:12 sipcluster-backend2 /usr/sbin/kamailio[22976]: INFO: <core> [main.c:755]: handle_sigs(): terminating du e to SIGCHLD Sep 19 16:00:12 sipcluster-backend2 /usr/sbin/kamailio[22978]: INFO: <core> [main.c:810]: sig_usr(): signal 15 received .... Backtrace, Program terminated with signal SIGSEGV, Segmentation fault. #0 send_rtpp_command (node=node@entry=0x7f5a522b2140, dict=dict@entry=0x7f5a644b97c8, outlen=outlen@entry=0x7ffcac5294ec) at rtpengine.c:2619 2619 rtpengine.c: No such file or directory. (gdb) thread apply all bt full Thread 1 (Thread 0x7f5a66a49700 (LWP 22992)): #0 send_rtpp_command (node=node@entry=0x7f5a522b2140, dict=dict@entry=0x7f5a644b97c8, outlen=outlen@entry=0x7ffcac5294ec) at rtpengine.c:2619 addr = {sun_family = 12616, sun_path = "A\000\000\000\000\000t\326_", '\000' <repeats 13 times>, "\240ZGMZ\177\000\000`_GMZ\177\000\000\360\224R\254\374\177\000\000\a\000\000\000\000\000\000\000\374\257DMZ\177\000\000(\230KdZ\177\000\000\214\260DMZ\177\000\000\004\000\000\000\000\000\000\000\374\257DM\001\000\000\000\360\224R\254\374\177\000\000(\230KdZ\177"} fd = <optimized out> len = 0 i = <optimized out> vcnt = 6 cp = 0x7f5a4d687dc0 <buf> "" buf = '\000' <repeats 65535 times> fds = {{fd = 223, events = 0, revents = 0}} v = 0x7f5a644b98f6 out = {s = 0x0, len = 0} __FUNCTION__ = "send_rtpp_command" #1 0x00007f5a4d44f81c in rtpp_test_ping (node=0x7f5a522b2140) at rtpengine.c:1041 bencbuf = {pieces = 0x7f5a644b97b0, free_list = 0x0, error = 0} dict = 0x7f5a644b97c8 cp = <optimized out> ret = 0 #2 0x00007f5a4d454d89 in mi_enable_rtp_proxy (cmd_tree=0x1, param=0x1) at rtpengine.c:1137 crt_node = 0x0 rtpp_list = 0x7f5a522b1f80 root = 0x16 attr = 0x0 found = 0 __FUNCTION__ = "mi_enable_rtp_proxy" #3 0x00007f5a515f071e in run_mi_cmd (cmd=<optimized out>, cmd=<optimized out>, t=<optimized out>) at ../../lib/kmi/mi.h:77 No locals. #4 mi_fifo_server (fifo_stream=0x27be8a0) at fifo_fnc.c:509 line_len = 39 file_sep = 0x0 command = 0x7f5a644a2b41 "nh_enable_rtpp" f = 0x276f838 reply_stream = 0x27c4cf0 __FUNCTION__ = "mi_fifo_server" #5 0x00007f5a515e6345 in fifo_process (rank=1) at mi_fifo.c:270 fifo_stream = 0x27be8a0 __FUNCTION__ = "fifo_process" #6 0x00007f5a515e8f1e in mi_child_init (rank=0) at mi_fifo.c:234 pid = 0 #7 0x0000000000512ba9 in init_mod_child (m=0x1, rank=1) at sr_module.c:921 __FUNCTION__ = "init_mod_child" #8 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #9 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #10 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #11 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #12 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #13 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #14 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #15 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #16 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #17 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #18 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #19 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #20 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #21 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #22 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #23 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #24 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #25 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #26 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #27 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #28 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #29 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #30 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #31 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #32 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #33 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #34 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #35 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #36 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #37 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #38 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #39 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #40 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #41 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #42 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #43 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #44 0x0000000000512ad8 in init_mod_child (m=0x1, rank=1) at sr_module.c:918 __FUNCTION__ = "init_mod_child" #45 0x0000000000510f18 in main_loop () at main.c:1696 si_desc = "udp receiver child=0 sock=127.0.0.1:5060\000\060\066\060\000\000\000\000\270\357DdZ\177\000\000\017\000\000\000\000\000\000\000n\000\000\000\000\000\000\000\255\370_\000\000\000\000\000\071\222\343p\000\000\000\000\000\071X\207\335c\334e\000\000\000\000\000\000\000\000\000\071X\207\335c\334e\210\263\246\000\000\000\000\000\005\000\000\000\000\000\000" woneinit = 1 __FUNCTION__ = "main_loop" #46 0x000000000041cefc in main (argc=3, argv=0x0) at main.c:2631 cfg_stream = 0x0 r = 0 tmp = 0x7ffcac52bee5 "" tmp_len = 0 port = 0 proto = 1722066384 ret = 0 seed = 2123298762 debug_save = 0 debug_flag = 8 dont_fork_cnt = 0 st = {st_dev = 15, st_ino = 12771, st_nlink = 2, st_mode = 16832, st_uid = 112, st_gid = 118, __pad0 = 0, st_rdev = 0, st_size = 100, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1472635951, tv_nsec = 983999907}, st_mtim = {tv_sec = 1474293427, tv_nsec = 27546593}, st_ctim = {tv_sec = 1474293427, tv_nsec = 27546593}, __glibc_reserved = {0, 0, 0}} __FUNCTION__ = "main" (gdb)

1 0

yum install modules
by mjbemail2000-seti＠yahoo.co.uk 19 Sep '16

19 Sep '16

Hi, I've installed Kamailio on CentOS using yum but it's not installed the geoip2 module, amongst others. I'm I correct in saying that to install this module I'd need to install Kamailio from the source code? Regards, Mark.

1 0

tls with ubuntu 16.04
by Jayesh Nambiar 19 Sep '16

19 Sep '16

Hi, Trying kamailio with Ubuntu 16.04 and I'm getting errors as follows: ERROR: tls [tls_init.c:490]: tls_pre_init(): Unable to set the memory allocation functions The error is identical to the one mentioned here: https://bugs.launchpad.net/ubuntu/+source/kamailio/+bug/1591992 Is there a solution to this. Thanks for any response. - Jayesh

2 21

Extract param from RPID header
by Grant Bagdasarian 19 Sep '16

19 Sep '16

Hello, I'm trying to extract the privacy param from the RPID header, but without success. Could someone point me in the right direction? Remote-Party-ID: "Cullen Jennings" <sip:+3123456789@1.1.1.1>;party=calling;privacy=off;screen=no. I've tried the following ways: 1) TEXTOPSX: $var(rpid_privacy_value) = @hf_value.remote_party_id.privacy; 2) Param Transformations: a. $(hdr(Remote-Party-ID){param.value,privacy}) b. $(re{param.value,privacy}) c. 4re{param.value,privacy} I can't seem to extract the value of privacy from the RPID header. Regards, Grant

2 2

Self challenged packets
by Torsten Hantzsche 19 Sep '16

19 Sep '16

Hi, while playing with the script from http://kb.asipto.com/kamailio:presence:k43-blf and "kamailio 4.4.3 (x86_64/freebsd) 5a2195" I could not see any NOTIFY packet going out to the subscribed watcher. "tcpdump -i lo0" shows self injected PUBLISH packets challenged with "407 Proxy Authentication Required" and there they got stuck. The PUBLISH packets are sent on lo0 from my network address to the same network address (here 10.0.0.106 to 10.0.0.106). The proxy challenge is also sent from 10.0.0.106 to 10.0.0.106. An xlog entry with the variable "$Ri" in route REQUEST shows 10.0.0.106 too. Since these packets are only seen on lo0, should "$Ri" not show 127.0.0.1? After disabling the WITH_AUTH variable, the watcher is getting its NOTIFY's. Now I wonder under which circumstances this script is supposed to function correctly without changes in the AUTH route? By the way, I had the same problem with the provided script of the msilo module (http://kamailio.org/docs/modules/4.4.x/modules/msilo.html#idp42867388) which too only works if there is no "auth_challenge" in the way. Could somebody with more insight in Kamailio internals give some hints on this, please? How can it get solved in an elegant way? Thanks, Torsten

3 3

Re: [SR-Users] help with kamailio rpm made from source
by Dmitry 18 Sep '16

18 Sep '16

Hello I tested a package http://download.opensuse.org/repositories/home:/kamailio:/v4.3.x-rpms/CentO… ( I downlowaded 4.3.6 version rpm and installed it. Kamailio restarts well. The behaviour is the same. the phone registeres without nonce. I install kamailio, kamailio-kazoo - 2 rpm зфслфпуы On Thursday, September 15, 2016 9:34 PM, Dmitry <mbike2000ru(a)yahoo.com> wrote: I made loadmodule and modparam("debugger", "cfgtrace", 1) but anyway - no logs when I register. As I understand - it's like no config file. On Thursday, September 15, 2016 6:01 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: I am not familiar with kazoo configs, maybe asking on their mailing list can help you more. From Kamailio point of view, you can load debugger module and set its cfgtrace parameter to 1, then see what actions from config are executed and why is not getting to the authentication part. Cheers, Daniel On 15/09/16 14:09, Dmitry wrote: /etc/kazoo/kamailio/default.cfg - which containes all routes. 2600hz/kazoo-configs | | | | | | | | | | | 2600hz/kazoo-configs kazoo-configs - Kazoo Configuration Files for Software We Use | | | | I test on a working server (testing one) and a working config -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda On Thursday, September 15, 2016 3:56 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Are you using default kamailio.cfg or another one? Cheers, Daniel On 15/09/16 12:39, Dmitry wrote: Hello I took this spec from suse. It generates no errors. When I installed from the RPM I had made - the phone register, but The phone sends a REGISTER and the KAmailio sends 200ok back to the phone (so no NONCE authorization) and no logs during it. In default.cfg I set L_DBG but no logs are generated during registering a phone. And I see no AMQP messages (I set to 1 Kazoo_enable in the SPEC file). Why there are no logs during register (I saw register in tcpdump)? On Wednesday, September 14, 2016 7:42 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Then you just need add those files in various packages inside the spec file, so they are not detected to be orphaned. Maybe you can inspire from: - https://build.opensuse.org/package/view_file/home:kamailio:v4.3.x-rpms/kama… Cheers, Daniel On 14/09/16 16:35, Dmitry wrote: 4.3.4 version is for Kazoo It is on production server currently. I need to rebuild the current RPM so as to apply patches. But first I want to get a working Kamailio and only after it I will apply the patches. I think I may take a list of modules from the production Kazoo-kamailio and rearchive the tar.gz because now the Fedora 18 rpmbuild gives a lot of files: RPM build errors: Installed (but unpackaged) file(s) found: /usr/lib64/kamailio/modules/auth_xkeys.so /usr/lib64/kamailio/modules/dmq_usrloc.so /usr/lib64/kamailio/modules/ims_charging.so /usr/lib64/kamailio/modules/jsonrpc-s.so /usr/lib64/kamailio/modules/mohqueue.so /usr/lib64/kamailio/modules/nosip.so /usr/lib64/kamailio/modules/rtjson.so /usr/lib64/kamailio/modules/rtpengine.so /usr/lib64/kamailio/modules/statsd.so /usr/lib64/kamailio/modules/tcpops.so /usr/lib64/kamailio/modules/tsilo.so /usr/share/doc/kamailio/modules/README.auth_xkeys /usr/share/doc/kamailio/modules/README.dmq_usrloc /usr/share/doc/kamailio/modules/README.ims_charging /usr/share/doc/kamailio/modules/README.ims_registrar_scscf /usr/share/doc/kamailio/modules/README.jsonrpc-s /usr/share/doc/kamailio/modules/README.mohqueue /usr/share/doc/kamailio/modules/README.nosip /usr/share/doc/kamailio/modules/README.rtjson /usr/share/doc/kamailio/modules/README.rtpengine /usr/share/doc/kamailio/modules/README.statsd /usr/share/doc/kamailio/modules/README.tcpops /usr/share/doc/kamailio/modules/README.tsilo /usr/share/snmp/mibs/KAMAILIO-MIB /usr/share/snmp/mibs/KAMAILIO-REG-MIB /usr/share/snmp/mibs/KAMAILIO-SIP-COMMON-MIB /usr/share/snmp/mibs/KAMAILIO-SIP-SERVER-MIB /usr/share/snmp/mibs/KAMAILIO-TC On Wednesday, September 14, 2016 5:05 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Hello, any reason not to use series 4.4.x? Iirc, the latest spec that got update on 4.4 are those for oracle enterprise linux, perhaps is something that you can reuse a lot for upgrading to the centos flavour. On the other hand, you can use opensuse build service if you want to build yourself, the spec from the kamailio project there is up to date and the ones for 4.3 should still be there. Anyhow, as I said, I would recommend to go with 4.4.x version. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

avp_db_query
by Angel Elena 18 Sep '16

18 Sep '16

Hi, good moorning. I have a Kamailio 4.0 with avp module working. Now, I need write some values in my database and I don't find and example for this. if (avp_db_query("update subscriber set bloqueado=0 where name=xxxxx")) { } Is possible make this operation with avp_db_query ? Thanks -------------------------------- Ángel Elena Medina _o) craem(a)craem.net / \\ http://blog.craem.net _(___V @craem_ --------------------------------

2 2

Replacing Asterisk with Kamailio
by Valter Nogueira 17 Sep '16

17 Sep '16

I use Asterisk for SIP and Media Proxy. Despite the fact that Asterisk is not a SIP Proxy at all. Customer registers in a SIP account, sends the invite and thru de context Asterisk dials out thru a SIP Trunk. Asterisk does the media proxy, since customer can't route directly to the SIP Trunk (altough it has a valida address, it don't have a public route allowed to it). I need limit customer concurrent calls, mangle some dial-in/dial-out numbers, keep track of ongoing call, control SIP dialog, retransmit correct hang-up causes and do media proxy (no transconding at all) After reading about Kamailio and Opensips, and due to the Kamailio Admin Book, I decided to go with Kamailio. Well, I understand that I have to use some kamailio modules, like auth, dialplan, rtpproxy and db_mysql. What make me stuck is how does everything fit together in kamailio.cfg and how do I get ongoing calls and CDR's? Can anyone point me a direction? Thanks

6 12

Commercial SBC or Kamailio
by Linux Vince 16 Sep '16

16 Sep '16

We are trying to setup VOIP infrastructure, mainly wholesale and retail. What is best option if money is not the problem? Developing our own infrastructure using Kamailio and other open source packages to act as SBC and switch or buy a commercial solution like GenBand/Sonus/Sansay. We are looking for high performance with scalability to handle thousands of call setups per second over the period of few years. I have no idea on how/if can kamailio outperform commercial solutions or not. Advantage of using kamailio is flexibility and possibility of customization as per requirement, I am new to this list so please advise if this is not a valid question to be asked here.

7 7

Kamailio fails to start when MySQL down
by Daniel Pocock 16 Sep '16

16 Sep '16

I recently set up a Kamailio instance using the default configuration for HOMER with MySQL[1] The database was not running and Kamailio refused to start. Would it be better for Kamailio to start anyway and go into a loop trying to connect to the database, just as if auto_reconnect was set? This is useful for people who start Kamailio from their init script and can't be sure if their database host is up before their Kamailio host boots 1. https://github.com/sipcapture/homer-api/blob/master/examples/sipcapture/sip…

3 2

Integration with Broadsoft and Config Question
by Daryn Johnson 16 Sep '16

16 Sep '16

I’m looking to briefly check the sanity of my configuration. I have been working w/Kamailio the last few months, and have read the SIP router ebook. Thank you for your efforts and the documentation. It has been a GREAT HELP! We are attempting to replace an Acme Packet SBC, and our setup is as follows: |ACCESSUSERS| ⇔ |KAMAILIO/SEMS| ⇔ |BROADSOFT| ⇔ |GENBAND| ⇔ |PSTN| Kamailio & SEMS are (currently) on the same box, and communicate using localhost via different ports. They also listen on the same private (core) IPs, using different ports. Since Broadsoft doesn’t support the PATH header, I manipulate the contact header w/the SEMS privateIP & port in the contact header. This causes it to be stored in the Broadsoft Registration string, and calls from PSTN route directly to SEMs to start calls etc… Registraton code looks like this: route[REGISTRATION] { remove_hf("Route"); add_path_received(); $avp(contact) = $ct; xlog("L_INFO", "found contact header $avp(contact)"); setflag(FLAG_REGISTRATION); if (nat_uac_test("20")){ xlog("L_INFO", "[REGISTRATION_ATTEMPT] [FROM_NAT] R=$ru ID=$ci UA='$ua'\n"); force_rport(); remove_hf("Contact"); append_hf("Contact: <sip:$fU" + "@" + "!SEMSPRIVATEIP!:5060;" + "alias=$si~$sp~1>\n"); setflag(FLT_NATS); } else { xlog("L_INFO", "[REGISTRATION_ATTEMPT] [PUBLIC_IP] \r\n[REGISTRATION_INFO] R=$ru UA='$ua'\r\n"); remove_hf("Contact"); append_hf("Contact: <sip:$fU@" + "!SEMSPRIVATEIP!:5060;" + "alias=$si~$sp~1>\n"); } $du = "sip:R_IP:REGISTRAR_PORT"; t_on_reply("REG"); route(RELAY); exit; } Then in the relay route, I force the registration out of the private interface & port of Kamailio if (isflagset(FLAG_REGISTRATION)){ $fs = "LISTEN_UDP_PRIVATE"; } Finally, if a registration is successful from BROADSOFT, I store it in the location table, to allow me to screen calls from access to ensure they are ‘registered’ onreply_route[REG]{ if(status=="200") { xlog("L_INFO", "REGISTRATION BE GOOD FOR $fu"); save("location", "0x04"); } } Calls from ACCESS, I check USRLOC for screening, then pass the call to SEMS via the localhost:port. From there, sems enacts the B2BUA/SBC module and routes the call to BROADSOFT, and along to our GENBAND Gateway… I could never get calls to work properly without SEMS, just using RTP PROXY/ENGINE, possibly because the RTP for the GENBAND & KAMAILIO/SEMS are on different private subnets. Our setup is fairly legacy ‘heavy’ and many things can’t be changed. Calls from PSTN route to GENBAND, then BROADSOFT, then SEMS, then KAMAILIO. Since I have manipulated the contact header, BROADSOFT maintains that in the INVITE and I can use the ‘handle_ruri_alias’ to route the call properly. SEMS sets the PUBLIC IP of the KAMAILIO/SEMS box in the SDP, and facilitates rtp relaying. if (isflagset(FLAG_TOUSER)){ xlogl("L_INFO", "TOUSERFLAGSET"); if(is_method("INVITE|BYE")){ handle_ruri_alias(); } /*if(has_body("application/sdp") && nat_uac_test("8")) { fix_nated_sdp("10"); xlog("L_INFO", "SDP CONTAINS NAT"); }*/ $fs = "LISTEN_UDP_PUBLIC"; } I’ve gone headlong into this setup, and it works! But, I wanted to get a ‘Sanity Check’ to determine if there is a better way to do this. We are at the point where I’m doing load testing and attempting to validate this for a ‘production’ ready state. If possible, I’d just like to ask the community if they have encountered a similar setup, and how they solved it? Thanks in advance!!! Daryn Johnson Senior VoIP Engineer 248.485.1109 www.telnetww.com 1175 W. Long Lake Rd. | Suite 101 | Troy, MI 48098

1 0

Kamailio Authentication failing
by Eric Koome 16 Sep '16

16 Sep '16

Hi all - my Kamailio - 4.1.6 is receiving this particular structured INVITES from multiple IPs, and for some reason it is not requesting authentication. I have AUTH & IPAUTH modules in use for two years now, but this is bypassing that and actually forwarding the invite to asterisk servers behind Kamailio. I notice from the invite that the contact (c=IN IP4 10.10.10.10) is unusual and in the private range. Is this what is bypassing Authentication? Any Pointers on how to stop this. This is flooding my boxes! ========================================== 2016/09/16 06:45:16.824768 83.48.28.27:63029 -> MYPUBLICIP:5060 INVITE sip:9441134018000@MYPUBLICIPSIP/2.0 Via: SIP/2.0/UDP 83.48.28.27:5060;branch=z9hG4bK898408493 Max-Forwards: 70 From: "442030930112" <sip:442030930112@83.48.28.27>;tag=5044916148 To: <sip:9441134018000@MYPUBLICIP> Contact: <sip:442030930112@83.48.28.27> Call-ID: 9xCjfQkrmDwT7D6AqM0BTYAgmWjG84nq CSeq: 102 INVITE User-Agent: SIP Call Date: Fri, 16 Sep 2016 05:45:48 +0000 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 308 v=0 o=442030930112 8982 1536 IN IP4 10.10.10.10 s=SIP c=IN IP4 10.10.10.10 t=0 0 m=audio 14806 RTP/AVP 0 3 8 4 18 101 a=rtpmap:0 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv Regards Eric

2 1

http_async_query - How to Allow SIP transaction to continue
by Jonathan Hunter 16 Sep '16

16 Sep '16

Hi Guys, I am just testing with jansson and http_async_query. Essentially I sent an INVITE into kamailio ,it hits CALL_START_REQUEST (see below) and the query allows the http site I send to, to provide routing details for kamailio. Logging the $rm that hits CALL_START_ REQUEST, it shows as INVITE, however when CALL_START_REQUEST_REPLY is hit, the message type is OPTIONS, is this expected behaviour? Essentially I want it to continue in the script with the same $ci,$rm so I can onwards route it for termination, but all I get are options messages being sent, any hints please? As Ideally I want the transaction to continue to a route where I can forward on modified request for termination. Thanks Jon Routes; # Send to FreeSWITCHroute[CALL_START_REQUEST]{xlog("L_INFO","route[CALL_START_REQUEST] Call received $ci,$rm,$ru,$du\n"); ###Build JSON request to send to MAP$var(params) = $null;jansson_set("string", "username", "xxx", "$var(params)");jansson_set("string", "password", "xxx", "$var(params)");jansson_set("string", "source", "$fU", "$var(params)");jansson_set("string", "route", "$var(carriername)", "$var(params)");jansson_set("string", "destination", "$rU", "$var(params)");/// other fields set herehttp_set_timeout(5000);http_set_method("POST");http_append_header("Content-Type: application/json");http_async_query("http://test.com", "$var(params)", "CALL_START_REQUEST_REPLY");xlog("L_INFO", "Checking here we have status=$var(status)\n");xlog("L_INFO", "Current Destination is $du\n");#return;#exit;}#########################route[CALL_START_REQUEST_REPLY]{xlog("L_INFO", "route[CALL_START_REQUEST_REPLY]: $ci,$rm,$ru,$du\n"); Logs show; route[CALL_START_REQUEST] Call received b087fa53-f5fc-1234-94ac-f68eb8da94f7,INVITE,sip:441624891676@8.8.8.8:5060;user=phone,<null> route[CALL_START_REQUEST_REPLY]: 123,OPTIONS,sip:G711A-282951200@kamailio.org,sip:G711A-282951200@8.8.8.8:5080

2 3

help with kamailio rpm made from source
by Dmitry 16 Sep '16

16 Sep '16

Hello I obtained the source from https://www.kamailio.org/pub/kamailio/4.3.4/src/kamailio-4.3.4_src.tar.gz I found a SPEC file there and I prepared a RPM with this spec file. When I start a Kamailio (from this rpm) - it starts but the LOG file gives only 1 line: INFO: <core> [tcp_main.c:4657]: init_tcp(): using epoll_lt as the io watch method (auto detected) I want to test this rpm and if successfull i willl add patches and rebuild the rpm. How to get correct RPM?

3 17

Branch variables in branch_failure_route
by Paul Smith 14 Sep '16

14 Sep '16

Hi, I am struggling to figure out variable scope in branches and branch_failure_routes. Is there a way to store a variable in a branch route so that I can then read it from the branch failure route if that branch fails? Also does writing $ru in branch_failure_route append a branch? For example I have multiple SIP devices registered to Kamailio on the same account credentials. The devices are behind same NAT, so same public IP. Some devices have SRTP required and some cannot use SRTP at all, Kamailio does not know in advance of the call whether the call will use SRTP or RTP. When I send a call to that user Kamailio correctly parallel forks to all the registered devices. I use rtpengine offer to send SDP with RTP/SAVP in the initial invites. If one of the devices sends a 488 due to not being able to handle the RTP/SAVP I then want to run rtpengine_offer with RTP/AVP and send another invite to the same device. In the branch failure route I seem to have to read the current $ru and then write it back into $ru before calling t_relay() or I get Sep 14 14:56:06 registrar-secure /usr/sbin/kamailio[10168]: ERROR: tm [t_fwd.c:1771]: t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding This code seems to work, but it feels like I’m doing something very wrong…. in particular $avp(triedrtpout) is set for the transaction, I should be setting it separately for each branch. event_route[tm:branch-failure:SRTP] { # Handle failure response xlog("L_INFO", "Handling $T_reply_code response to $rm to ru: <$ru> and du: $du\n"); if(t_check_status("488") && is_method("INVITE")) { if ($avp(triedsrtpout)==1) { # set $ru seems to create a new branch ... even when we set it to its current value. # we want to generate a new SIP INVITE to the same AOR but with different SDP $avp(buffer)=$ru; $ru=$avp(buffer); xlog("L_INFO","488 caught. Resending to $ru"); # try with unencrypted RTP/AVP rtpengine_delete(); rtpengine_offer("to-tag trust-address replace-origin replace-session-connection ICE=force RTP/AVP"); if (!t_relay()) { sl_reply_error(); } } } } I am using kamailio 4.2 debian package for this registrar. Sorry for the rambling question … I don’t understand enough to write the short version! Paul

3 3

Kamailio v4.4.3 Released
by Daniel-Constantin Mierla 14 Sep '16

14 Sep '16

Hello, Kamailio SIP Server v4.4.3 stable release is out. This is a maintenance release of the latest stable branch, 4.4, that includes fixes since the release of v4.4.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.4.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.4.3. For more details about version 4.4.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2016/09/kamailio-v4-4-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

jansson_get path to access a JSON reponse containing an array
by Jonathan Hunter 14 Sep '16

14 Sep '16

Hi guys, Sorry for the question but Its not clear to me using json_path and the jansson module how to get the contents of a response containing an array. Please see the returned response below that I am getting on kamailio; {"status":"success","destination"[{"msisdn":"4412345566","route":"Carrier1"}],"maxduration":"60","source":"447777777","message_id":"c41c1823aafb80f44620f4cdc6802c0d"} I can access any of the fields status,destination,maxduration,source,message_id fine, however you will notice with destination it has an array (as multiple destinations with other queries could return more than one choice). So currently when I do jansson_get_field("$http_rb", "destination", "$var(mapdestination)"); I get; [{"msisdn":"447107110194","route":"carrier1"}] How do I get into the contents of the array in the first place so I can return the msisdn value itself, and the route? This is not a problem if its not an array, however if the result is contained within [] how do I indicate the path? I have tried destination[0], destination.msisdn with no luck, and Im sure I am missing something so any advice would be great. Thanks Jon

2 2

Planning the release of Kamailio v4.4.3
by Daniel-Constantin Mierla 14 Sep '16

14 Sep '16

Hello, now that the summer in the northern hemisphere is gone and many of us returned at the work, I am considering to release a new version of Kamailio out of the latest stable branch 4.4. The plan is to have it out next week, either on Wednesday Sep 14 or Thursday, the following day. If the are issues not reported on the bug tracker from github project, it is better to open one item. Do it also if you reported something on sr-users or sr-dev in the past two months and it is not actual discussion about it, likely it was forgotten or not noticed during the holidays and digesting the archive is going to take a while. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

Kamailio as Proxy with public and private interface
by gmele 13 Sep '16

13 Sep '16

Hello, we are building a new SIP service composed by a 2 SIP Proxies and 2 SIP Registrars. There is only one proxy active (controlled by keepalived). To secure our SIP infra, we have decided to publish a TLS public interface on the proxy side. The SIP registrar will not be reachable from the outside, so we have added a provate interface on the SIP proxy and another one on the SIP registrar (192.168.1.x). In brief: SIP Proxy, 2 I/F, 213.xxx.xxx.xxx (TLS public) and 192.168.1.x (UDP private) SIP Registrar, 1 I/F, 192.168.1.x (UDP private). Is this architecture possible with kamailio? We tried to implement this, but it seems that all requests incoming from the public IP address are correctly routed to the SIP registrar, but with the public IP instead of the private one, resulting in the impossibility to send the SIP requests to the registrar. Should we add a VIA toute in the SIP message before forwarding it to the SIP registrar? How can this be done? Thx for your help Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-as-Proxy-with-public-and-p… Sent from the Users mailing list archive at Nabble.com.

1 1

Issues on getting Notify Body data and parse it
by José Seabra 13 Sep '16

13 Sep '16

Hello there, I'm facing some issues trying to get the attributes from notify Body that contains information that i need for my calls . First i save the $rb into $var then i try to replace the characters #012 that seems to be the separator of all attributes in the body, i have tried it using s.select, s.strip, s.replace but without any success in all of them. The body msg is something like this: Calling-Name-Status: available#012Calling-Name: "teste XX" < sip:XXXXXXXXX@domain.com>#012Presentation-Indicator: allowed Anyone have any idea how can I get the value of each attribute: e.g: Calling-Name-Status: available Calling-Name: "teste XX" <sip:XXXXXXXXX@domain.com> Presentation-Indicator: allowed Best Regards -- José

2 3

Dump running config out of Kamailio into a file
by Jesus Tovar 13 Sep '16

13 Sep '16

Hi, I currently have a situation with my Kamailio servers sitting in production, the folder containing all configuration files at /etc/kamailio was deleted from the server and there is not backup. Is there a possibility to show or dump the current running configuration into a file so I could restore the configuration file from it. If it happens these servers get restarted then we are going to be in real trouble. Thanks for your help Kind regards, Albert T. Jesus Tovar | Senior Network Engineer | The Summit Group jtovar(a)thesummitgroup.com.au thesummitgroup.com.au 1300 049 749 Level 1, 39 Railway Road, Blackburn VIC 3130 The Summit Group

1 0

Radius Authentication for kamailio-4.4.2
by Chan Kwang Mien 12 Sep '16

12 Sep '16

Hi, I have installed Kamailio-4.4.2 and radiusclient-ng-0.5.6 for Radius Authentication to a Free Radius Server. I have used the steps in http://www.kamailio.org/docs/openser-radius-1.0.x.html to enable Radius Authentication for Kamailio-4.4.2. However, when I restart kamailio and my SIP phone register to Kamailio, Kamailio did not Radius Authenticate to Free Radius Server. Can anyone advise me on other document for Radius Authentication for Kamailio ? Thanks. Regards, Kwang Mien

2 1

pua_dialoginfo: Tags only in "early" state PUBLISH
by Stefan Kohlhauser 12 Sep '16

12 Sep '16

Hello everyone! I've been playing around with the pua_dialoginfo. I noticed that the From- and To-Tags are only included in the PUBLISH's dialog-info of the "early" state. This is also true if include_tags of pua_dialoginfo is explicitly set to '1'. The reason for this behaviour is probably that in pua_dialoginfo.c in the function __dialog_sendpublish() the call to dialog_publish_multi() only happens with the tag parameters set for the "early" state. For all other dialog states the tag parameters are statically set to '0'. Even though the local- and from-tags are optional I tend to regard this as a bug. I would have expected for the tags to be included in the dialog-info for all states when include_tags is set. Or is there a reason for this behaviour? Best regards Stefan

2 1

Kamailio RFC/3GPP standards
by Owais Ahmad 12 Sep '16

12 Sep '16

Hello, I can only see the 3GPP specification mentioned in ims_qos module. Please have the other IMS modules also updated with the respective 3gpp versions. Thanks, Owais

2 1

Re: [SR-Users] Kamailio - Multiple NAT
by Gholamreza Sabery 12 Sep '16

12 Sep '16

The best way to see what is happening is to capture some packets on the Kamailio system on which calls happen and see if any IP address or port number issues exist in body of SIP messages or not. You can simple use Wireshark to see what is going on. Regards On Mon, Sep 12, 2016 at 4:58 PM, Marino Mileti <marino.mileti(a)alice.it> wrote: > Hi guys, > > > > i'm going in trouble with multinat scenario and Kamailio J > > > > I've this scenario... > > > > Two kamailio boxes with the same configuration (KAM_A and KAM_B). Each box > has a WIFI in AP_MODE with the same address (but different SSID obviously) > 192.168.254.1. Finally there are 2 clients...CLIENT_1 connected to wifi > with KAM_A, and CLIENT_2 connected using wifi with KAM_B. Each box has an > instance of RTPEngine configured. > > KAM_A & KAM_B are also connected using wired connection with other clients > in order to allow that a calls to KAM_B (or KAM_A) generate a forking > scenario and all clients connected starts to RINGS (wired & wireless client) > > > > The mobile client connected on KAM_A makes a call to KAM_B... the forking > scenario starts and all clients on KAM_B starts to ring. > > If any wired client pick up the call everything works fine. If a wireless > client pick up the call...no media L > > > > Is it possible? Should I've to instruct RTPENGINE in any particular mode? > > > > > > *Marino Maria Mileti* > > *marino.mileti(a)alice.it <marino.mileti(a)alice.it>* > > > > *[image: cid:006a01cb6b0e$67eecdae$_CDOSYS2.0]**Reduce your energy > consumption and keep polar bears on ice!* > > > > > ------------------------------ > [image: Avast logo] > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > _______________________________________________ > 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 > >

1 0

Kamailio - Multiple NAT
by Marino Mileti 12 Sep '16

12 Sep '16

Hi guys, i'm going in trouble with multinat scenario and Kamailio J I've this scenario... Two kamailio boxes with the same configuration (KAM_A and KAM_B). Each box has a WIFI in AP_MODE with the same address (but different SSID obviously) 192.168.254.1. Finally there are 2 clients...CLIENT_1 connected to wifi with KAM_A, and CLIENT_2 connected using wifi with KAM_B. Each box has an instance of RTPEngine configured. KAM_A & KAM_B are also connected using wired connection with other clients in order to allow that a calls to KAM_B (or KAM_A) generate a forking scenario and all clients connected starts to RINGS (wired & wireless client) The mobile client connected on KAM_A makes a call to KAM_B... the forking scenario starts and all clients on KAM_B starts to ring. If any wired client pick up the call everything works fine. If a wireless client pick up the call...no media L Is it possible? Should I've to instruct RTPENGINE in any particular mode? Marino Maria Mileti <mailto:marino.mileti@alice.it> marino.mileti(a)alice.it cid:006a01cb6b0e$67eecdae$_CDOSYS2.0Reduce your energy consumption and keep polar bears on ice! --- Questa e-mail è stata controllata per individuare virus con Avast antivirus. https://www.avast.com/antivirus

1 0

Advice for routing and sbc functions
by Oliver Roth 12 Sep '16

12 Sep '16

Hi all I am using kamailio for several years now and I am really happy with it! Now an new scenario to solve: Sip-carrier A --> Kamailio loadbalancer --> multiple kamailio routing gateways --> freeswitch/sbc/media-gw/whatever --> sip - carriers B,C & D Kamailio gateways: They are doing different checks, manipulate the numbers (national /international format) and they use the carrierroute module freeswitch/sbc/media-gw/whatever: how can I use these services with the routing information from the carrierroute module? These media-gws will do the connection and maybe transcoding to the carriers. Can you please help me with the interaction within the systems "routing gateways" and the "media/sbc gateways"? Do I have to add all the carriers in the carrierroute as "routes" with the ip of the media gw/freeswitch and then splitting it up on the freeswitch again? Thanks for helping Oli

1 0

DSCP options
by Marcelo Bonin 09 Sep '16

09 Sep '16

I need to set DSCP values in Kamailio, I couldn't found a way to set in the configuration. Any help will be appreciated. Thanks.

2 1

4.4 topos ignoring Record-Route?
by Daniel Tryba 09 Sep '16

09 Sep '16

Looking into the topos module of 4.4.2 I see something unexpected, ACK and BYE are routed directly to the Contact in the 200 OK on an INVITE where the 200 OK has a Record-Route. Without the topos module the message flow is as expected: 109.235.32.57<->185.61.68.106<->109.235.32.55 With topos the ACK and BYE take a different route: 109.235.32.57<->185.61.68.106<->109.235.32.55 <->109.235.32.48 109.235.32.5x are proxies. 185.61.68.106 is the registrar To my understanding, in the callflow below the ACK should be send via the proxy that inserted Record-Route: <sip:109.235.32.55;lr> in the 200 OK. Am I wrong to conclude this or the topos module ignoring the routing? Also note the To/From are passed untouched in the ACK/BYE and the BYE to the called endpoint hasn't had its Via headers stripped. Call scenario: tryba/+31407110xxx calls 0880100XXX. Both are registered to 185.61.68.106 via proxies (resp. 109.235.32.57 and 109.235.32.55) >From 109.235.32.57 -> 185.61.68.106: INVITE sip:0880100XXX@sip.itco.nl SIP/2.0 Record-Route: <sip:109.235.32.57;r2=on;lr> Record-Route: <sip:109.235.32.57;transport=tcp;r2=on;lr> Via: SIP/2.0/UDP 109.235.32.57;branch=z9hG4bK7dd7.e15a4ffb2c3a7f882117b80c85f75223.0;i=60c Via: SIP/2.0/TCP 10.0.3.169:5067;rport=5067;received=109.235.34.226;branch=z9hG4bKb5004b60de287221 Route: <sip:109.235.32.57;transport=tcp;lr> From: <sip:tryba@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:0880100XXX@sip.itco.nl> Contact: <sip:tryba@10.0.3.169:5067;transport=tcp;alias=109.235.34.226~5067~2> >From 185.61.68.106 -> 109.235.32.55: INVITE sip:+31880100XXX@109.235.32.48 SIP/2.0 Via: SIP/2.0/UDP 185.61.68.106;branch=z9hG4bK7dd7.d4cdfe8014e0d5d806b4dc4a7422c8e0.1 Route: <sip:loadbalancer@109.235.32.55;lr;received=sip:109.235.32.48:5060> From: <sip:+31407110XXX@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:+31880100XXX@sip.itco.nl> Contact: <sip:btpsh-57b6cc75-f42d-1@185.61.68.106> >From 109.235.32.55 -> 185.61.68.106: SIP/2.0 200 OK Via: SIP/2.0/UDP 185.61.68.106;rport=5060;branch=z9hG4bK7dd7.d4cdfe8014e0d5d806b4dc4a7422c8e0.1 Record-Route: <sip:109.235.32.55;lr> From: <sip:+31407110XXX@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:+31880100XXX@sip.itco.nl>;tag=as61fc6a47 Contact: <sip:%2b31880100XXX@109.235.32.48> >From 185.61.68.106 -> 109.235.32.57 SIP/2.0 200 OK From: <sip:tryba@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:0880100XXX@sip.itco.nl>;tag=as61fc6a47 Via: SIP/2.0/UDP 109.235.32.57;branch=z9hG4bK7dd7.e15a4ffb2c3a7f882117b80c85f75223.0;i=60c,SIP/2.0/TCP 10.0.3.169:5067;rport=5067;received=109.235.34.226;branch=z9hG4bKb5004b60de287221 Contact: <sip:atpsh-57b6cc75-f42d-2@185.61.68.106> Record-Route: <sip:109.235.32.57;r2=on;lr>,<sip:109.235.32.57;transport=tcp;r2=on;lr> >From 109.235.32.57 -> 185.61.68.106: ACK sip:atpsh-57b6cc75-f42d-2@185.61.68.106 SIP/2.0 Via: SIP/2.0/UDP 109.235.32.57;branch=z9hG4bK7dd7.a49ec0c055166a1935ff84cfca4d1a5c.0;i=60c Via: SIP/2.0/TCP 10.0.3.169:5067;rport=5067;received=109.235.34.226;branch=z9hG4bK24aff2df2ee7b43f From: <sip:tryba@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:0880100XXX@sip.itco.nl>;tag=as61fc6a47 Contact: <sip:tryba@10.0.3.169:5067;transport=tcp;alias=109.235.34.226~5067~2> >From 185.61.68.106 -> 109.235.32.48 ACK sip:%2b31880100XXX@109.235.32.48 SIP/2.0 Via: SIP/2.0/UDP 185.61.68.106;branch=z9hG4bK7dd7.ac1b9a45042f9c26277afb9ec32bebe0.0 From: <sip:tryba@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:0880100XXX@sip.itco.nl>;tag=as61fc6a47 Contact: <sip:btpsh-57b6cc75-f42d-1@185.61.68.106> >From 109.235.32.57 -> 185.61.68.106 BYE sip:atpsh-57b6cc75-f42d-2@185.61.68.106 SIP/2.0 Via: SIP/2.0/UDP 109.235.32.57;branch=z9hG4bK4dd7.24da86af0a970ee399c9504933d8d751.0;i=60c Via: SIP/2.0/TCP 10.0.3.169:5067;rport=5067;received=109.235.34.226;branch=z9hG4bK0d8db1380c5011b6 From: <sip:tryba@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:0880100XXX@sip.itco.nl>;tag=as61fc6a47 >From 185.61.68.106 -> 109.235.32.48 BYE sip:%2b31880100XXX@109.235.32.48 SIP/2.0 Via: SIP/2.0/UDP 185.61.68.106;branch=z9hG4bK4dd7.4916bf7bcaa3b0346e43c3785a9be782.0 Via: SIP/2.0/UDP 109.235.32.57;branch=z9hG4bK4dd7.24da86af0a970ee399c9504933d8d751.0;i=60c Via: SIP/2.0/TCP 10.0.3.169:5067;rport=5067;received=109.235.34.226;branch=z9hG4bK0d8db1380c5011b6 From: <sip:tryba@sip.itco.nl>;tag=576aff337d33fbe5 To: <sip:0880100XXX@sip.itco.nl>;tag=as61fc6a47

2 19

Video calls with delay on one side
by Yuriy Gorlichenko 09 Sep '16

09 Sep '16

Hi. I usein kamailio 4.4 +rtpengine 4.5 for making videocalls though Web And have issue with it: I have one way audio and video till video not started. For example i calling form A point to B point B point accept call and have Audio and Video flow from the point A but point A have no any media flow. For now i using scheme when kamailio runs together with asterisk but also i had this issue with calls without asterisk. There is a link on issue i discribed before https://github.com/sipwise/rtpengine/issues/260 I use settings at the kamailio rtpengine_manage("force trust-address replace-origin replace-session-connection RTP/SAVPF") for making UDP call to WS And rtpengine_manage("trust-address replace-origin replace-session-connection ICE=remove DTLS=passive RTP/AVP media-address=MY_IP_ADDR"); For converting to back I use folloving scheme (Call from A to B) WSphone(A)->(1-st leg)-> kamailio+rtpeinge->(2-nd leg->)asterisk asterisk ->(3-st leg)-> kamailio+rtpeinge->(4-th leg)->WSphone(B) At the example below rtpengine (with kamailio) an asteirsk uses same server But connection made thoufh external interface. not local (it is made because in a production scheme asteirsk will be in a different server, but actuaaly it is does not have any matter, just for understanding topology) So at the logs after call i see next things First call from point A to asterisk (first leg and second leg) [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: ------ Media #1 <https://github.com/sipwise/rtpengine/issues/1> (audio over RTP/SAVPF) using opus/48000/2 [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30170 <> A.po.int.ip:51918, 244 p, 22944 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30171 <> A.po.int.ip:51920 (RTCP), 4 p, 196 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: ------ Media #2 <https://github.com/sipwise/rtpengine/issues/2> (video over RTP/SAVPF) using VP8/90000 [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30200 <> A.po.int.ip:51922, 209 p, 164499 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30201 <> A.po.int.ip:51924 (RTCP), 129 p, 4292 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --- Tag 'as76205e6a', created 0:39 ago for branch '', in dialogue with '5aalmrbaek' [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: ------ Media #1 <https://github.com/sipwise/rtpengine/issues/1> (audio over RTP/AVP) using opus/48000/2 [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30156 <> my.ser.ver.ip:29236, 242 p, 23861 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30157 <> my.ser.ver.ip:29237 (RTCP), 0 p, 0 b, 0 e, 1473403340 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: ------ Media #2 <https://github.com/sipwise/rtpengine/issues/2> (video over RTP/AVP) using VP8/90000 [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30184 <> my.ser.ver.ip:27252, 148 p, 152972 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [ga5c8qb2uaasr42j63h4]: --------- Port my.ser.ver.ip:30185 <> my.ser.ver.ip:27253 (RTCP), 0 p, 0 b, 0 e, 1473403340 last_packet call from asterisk to point B (3-d leg and 4-th leg) [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: ------ Media #1 <https://github.com/sipwise/rtpengine/issues/1>(audio over RTP/AVP) using opus/48000/2 [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30238 <> my.ser.ver.ip:29958, 239 p, 24902 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30239 <> my.ser.ver.ip:29959 (RTCP), 0 p, 0 b, 0 e, 1473403340 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: ------ Media #2 <https://github.com/sipwise/rtpengine/issues/2>(video over RTP/AVP) using VP8/90000 [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30264 <> my.ser.ver.ip:40374, 205 p, 163663 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30265 <> my.ser.ver.ip:40375 (RTCP), 0 p, 0 b, 0 e, 1473403340 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --- Tag 'fgl9f33gf0', created 0:39 ago for branch '', in dialogue with 'as5f26dbd9' [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: ------ Media #1 <https://github.com/sipwise/rtpengine/issues/1>(audio over RTP/SAVPF) using opus/48000/2 [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30218 <> B.po.int.ip:26138, 242 p, 21481 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30219 <> B.po.int.ip:30161 (RTCP), 4 p, 188 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: ------ Media #2 <https://github.com/sipwise/rtpengine/issues/2>(video over RTP/SAVPF) using VP8/90000 [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30246 <> B.po.int.ip:31174, 198 p, 163302 b, 0 e, 1473403349 last_packet [1473403379.000108] INFO: [7742bfcf0e9570ab6d7d40a52cd4cd4a@my.ser.ver.ip:5999]: --------- Port my.ser.ver.ip:30247 <> B.po.int.ip:59057 (RTCP), 12 p, 540 b, 0 e, 1473403349 last_packet As i see there is no any lost pakets and errors and all packets going through but at the asteirsk log i see that Sent RTP P2P packet to my.ser.ver.ip:30238 (type 107, len 000102) Sent RTP P2P packet to my.ser.ver.ip:30156 (type 111, len 000071) Sent RTP P2P packet to my.ser.ver.ip:30238 (type 107, len 000079) when packets sended from asterisk to rtpengine they have no response So it looks like that asterisk have no response from rtpengine when sending him any IP. Also i checked at the webphone side but it sending pakents. So at this side no trouble... Thanks for advices!

1 0

app_lua with lua5.3
by Péter Barabás 08 Sep '16

08 Sep '16

Hi, I would use lua5.3 with app_lua module. Is it possible since it is not compiled with lua5.1? Thanks. Peter

2 2

Call forking with delay
by Igor Potjevlesch 08 Sep '16

08 Sep '16

Hello list, I have a proxy that manage fix calls and a proxy that manage mobile calls. I want to implement call forking so when I call to a fix phone the call goes to the fix proxy and it forks to the mobile proxy who manage the call and viceversa. I want the fix to ring 2 or 3 times before the mobile. In order to do so in my fix proxy I added a diversion field with the reason "forking" and in my mobile proxy I check if the reason is "forking" to wait with the fonction "async_route" from the ASYNC module. When I call the fix I can fork to the mobile with delay without any problem because in that case my fix proxy manage one call and my mobile proxy the other one. But when I call the mobile the call doesn't even get to my fix proxy because my mobile proxy has to make the original call to wait while he is forking the fix call to my other proxy. And i think that calling the async_route fonction makes both calls to wait. My code is: Fix proxy: if($tU==123456789){ add_diversion("forking"); append_branch("sip:987654321@proxy_mobile"); } Mobile proxy: if($tU==987654321){ $var(z) = "1"; append_branch("sip:123456789@proxy_fix"); } route[INVITE]{ if($dir == "forking"){ async_route("RELAY", "7"); }else if($var(z) == "1"){ async_route("RELAY", "7"); }else{ route(RELAY); exit; } } How can I differentiate in the second scenario when I call the mobile phone to fork to the fix phone? Thank you for your help. Igor.

2 3

Kamailio send TCP packets from random ports
by Yuriy Gorlichenko 08 Sep '16

08 Sep '16

Hi. I try to make working kamailio on TCP infront of asterisk Before to send to asteisk any packet i added $fs=ip.add.re.ss:port Also discribed listen=tcp:ip.add.re.ss:port But kamailio send outgoing packets from random prot throug TCP Presume i configured 5060 port but it send from 35410 port. googling on a core functions didn gives me any answer. May be i forget something to set at the config?

3 12

Re: [SR-Users] Kamailio send TCP packets from random ports
by Yuriy Gorlichenko 08 Sep '16

08 Sep '16

I know about dispatcher but it not always canbe heplfull. Sometimes i need my own logic that not implemented at the dispatcher. For checking kamailio live from asterisk im use qualfy for keepalives, if i use asterisk only as media server but it not always usefull. Sometimes scenarios need to use another logic. 2016-09-08 18:24 GMT+03:00 Yuriy Gorlichenko <ovoshlook(a)gmail.com>: > I know about dispatcher but it not always canbe heplfull. Sometimes i need > my own logic that not implemented at the dispatcher. > > For checking kamailio live from asterisk im use qualfy for keepalives, if > i use asterisk only as media server but it not always usefull. Sometimes > scenarios need to use another logic. > > 2016-09-08 17:50 GMT+03:00 Federico Cabiddu <federico.cabiddu(a)gmail.com>: > >> The issue with dispatcher is that, in case of TCP transport, you cannot >> set the sending socket for the same reason. >> Basically, each time a client TCP socket is open by Kamailio, the SO >> select the port, due to the lack of support for SO_REUSEPORT. >> >> Cheers, >> >> Federico >> >> On Thu, Sep 8, 2016 at 4:33 PM, Daniel Tryba <d.tryba(a)pocos.nl> wrote: >> >>> On Thu, Sep 08, 2016 at 03:38:32PM +0300, Yuriy Gorlichenko wrote: >>> > I didnt thought about keepalive. I suppose it can help. >>> >>> Better than using qualify in asterisk is to use the dispatcher module in >>> kamailio. The idea is the same, but more configurable and it is just 1 >>> keepalive mechanisme so the amount of keepalives doesn't scale linear >>> with the number of registered 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 >>> >> >> >

1 0

Kamailio 4.4.2 S-CSCF DB error
by Owais 08 Sep '16

08 Sep '16

I can see queries by S-CSCF like: The subscriber table doesn't contain presentity_uri as per: http://kamailio.org/docs/db-tables/kamailio-db-4.4.x.html#idp58608 <http://kamailio.org/docs/db-tables/kamailio-db-4.4.x.html#idp58608> Please share a valid mysql schema for tables used by s-cscf and presence. Thanks, Owais -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-4-4-2-S-CSCF-DB-error-tp15… Sent from the Users mailing list archive at Nabble.com.

1 0

UAC reg_contact_addr with rinstance
by DanB 08 Sep '16

08 Sep '16

Hey Guys, Have hit an issue when using uacreg. When working with Broadsoft (I know, life is not perfect), it appears that their SBC sitting in front of registration server is caching REGISTRATION requests and it will not be able to re-REGISTER in case of credential changes for example. This will break the registration process in registrar since that one flushes the registrations on password change. Their "standard" workaround it is using "rinstance" uri parameter in contact (each contact should be unique in order to be cached by their SBC). Checking it appears that I can only add rinstance statically in reg_contact_addr configuration parameter. Is there any chance to provide things like $avp or selects which I can later modify via mi maybe? Thanks in advance for any kind of tip! DanB

2 6

Query - REST API calls in async
by Agalya RAMACHANDRAN 08 Sep '16

08 Sep '16

Hi team, I am Agalya and I work for Comcast. We are trying to use kamailio as proxy server for our project. I have a query regarding the same. We are trying to use REST API calls asynchronously from the config file. I see that there is a module "http_async_client" which is used to handle REST services asynchronously. Found that libcurl is used to perform these operations. My question is: 1) How much time kamailio is taking to send the HTTP request out after TCP port is opened? Is there any time dependency we need to wait on libcurl to send HTTP request out? B'coz in opensips I see a delay time and libcurl is used in-turn. I found kamailio doing similar to opensips and hence am posting this question. 2) We can also use PUT service by using http_set_method(PUT)? Should we add code to support PUT service? Thanks in advance. Regards, Agalya

2 1

Mitigating DDOS attacks from carrier based on ani or dnis by limitating cps or blocking when detecting
by anfecora 08 Sep '16

08 Sep '16

Hi is there any way to use pike and htable to mitigate ddos or flood attack from trusted trunks. I need help to build it the same way kamailio control registrations. Case a carrier trunk star calling several users from the system to more than 50 CPS(calls per second), it will affect the system but cannot block the trunk since it is pstn traffic coming from a sip provider, therefore we need to find a way to identify this traffic based on ANI or DNIS or any other header and then blocked for a time just like pike does with registrations, then start the cycle all over. in less words make kamailio be aware of invite request rate, then verify that is from the same source means ani or dnis then tagged as bad traffic them star dropping it for a specified time, while normal traffic still flowing unaffected. any recommendations will be highly appreciated. thank you

2 3

New links for "IMS getting started"
by Franz Edler 07 Sep '16

07 Sep '16

Hi, Due to some storage limits I recently encountered I had to temporarily remove the image from my drop-box. Now this issue is fixed and I have uploaded the image again. Here are the links: The VMware image: https://www.dropbox.com/s/swhlgzpflgl3p3e/Debian_8.2-step5-basicKamailio%26b asicIMScore.zip?dl=0 The config-files: https://www.dropbox.com/s/nm51v4orxrox4ac/Kamailio-IMS%20config-files.zip?dl =0 Good luck. Franz DI Franz Edler University of Applied Sciences VIENNA, Austria

4 3

deb.kamailio.org down?
by Carsten Bock 06 Sep '16

06 Sep '16

Hi, currently, deb.kamailio.org is down. Is anyone taking care of this? Thanks, Carsten -- Carsten Bock CEO (Geschäftsführer) ng-voice GmbH Millerntorplatz 1 20359 Hamburg / Germany http://www.ng-voice.com mailto:carsten@ng-voice.com Office +49 40 5247593-40 Fax +49 40 5247593-99 Sitz der Gesellschaft: Hamburg Registergericht: Amtsgericht Hamburg, HRB 120189 Geschäftsführer: Carsten Bock Ust-ID: DE279344284 Hier finden Sie unsere handelsrechtlichen Pflichtangaben: http://www.ng-voice.com/imprint/

4 5

Resume suspended invite transation after x seconds
by José Seabra 06 Sep '16

06 Sep '16

Hello, I have the following scenario: Kamailio is suspending an INVITE then generates a SUBSCRIBE msg to an external server with UAC module, and only resume the suspended INVITE when receives an NOTIFY(this NOTIFY "belongs" to the SUBSCRIBE dialog) from the external server. The issue is that sometimes kamailio doesn't receive a NOTIFY and the transaction is not resumed, any one have an idea how can i implement a solution to resume the suspended invite after x seconds without receiving NOTIFY msg? Best Regards -- José Seabra

2 4

detect failed socket
by Péter Barabás 05 Sep '16

05 Sep '16

Hello, I want to refine the instant message handling in our application. It works really fine in most cases. I use the msilo module to store offline messages if the user is not registered with sending PUSH notification. But, there is a situation when I have problems: When I send a message to a user whose phone is in airplane mode. In this case his location record exists but the communication on the socket is not possible and the client timed out the request. Is it possible to check the socket towards the receiver user at the moment when the message arrives at Kamailio? I think if I would check that if the socket is not fine, I could send the message to the msilo directly and the client would not have wait for timeout to resend. Thanks. Peter

5 9

presence modules not send NOTIFY events
by Yuriy Gorlichenko 05 Sep '16

05 Sep '16

Hi. I trying to implement full presence server I set config like this (part of it) #!define FLT_DLG 9#!define FLT_DLGINFO 10 modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_flag", 4) modparam("dialog", "initial_cbs_inscript", 1) modparam("dialog", "profiles_with_value", "caller") modparam("dialog", "track_cseq_updates", 1) modparam("dialog", "enable_stats", 1) modparam("dialog", "dlg_flag", FLT_DLG) modparam("presence", "db_url", DBURL) modparam("presence", "server_address", SIPURI) modparam("presence", "send_fast_notify", 1) modparam("presence", "db_update_period", 20) modparam("presence", "clean_period", 40) modparam("presence", "subs_db_mode", 1) modparam("presence", "fetch_rows", 1000) modparam("presence", "local_log_level", 3) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #-------presence_dialoginfo params ----- modparam("presence_dialoginfo", "force_dummy_dialog", 1) modparam("presence_dialoginfo", "force_single_dialog",1) # ----- pua params ----- modparam("pua", "db_url", DBURL) modparam("pua", "db_mode", 2) modparam("pua", "update_period", 10) modparam("pua", "dlginfo_increase_version", 0) modparam("pua", "reginfo_increase_version", 0) modparam("pua", "check_remote_contact", 1) modparam("pua", "fetch_rows", 1000) # ----- pua_dialoginfo params ----- modparam("pua_dialoginfo", "include_callid", 1) modparam("pua_dialoginfo", "send_publish_flag", FLT_DLGINFO) modparam("pua_dialoginfo", "caller_confirmed", 0) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "override_lifetime", 124) modparam("presence", "db_url", DBURL) modparam("presence", "server_address", SIPURI) modparam("presence", "send_fast_notify", 1) modparam("presence", "db_update_period", 20) modparam("presence", "clean_period", 40) modparam("presence", "subs_db_mode", 1) modparam("presence", "fetch_rows", 1000) modparam("presence", "local_log_level", 3) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #-------presence_dialoginfo params ----- modparam("presence_dialoginfo", "force_dummy_dialog", 1) modparam("presence_dialoginfo", "force_single_dialog",1) # ----- pua params ----- modparam("pua", "db_url", DBURL) modparam("pua", "db_mode", 2) modparam("pua", "update_period", 10) modparam("pua", "dlginfo_increase_version", 0) modparam("pua", "reginfo_increase_version", 0) modparam("pua", "check_remote_contact", 1) modparam("pua", "fetch_rows", 1000) # ----- pua_dialoginfo params ----- modparam("pua_dialoginfo", "include_callid", 1) modparam("pua_dialoginfo", "send_publish_flag", FLT_DLGINFO) modparam("pua_dialoginfo", "caller_confirmed", 0) modparam("pua_dialoginfo", "include_tags", 1) modparam("pua_dialoginfo", "override_lifetime", 124) At the config file i made some changes route[BLF] { # absorb retransmissions if (! t_newtran()) { sl_reply_error(); exit; }; if(is_method("PUBLISH")) { if (handle_publish()) { xlog("L_INFO","{$rm} handled"); }; t_release(); } else if( is_method("SUBSCRIBE")) { if (handle_subscribe()) { xlog("L_INFO","{$rm} handled"); }; t_release(); }; exit; } Also added for evey call dlg_manage() function So for now i have issue that no NOTIFY event was sent at the any update of any device I see PUBLISH events and SUBSCRIBE events but it looks like handle_publish() and handle_subscribe() not sending any NOTIFY But at the dump i see xlog("L_INFO","{$rm} handled"); where $rm is correct SUBSCRIBE and PUBLISH if it is. Just dont know what i need to change. I will be very grateful if someone heps me to resolve this issue. Thanks

2 2

Kamailio - passing registration to another sip server
by Mike Patterson 05 Sep '16

05 Sep '16

I have configured Kamailio to pass a registration request to another VoIP provider. The reason I am doing this is to provide a sip port for users where the ISP is blocking SIP. I am able to pass the registrations to the VoIP provider but I'd like to have inbound calls working as well. I see the inbound invites coming into the Kamailio server but it appears that the server does not 'know' about the UA that is registered 'through' the server to the VoIP provider. Can anyone tell me how to fix this problem? Below is my config. Thank you, Mike #!KAMAILIO # # Edge proxy configuration # #!subst "/REGISTRAR_IP/xxxxxxxxxx/" #IP address of voip provider #!subst "/REGISTRAR_PORT/5060/" #!substdef "/FLOW_TIMER/20/" debug=2 log_stderror=no log_facility=LOG_LOCAL0 fork=yes children=4 alias="example.com" #mpath="/usr/lib64/kamailio/modules" mpath="/usr/lib/x86_64-linux-gnu/kamailio/modules" tcp_connection_lifetime=30 # FLOW_TIMER + 10 auto_aliases=no mhomed=0 listen=udp:X.X.X.X:531 advertise X.X.X.X:531 alias="X.X.X.X:531" alias=X.X.X.X:531 alias="X.X.X.X:531" alias=X.X.X.X:531 port=531 ####### Modules Section ######## loadmodule "tm.so" loadmodule "sl.so" #loadmodule "outbound.so" loadmodule "rr.so" loadmodule "path.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "mi_rpc.so" loadmodule "mi_fifo.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "rtimer.so" loadmodule "sqlops.so" # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- tm params ----- modparam("tm", "failure_reply_mode", 3) # ----- rr params ----- modparam("rr", "append_fromtag", 0) ####### Routing Logic ######## request_route { route(REQINIT); if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } route(WITHINDLG); t_check_trans(); if (is_method("REGISTER")) { remove_hf("Route"); add_path(); $du = "sip:REGISTRAR_IP:REGISTRAR_PORT"; } else { if (is_method("INVITE|SUBSCRIBE")) record_route(); if ($si == "REGISTRAR_IP" && $sp == "REGISTRAR_PORT") { if (!loose_route()) { switch($rc) { case -2: sl_send_reply("403", "Forbidden"); exit; default: sl_reply_error(); exit; } } t_on_failure("FAIL_OUTBOUND"); } else { if ($rU == $null) { sl_send_reply("484", "Address Incomplete"); exit; } remove_hf("Route"); $du = "sip:REGISTRAR_IP:REGISTRAR_PORT"; } } route(RELAY); } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } route[REQINIT] { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } route[WITHINDLG] { if (has_totag()) { if (!loose_route()) { switch($rc) { case -2: sl_send_reply("403", "Forbidden"); exit; default: if (is_method("ACK")) { if ( t_check_trans() ) { route(RELAY); exit; } else { exit; } } sl_send_reply("404","Not Found"); } } else { if (is_method("NOTIFY")) { record_route(); } route(RELAY); } exit; } } onreply_route { if (!t_check_trans()) { drop; } if ($rm == "REGISTER" && $rs >= 200 && $rs <= 299) { remove_hf("Flow-Timer"); if ($(hdr(Require)[*])=~"outbound") insert_hf("Flow-Timer: FLOW_TIMER\r\n", "Call-ID"); } } failure_route[FAIL_OUTBOUND]{ if (t_branch_timeout() || !t_branch_replied()) { send_reply("430", "Flow Failed"); } }

4 9

VUC - SER-Kamailio At 15 Years - On Friday, Sep 2, 2016
by Daniel-Constantin Mierla 02 Sep '16

02 Sep '16

Hello, on September 3, 2001, the first line of code was committed to SIP Express Router (SER) project, since 2008 turning into Kamailio project. >From pioneering large scale deployments of using SIP for Voice over IP in early 2000s, the project impacted the evolution of real time communications, disrupted the telecom vendors with its open source model and enabled many businesses to grow and innovate. Let's get together to a VUC session later this week, on Friday, Sep2, 2016, to share the good or bad stories while using SER or Kamailio during the past 15 years. If it made your life easy or a nightmare, it's time to tell it. If you have a funny photo or video, no matter is old or recent, join us and show it. More details about VUC and how one can join or watch the session are available at: - https://vuc.me If you want to actively participate during the session, email me shortly to coordinate a bit. If you cannot participate but you have something interesting to share (story, media files, etc...) email me as well and I will present it. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

IMS RFCs
by Owais Ahmad 02 Sep '16

02 Sep '16

Hi guys, Can you list the RFCs and 3GPP standards versions for each of the CSCFs in Kamailio? Regards, Owais

1 0

Kamailio with Dispacher
by Rodrigo Moreira 02 Sep '16

02 Sep '16

Hello, I would like to clarify a question. Can you help me? I'm deploying a solution based on Kamailio, but I need incorporate the load balancer funcionality in the environment. See the cenario: I have five VMs. PCSCF, ICSCF, SCSCF, HSS, DNS. Each VM runs the especific funcionality of the IMS. Therefore, which machines should receive the config provided by dispacher module? Is the PCSCF? Another question: The DNS server should be updated, I may do this previously? The HSS, should be updated too, I may do this previously? Do you known about approaches of the LB available? Best Regards. Thanks. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

2 1

issue with memcached module
by jay binks 02 Sep '16

02 Sep '16

if I use the memached module in kamailio to set STRING data in memcache, this is not readable from PHP. This is because kamailio uses the "val_flag" to distinguish the type of data ( int or string ). There does not appear to be a standard for how data type is indicated ( in fact the protocol doc for memcache says "<flags> is an arbitrary 16-bit unsigned integer" ) I think the memcache module really needs a way to let the user specify the value of the flag. something like a modparam to override the flag value on every set. has anyone else run in to this ? Sincerely Jay Binks

2 2

Centos, kamailio 4.4 and jansson module load issue.
by Jonathan Hunter 01 Sep '16

01 Sep '16

Hi Guys, I am running Centos 6.5 and have installed kamailio 4.4 from Git. All is running well until I try and load the jansson module. I have installed jansson-2.7 from src, but when loading the module in kamailio I get; Aug 30 20:06:24 sip1 kamailio: ERROR: <core> [sr_module.c:582]: load_module(): could not open module </usr/local/kamailio-4.4/lib64/kamailio/modules/jansson.so>: /usr/local/kamailio-4.4/lib64/kamailio/modules/jansson.so: undefined symbol: json_array_set_new Anyone else had this issue? Thanks Jon

3 5

Domain limits and domain.reload
by Bruno Emer 01 Sep '16

01 Sep '16

Hello all. I am currently working in a company that built an online platform that allows companies talk to their customer using different medias, such as video, voice and chat. To this, we are using Kamailio to provide SIP services such as WebSockets, Registrar and proxy, and ir works really well. The point is that today we are creating a new domain for each company that uses our system, and for now we have more than 3k companies using it, so my first question is: is there a limit about how many domains we can create in a Kamailio environment? I tried to find this on the documentation but I haven't found anything. Second: every time that we create a new domain, we perform a domain.reload using the XMLRPC module. We are just using the python script to do this, but sometimes it is returning the error: Fault 400: 'Reload of domain tables failed' and we don't know why. The strange thing is that if we access the server and execute a 'kamctl domain reload' just after the error, everything works fine. The problem appears to be in the RPC module. So the question is: does someone knows what can be the cause of this problem? Thanks. *Bruno Emer* Mobile: +55 11 96540-0044 email: brunoemer(a)gmail.com <https://www.facebook.com/bruno.emer.5> <https://www.linkedin.com/profile/view?id=86461237&trk=spm_pic> <https://twitter.com/brunoemer_> <https://www.flickr.com/photos/122070309@N03/> <http://google.com/+BrunoEmer> <http://instagram.com/brunoemer_>

2 1

kamailio Bigning
by Gaurav Bmotra 01 Sep '16

01 Sep '16

hi i m new with kamailio so ple help me what i need to next i have install kamailio in centos 6.7 and also testing SIP to SIP call which is working but now i want to do some practice on routing (in kamailio.cnfg) so tell me what are the basics of it and what task i need to do for practice Thank you -- *Regards:* Gaurav Bmotra Smith Infotech E-46, phase 8 , Indus Area S.A.S Nagar Mohali 160071

2 1

Kamailio Errors that I see in the system logs
by Kevin Greene 01 Sep '16

01 Sep '16

Hi: Looking for help in this logging event. Aug 26 22:34:17 sipserver /usr/sbin/kamailio[14790]: ERROR: db_mysql [km_dbase.c:124]: db_mysql_submit_query(): driver error on query: PROCEDURE kamailio.kamailio_cdrs does not exist (1305) Aug 26 22:34:17 sipserver /usr/sbin/kamailio[14790]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Aug 26 22:34:17 sipserver /usr/sbin/kamailio[14790]: ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query [call kamailio_cdrs()] Aug 26 22:34:17 sipserver /usr/sbin/kamailio[14790]: ERROR: db_mysql [km_dbase.c:124]: db_mysql_submit_query(): driver error on query: PROCEDURE kamailio.kamailio_rating does not exist (1305) Aug 26 22:34:17 sipserver /usr/sbin/kamailio[14790]: ERROR: <core> [db_query.c:181]: db_do_raw_query(): error while submitting query Aug 26 22:34:17 sipserver /usr/sbin/kamailio[14790]: ERROR: sqlops [sql_api.c:265]: sql_do_query(): cannot do the query [call kamailio_rating('default')] I assume the database is missing table entries for these two data inserts. Any help would be useful. Thanks. Kevin

3 2

Htable keys and key::size inconsistencies
by Fabian Borot 01 Sep '16

01 Sep '16

Hi, I am using htable module with version 4.4 storing multiple values in the same key (I set the key_type = 1 for each entry, array key). I have found an issue with the htable module. Some keys show up when using: "./kamcmd htable.get my_table key[0]" but when asking for the "key::size" I says that the key does not exist. Also some entries have the "key::size" entry but the actual "key[0]" for example is reported as non existent: See these 2 examples: Ex:1 "key[0]" is shown but "key::size" is not: [root@ar-mia-proxy-01 sbin]# ./kamcmd htable.get htable_lcrgwA 8[0] { item: { name: 8[0] value: 555|5875|1.1.1.1|5060|0|4|1|1|Broadvox2010IN flags: 2 expire: NEVER } } [root@ar-mia-proxy-01 sbin]# ./kamcmd htable.get htable_lcrgwA 8::size error: 500 - Key name doesn't exist in htable. Ex2: "key::size" is shown but "key[0]" is not: [root@ar-mia-proxy-01 sbin]# ./kamcmd htable.get htable_lcrgwA 9[0] error: 500 - Key name doesn't exist in htable. [root@ar-mia-proxy-01 sbin]# ./kamcmd htable.get htable_lcrgwA 9::size { item: { name: 9::size value: 6 flags: 0 expire: NEVER } } Also, for this case "9::size", supposedly there are 6 entries and in reality there are only 4 in the tables where I am loading them from For the case "8::size" that returns 2 the count it is right , the prob is that I can not get the "8[0]" and "8[1]" keys Any ideas are very appreciated, thank you

2 1

kamailio outgoing gateway
by Andrzej Kaczmarczyk 31 Aug '16

31 Aug '16

I have common, cheap TL-WR1043ND V1 router. For private purposes I've set kamailio voip on my own server INSIDE my LAN. Is it possible to connect to that server from outside, through openWRT router? Many tries, no success - too many jumps or something similar. I guess router should be fully transparent for sip protocol. How to manage it? is it possible at all? I don't ask how to do it in details on my router/openwrt (but if someone knows, it will be nice), but - is this possible to setup at all? should I try or search for resolution, or just give up with that idea? -- Pozdrawiam Andrzej Kaczmarczyk

3 2

Segfault in qm_insert_free
by Igor Potjevlesch 31 Aug '16

31 Aug '16

Hello, We got a crash yesterday of a Kamailio instance (4.2.7). I can see a segfault: "Program terminated with signal 11, Segmentation fault. #0 0x000000000061a856 in qm_insert_free (qm=0x7fcbde513000, frag=0x7fcbded7dd08) at mem/q_malloc.c:180 180 prev=FRAG_END(f)->prev_free;" The output of "bt full": (gdb) bt full #0 0x000000000061a856 in qm_insert_free (qm=0x7fcbde513000, frag=0x7fcbded7dd08) at mem/q_malloc.c:180 f = 0x7fcbdf2d1e18 prev = 0x0 hash = 78 #1 0x000000000061d6dc in qm_free (qm=0x7fcbde513000, p=0x7fcbded7dd38, file=0x7fcbf4ffe34d "tm: h_table.c", func=0x7fcbf4ffe628 "free_cell", line=186) at mem/q_malloc.c:537 f = 0x7fcbded7dd08 size = 624 next = 0x400 prev = 0x7fffc283f720 __FUNCTION__ = "qm_free" #2 0x00007fcbf4f3fc9d in free_cell (dead_cell=0x7fcbdf2fa360) at h_table.c:186 b = 0x7fcbded7dd38 "PRACK sip:1.2.3.4:5060;transport=UDP SIP/2.0\r\nVia: SIP/2.0/UDP 4.5.6.7;branch=z9hG4bK7419.88222290c54ce503af46b89a1da0c125.0\r\nVia: SIP/2.0/UDP 9.8.7.6:5060;branch=z9hG4bK0cB6cd54b5"... i = 0 rpl = 0x0 tt = 0x7fcbde6b7778 foo = 0x7fffc283f7f0 cbs = 0x0 cbs_tmp = 0x7fcbde79fca0 __FUNCTION__ = "free_cell" #3 0x00007fcbf4f84a1c in wait_handler (ti=1160550038, wait_tl=0x7fcbdf2fa3e0, data=0x7fcbdf2fa360) at timer.c:675 p_cell = 0x7fcbdf2fa360 ret = 1 #4 0x00000000005fd30f in timer_list_expire (t=1160550038, h=0x7fcbde58c908, slow_l=0x7fcbde58ea28, slow_mark=19953) at timer.c:888 tl = 0x7fcbdf2fa3e0 ret = 1160550038 #5 0x00000000005fd757 in timer_handler () at timer.c:953 saved_ticks = 1160550038 run_slow_timer = 0 i = 497 __FUNCTION__ = "timer_handler" #6 0x00000000005fdbc5 in timer_main () at timer.c:992 No locals. #7 0x00000000004a77e6 in main_loop () at main.c:1700 i = 8 pid = 0 si = 0x0 si_desc = "udp receiver child=7 sock=91.213.145.60:5060\000\177\000\000\000\372\203\302\377\177\000\000\033{ N\000\000\000\000\000P\372\203\302\377\177\000\000\004\000\000\000\000\000\0 00\000`TA\000\000\000\000\000(\205T\336\313\177", '\000' <repeats 14 times>, "\001\000\000\000P\372\203\302\377\177\000\000\276{N\000\000\000\000" nrprocs = 8 __FUNCTION__ = "main_loop" #8 0x00000000004acfab in main (argc=7, argv=0x7fffc283fcc8) at main.c:2581 cfg_stream = 0xe5e010 c = -1 r = 0 tmp = 0x7fffc283ff70 "" tmp_len = 32767 port = -1031537762 proto = 0 options = 0x7033b8 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 1876790668 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x40d134 p = 0xc2 <Address 0xc2 out of bounds> __FUNCTION__ = "main" I don't figure out what is gone wrong. Thank you for your help. Regards, Igor.

1 1

Looking for network version of ds_reload
by Chad McElligott 30 Aug '16

30 Aug '16

Hello all, I'm in the process of converting an on-host install of Kamailio to be run inside a docker container. A part of this Kamailio install is a small node.js application that modifies the sqlite database Kamailio uses, then runs `kamctl fifo ds_reload` to tell Kamailio to reload its knowledge from that database. I'm trying to split these two processes (the node app and the Kamailio server) into separate docker containers, in an effort to follow the one-process-per-container philosophy, but having trouble figure out exactly how I should perform the ds_reload step without access to the kamctl in the Kamailio container. The obvious direction that I'm looking is to perform some sort of network request, since I can configure the two docker containers to have network access to each other. Is there some way to perform a ds_reload over the network? Thanks for all you do. Kamailio is a great SIP server and we use it to great effect. -Chad McElligott

2 4

uac_auth and (serial) forking
by Daniel Tryba 30 Aug '16

30 Aug '16

Trying to implement a way to authenticate outgoing INVITEs to endpoints that need/want to. uac_auth in the initial failure route works fine when there is only 1 registered user to send the INVITE to. But with at least 2 locations are available and 1 accepts the INVITE without authentication, the 401 of the location that wants authentication isn't processed in the failure route until a timeout is triggered (t_set_fr). At that point the 401 is processed and a response generated with Authentication header. But since t_set_fr invite timeout was exceeded it is immediatly destroyed due to a CANCEL having been generated to the location(s). t=0: t_set_fr(10000); INVITE to A anb B A responds 100 Trying,180 Ringing B responds 401 Unauthorized t=10s: T_code=180, new_code=408 failure route status: 401 uac_auth() INVITE with Authorization is build to B CANCEL to A INVITE to B is never send Is this expected behaviour? Obviously I didn't expect this.

2 4

Configuration Issue on Kamailio.
by Laura 30 Aug '16

30 Aug '16

Dear list, i'm asking here a question about Kamailio config. We are testing a wide area configuration of Kamailio over separates countries and we are still facing with an issue. We configured Kamailio 4.3.5 with dialog support over the TM modules and we use LCR module for menage ours LCRs rule set profiles. For some technicals reasons we use tech prefix for our customer so for exaples customer1 send traffic to us with 1111 prefix, customer2 send traffic to us with 2222 and something similar.. Our supplier, of course, are using tech prefix too so for examples if i want to send the call to supplier1 i need to use tech prefix 1789 or something similar.. The point is.. When customer1 is sending an invite to us.. it send us something like (Bangladesh mobile 8801xxx) INVITE sip:11118801xxxxxxx@aaa.bbb.ccc.ddd Our Kamailio will reply with the Trying and then it goes to LCR module and match our supplier1 so it make a new invite like this INVITE sip:17898801xxxxxx@supplier.ip The problem come when supplier1 reply to us and we replies back to customer1.. Customer1 view the From: field with the 17898801xxxxxx numbers.. and some of our customers don't like it. We don't use anymore the topoh module becuase we found some troubles using it.. so.. Is there a way that we can use for fix this situation ? Best regards.

4 7

Kamailio and user registration and a question on STUN servers
by Kevin Greene 29 Aug '16

29 Aug '16

Hello Group: I am looking for a way to allow any SIP client to register to my (development) network without having to be already registered in the database. Is there a way for the user client to register and create a valid user subscription in the database? The SIP clients I am using have hard coded passwords and I can not change them. I think once loaded into the database, I can delete the ones I do not want later after I close automatic registration feature... The second question is on STUN servers. Is there a good way to test to see if one is working from outside your network? I was looking for some sort of web based tool I could access to see if my public STUN server is really working. I am using the "reTurnServer" software on Linux. I am only allowing TCP based STUN requests. Thank you. Sample log entry for a external user trying to access the development network. I would like to give this person access to place calls on the system. 199.58.xxx.xxx.nebraskalink.com - - [25/Aug/2016:18:02:49 +0000] "GET /download/?ojo-mac=00:02:54:00:a5:eb&ojo-id=2000-0005-2F52&ojo-rev=rev02.01.15&ojo-config-id=e712e6cb088ab9b4ef246488b979cbf6&ojo-stun-type=Blocked&ojo-sn=15190252560146703100&ojo-key-id=e0142f126a5197845a24f33deedc358645e67e5f HTTP/1.0" 200 1812 "-" "Wget/1.9.1"

3 2

Kamailio S-CSCF Error
by Rodrigo Moreira 29 Aug '16

29 Aug '16

Hello, How can I deal with the error: Aug 13 13:55:43 scscf kamailio[2961]: tcp: 192.168.0.101 [192.168.0.101]:5060 Aug 13 13:55:43 scscf kamailio[2961]: Aliases: Aug 13 13:55:43 scscf kamailio[2961]: *: scscf.k-ims.test:5060 Aug 13 13:55:43 scscf kamailio[2961]: 0(2963) ERROR: ims_usrloc_scscf [ul_mod.c:294]: mod_init(): can't bind presence Aug 13 13:55:43 scscf kamailio[2961]: 0(2963) ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module ims_usrloc_scscf (/usr/lib/x86_6..._scscf.so) Aug 13 13:55:43 scscf kamailio[2961]: ERROR: error while initializing modules Aug 13 13:55:43 scscf kamailio[2961]: 0(2963) CRITICAL: cdp [diameter_peer.c:487]: diameter_peer_destroy(): destroy_diameter_peer(): Bye Bye from C Diameter Peer test Aug 13 13:55:43 scscf systemd[1]: kamailio.service: control process exited, code=exited status=255 Aug 13 13:55:43 scscf systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Aug 13 13:55:43 scscf systemd[1]: Unit kamailio.service entered failed state. Thank you. Regards., -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

2 2

ims_dialog- dlg set error
by Suresh.Waware＠relianceada.com 26 Aug '16

26 Aug '16

Hi, We are using kamailio 4.4.2 version source code. For building SCSCF we are using example configuration files provided in the source code. While running SCSCF; we are facing following error. ERROR: ims_dialog [ims_dialog.c:378]: mod_init(): no dlg flag set!! q Regards, Suresh The information contained in this electronic message (email) and any attachments to this email are intended for the exclusive use of the addressee(s) and access to this email by any one else is unauthorised. The email may contain proprietary, confidential or privileged information or information relating to Reliance Group. If you are not the intended recipient, please notify the sender by telephone, fax, or return email and delete this communication and any attachments thereto, immediately from your computer. Any dissemination, distribution, or copying of this communication and the attachments thereto (in whole or part), in any manner, is strictly prohibited and actionable at law. The recipient acknowledges that emails are susceptible to alteration and their integrity can not be guaranteed and that Company does not guarantee that any e-mail is virus-free and accept no liability for any damage caused by any virus transmitted by this email.

3 2

Change debug level without restarting Kamailio
by Federico San Martín 26 Aug '16

26 Aug '16

Hi guys, sorry for the newbie question, I searched for it and couldn't find any reference, is there a way to change the debug level of a kamailio server on the fly? If I need to add an additional module it's not a problem. I'm using version 4.4.2 on Debian Jessie. Thanks! Federico

3 3

kamailio crushes because of tcp errors
by Dmitry 26 Aug '16

26 Aug '16

hel >From time to time (it does not depend upon traffic) we see in logs: ERROR: <core> [tcp_read.c:274]: tcp_read_data(): error reading: Bad file descriptor (9) ([ip_address]:49832 -> [ip_address]5060) ERROR: <core> [tcp_read.c:1303]: tcp_read_req(): ERROR: tcp_read_req: error readingERROR: <core> [io_wait.h:608]: io_watch_del(): trying to delete already erased entry 3031 in the hash(0, 0, (nil)) flags 10)CRITICAL: <core> [tcp_read.c:1654]: handle_io(): io_watch_del failed for 0x7f8471c51820 id 867 fd 3031, state 0, flags 4018, main fd 179, refcnt 2 ([ip_address]:49832 -> [ip_address]:5060) there are a lot of such TCP CRITICAL errors and the kamailio crushes kamailio -vversion: kamailio 4.3.5 (x86_64/linux)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_RESADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MBpoll method support: poll, epoll_lt, epoll_et, sigio_rt, select.id: unknowncompiled on 17:51:55 Mar 8 2016 with gcc 4.4.7 It is still not clear - what exactly causes such behaviour. I want to recriate it in the lab - so what can cause it.? May be a process erases the file descriptors of TCP connections? Any help is appreciated.

1 1

Checking if a reply is part of a forked transaction
by Phil Lavin 25 Aug '16

25 Aug '16

We need to check if a 1XX reply is part of an INVITE transaction that was forked to 2 or more phones. We figure that we can do a location lookup, based on the to header of the 1XX... but is there a better way? Phil

4 4

Fwd: Kamailio with SNMP
by Rodrigo Moreira 24 Aug '16

24 Aug '16

Hello, Once set up and running the Kamailio environment would like to install SNMP and incorporate the functionality to collect the statistics of IMS entities. It's possible? When I call the module "snmpstats" I am informed that the module is not present. How can I solve? Another issue, which modules are monitorable via SNMP, P-CSCF, I-CSCF, S- CSCF? All? Regards. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

2 1

Monitor diameter state and number of current calls
by Ramiro Romero Arjona 23 Aug '16

23 Aug '16

Hi, We would like to know if there is some way to check the state of a diameter connection (to see is it's in state 6: connected), inside the kamailio configuration script. Also we would like to know if there is a way to see the number of current calls in a kamailio through kamcmd, kamctl or something like that. We have searched for a solution but we don't find anything. Many thanks.

1 0

Send JSON document over http and process response
by Jonathan Hunter 22 Aug '16

22 Aug '16

Hi Guys, I need to send a JSON document over HTTP to a server, and then process the response, both in the kamailio configuration file. I assume I need to use Jansson module to extract contents of response? What is the best method for generating the initial request containing JSON ? Many thanks Jon

2 2

Howto timeout INVITEs over TCP to something < 30s?
by Daniel Tryba 22 Aug '16

22 Aug '16

If an enduser INVITEted over TCP t_set_fr(t) with t < 30s have no effect, the INVITE will always timeout after 30s. What is controlling this 30s? The only default at 30s is the fr_timer, which should have no part in the timeout between an 1xx and a final answer for an INVITE.

2 5

Evapi module is not broadcasting any data
by Efelin Novak 22 Aug '16

22 Aug '16

Hi folks, I came into a problem with evapi module, while I was trying to make the CGRateS tutorial work. It does not send any messages at all. Taken steps: I start kamailio with debug=3. I start: CGRates (on the kamailio machine) telnet 192.168.30.29 8448 (from my PC) tcpdump port 8448 -A -s0 -i any (on the kamailio machine) Telnet command successfully connects to kamailio. Kamailio yells: root@cgrates:/opt/kamailio/src/kamailio-4.4/kamailio# 11(12205) DEBUG: evapi [evapi_dispatch.c:453]: evapi_accept_client(): new connection - pos[1] from: [192.168.30.1:40312] 11(12205) INFO: <script>: Starting Evapi connector srcaddr 192.168.30.1 srcport 40312 condix 1 11(12205) DEBUG: htable [ht_var.c:90]: pv_set_ht_cell(): set value for $sht(cgrconn=>cgr) A similar message is printed when CGRateS connects in. Afterwards I make a call to Kamailio. INVITE is routed to a following command: evapi_async_relay("{\"event\":\"CGR_AUTH_REQUEST\", \"tr_index\":\"$T(id_index)\", \"tr_label\":\"$T(id_label)\", \"cgr_reqtype\":\"$dlg_var(cgrReqType)\", \"cgr_tenant\":\"$dlg_var(cgrTenant)\", \"cgr_account\":\"$dlg_var(cgrAccount)\", \"cgr_destination\":\"$dlg_var(cgrDestination)\", \"cgr_setuptime\":\"$TS\", \"cgr_computelcr\":\"true\"}"); Now the kamailio writes: . . . 2(12196) DEBUG: dialog [dlg_var.c:260]: print_lists(): Dialog var-list (0x7f196d1c5248): 2(12196) DEBUG: dialog [dlg_var.c:265]: print_lists(): cgrDestination=1002 (flags 1) 2(12196) DEBUG: dialog [dlg_var.c:265]: print_lists(): cgrAccount=1001 (flags 1) 2(12196) DEBUG: dialog [dlg_var.c:265]: print_lists(): cgrTenant= cgrates.org (flags 1) 2(12196) DEBUG: dialog [dlg_var.c:265]: print_lists(): cgrReqType=*prepaid (flags 1) 2(12196) DEBUG: dialog [dlg_hash.c:922]: dlg_unref(): unref dlg 0x7f196d1c4c08 with 1 -> 1 2(12196) DEBUG: evapi [evapi_dispatch.c:603]: evapi_relay(): relaying event data [{"event":"CGR_AUTH_REQUEST", "tr_index":"59559", "tr_label":"1886589610", "cgr_reqtype":"*prepaid", "cgr_tenant":"cgrates.org", "cgr_account":"1001", "cgr_destination":"1002", "cgr_setuptime":"1471623526", "cgr_computelcr":"true"}] (246) 2(12196) DEBUG: evapi [evapi_dispatch.c:628]: evapi_relay(): sending [0x7f196d1c88b8] [246:{"event":"CGR_AUTH_REQUEST", "tr_index":"59559", "tr_label":"1886589610", "cgr_reqtype":"*prepaid", "cgr_tenant":"cgrates.org", "cgr_account":"1001", "cgr_destination":"1002", "cgr_setuptime":"1471623526", "cgr_computelcr":"true"},] (251) 2(12196) DEBUG: evapi [evapi_dispatch.c:630]: evapi_relay(): Len of write is 8 2(12196) INFO: <script>: suspended transaction: 59559 / 1886589610 . . . However no reply from CGRateS comes, tcpdump is quiet (no message was sent or received), as well as my testing telnet session (nothing is received). The network doesn't seems to be the issue, as when I write "6:Efelin," to the telnet session, kamailio executes the route event_route[evapi:message-received] Versions: # dpkg -l | grep libev-dev ii libev-dev 1:4.15-3 amd64 # uname -a Linux cgrates 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt25-2+deb8u3 (2016-07-02) x86_64 GNU/Linux # kamailio -V version: kamailio 4.4.2 (x86_64/linux) 384e90 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: 384e90 compiled on 12:04:04 Aug 19 2016 with gcc 4.9.2 I wasn't able to find anything that might block the connection. Can anyone point me to, how to debug this issue further? Thanks for any advice. Kind regards Jan Muranyi

3 7

Contact header / bye
by Oliver Roth 19 Aug '16

19 Aug '16

Hi all I do have a problem with getting a propper interconnection to our new carrier.. Situation as following User --> Class5 switch --> Kamailio --> carrier I always get the ip of the class5 switch in the contact. But my carrier needs the kamailios contact ip - because of the ack handling. I replaced in the pstn routing parte the contact by remove_hf("Contact"); append_hf("Contact: <sip:xyz@a.b.c.d>\r\n"); This worked - but the bye did not work any more. Please help me on that topic ... I am searching for a solution for more than 8 weeks now ... KR, Oli

2 1

Running both 4.3 and 4.4 with the same database with permission module
by Daniel Tryba 19 Aug '16

19 Aug '16

I want to upgrade my cluster from 4.3 to 4.4, beginning with the spare for live testing. But since there is a change to the trusted table (which we don't use, only address) this isn't possible without some tricks. There is no override for skipping the version check on any permissions table. I need to share the subscriber/location/address/usr_preferences and db_aliases tables between the 4.3 and 4.4 machines. The only thing I can think of is: -patch the permissions module to accept the old version of trusted -create a 4.4 database and use federated tables to access the 4.3 data Not hard to implement any of these 2, but some other modules have the option to skip these checks, eg: http://www.kamailio.org/docs/modules/4.4.x/modules/auth_db.html#auth_db.p.v… Is there a reason not all modules have this flag (which you could use at your own risk)? Or am I missing something here?

3 4

To/From tags in uac_req()
by Julia Boudniatsky 18 Aug '16

18 Aug '16

Hello, Is it possible to set To/From tags in uac_req() ? We want to implement service like "color ring back tone". *Kamailio uac* *Asterisk* ----INVITE SDP --> <-----200 SDP------ ---------BYE ------> <------- 200 ------- I need to send in BYE the same tags. Kamailio 4.4.2. Thanks, Julia

3 4

Trouble with nat_uac_test()
by Noah Mehl 18 Aug '16

18 Aug '16

Hey all, I have a Kamailio implementation (version: kamailio 4.4.1 (x86_64/solaris) 12b048) as part of the Kazoo platform. I’m having an odd problem with an e2e ACK (after a 200OK after a T38 re-invite), which I’m pretty sure has to do with a Contact header rewrite due to the following (can review attached PCAP for specifics): if (nat_uac_test("3")) { force_rport(); fix_nated_contact(); } I have a couple of questions about this: A flag of 3, I don’t understand. I read the documentation <http://kamailio.org/docs/modules/4.4.x/modules/nathelper.html#nathelper.f.n…>, and there is a 1,2,4,8,etc… So, I’m guessing that a flag of 3, means 1+2? If #1 is true, then "the "received" test is used: address in the “Via” header is compared against source IP address of signaling” I don’t understand this test, as I’m pretty sure that the 200OK that’s getting re-written is properly constructed. I guess I’m trying to understand when one would want to use this test? Thanks! ~Noah

3 3

msilo route error
by Eng Hooda 18 Aug '16

18 Aug '16

Hello All, I am trying to enable msilo offline message delivery module, I used the official technical documentation : https://www.kamailio.org/docs/modules/stable/modules/msilo.html I managed to add load module statement and modparam statement without error. At this point msilo is still not active , it was obvious that I needed to add route statement. I copied the provided route statement , and edited it so will be similar to route functions that are already there. that was the outcome: # Routing to MSILO route[MSILO] { #!ifdef WITH_MSILO if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); exit; }; if (uri==myself) { { # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; exit; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); exit; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; exit; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("$ru")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; exit; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); exit; }; # forward anything else t_relay(); } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { exit; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("$ou")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } #!endif return; } but after that I restarted kamailio service and got the following errors: root@debian:~# service kamailio restart Job for kamailio.service failed. See 'systemctl status kamailio.service' and 'journalctl -xn' for details. root@debian:~# systemctl status kamailio.service -l ● kamailio.service - Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/lib/systemd/system/kamailio.service; enabled) Active: failed (Result: exit-code) since Wed 2016-08-17 21:35:41 EET; 4s ago Process: 3693 ExecStart=/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f $CFGFILE -m $SHM_MEMORY -M $PKG_MEMORY -u $USER -g $GROUP (code=exited, status=255) Main PID: 3470 (code=exited, status=0/SUCCESS) Aug 17 21:35:41 debian kamailio[3693]: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 556, column 5: bad command Aug 17 21:35:41 debian kamailio[3693]: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 613, column 12: syntax error Aug 17 21:35:41 debian kamailio[3693]: : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 613, column 12: Aug 17 21:35:41 debian kamailio[3693]: ERROR: bad config file (5 errors) Aug 17 21:35:41 debian kamailio[3693]: WARNING: <core> [ppcfg.c:221]: pp_ifdef_level_check(): different number of preprocessor directives: N(#!IF[N]DEF) - N(#!ENDIF) = 3 Aug 17 21:35:41 debian kamailio[3693]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Aug 17 21:35:41 debian kamailio[3693]: loading modules under config path: /usr/lib/x86_64-linux-gnu/kamailio/modules/ Aug 17 21:35:41 debian systemd[1]: kamailio.service: control process exited, code=exited status=255 Aug 17 21:35:41 debian systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Aug 17 21:35:41 debian systemd[1]: Unit kamailio.service entered failed state. line 556 is a parenthesis. { I checked the code for missing parentheses and they are all there. line 613 is t_relay(); also tried to run with #!define WITH_DEBUG , there was no further info to debug the cfg file. Also I tried to mimic some cfg examples like this one : http://telephonynetworks.blogspot.com.eg/2012/08/configuracion-de-kamailio-… but it uses an older version of kamailio 3.3 , and I noticed considerable changes in mpath statement, so that was unsuccessful. kamailio.cfg is attached with and without msilo route. For the experts out there Please Advise . system details : {(OS:Debian8 i386),(Kamailio4.4 with mySQL , UDP only)} Thank you all in advance. Best Regards, Eng Hooda

2 1

RTPEngine
by Jack Stevens 17 Aug '16

17 Aug '16

Hi Guys, Does anyone know how to enable logging for RTPengine Kind Regards From: sr-users [mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of Jonathan Hunter Sent: 17 August 2016 13:17 To: sr-users(a)lists.sip-router.org Subject: [SR-Users] Core/ signal 11 Segmenation fault on kamailio 4.3.5 Hi guys, We seem to have a regular occurrence where one of the child processes is crashing, and causing a coredump. Please see the output of the corefile; gdb kamailio /core.31568 GNU gdb (GDB) Red Hat Enterprise Linux (7.2-83.el6) Copyright (C) 2010 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-redhat-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /usr/sbin/kamailio...(no debugging symbols found)...done. [New Thread 31568] Reading symbols from /lib64/ld-linux-x86-64.so.2...Reading symbols from /usr/lib /debug/lib64/ld-2.12.so.debug...done. done. Loaded symbols for /lib64/ld-linux-x86-64.so.2 Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f / /etc/kamailio//kamaili'. Program terminated with signal 11, Segmentation fault. #0 0x00000000004deefd in send_fd () (gdb) bt full #0 0x00000000004deefd in send_fd () No symbol table info available. #1 0x00000000004e0bc1 in pp_substdef_add () No symbol table info available. #2 0x00000000004da13c in build_res_buf_from_sip_req () No symbol table info available. #3 0x00007fc99099f4a1 in ?? () No symbol table info available. #4 0x00000000921cd3c8 in ?? () No symbol table info available. #5 0x00007fc9921cd3f8 in ?? () No symbol table info available. #6 0x0000000000415eb0 in do_action () No symbol table info available. #7 0x00007fc9909a2841 in ?? () No symbol table info available. #8 0x00007fc991faf030 in ?? () No symbol table info available. #9 0x0000000000000004 in ?? () No symbol table info available. #10 0x0000000000000001 in ?? () No symbol table info available. #11 0x0000000000000002 in ?? () The last transaction appears to be a REGISTER message from a User-Agent: SIPAUA/0.1.001 which runs on an android device. We are looking to reproduce in lab environment but I just wondered if anyone had seen this on this version? Or can advise on the core dump output? Thanks Jon -- CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED This email, and any attachments, is intended only for the above addressee. It may contain private and/or confidential information. If you have received this email in error you are on notice of its status, please immediately notify the sender by return email then delete this message and any attachments. If you are not the addressee, except to notify the sender, you must not use, disclose, copy or distribute this email and/or its attachments. Netcall Telecom accepts no responsibility for any changes made to this message after it has been sent by the original author. Opinions or views expressed in this email may be those of the individual sender and not Netcall Telecom. Nothing in this email shall bind Netcall Telecom in any contract or obligation Netcall Telecom Ltd Registered in England 2831215. Registered Office : 3rd Floor, Hamilton House, 111 Marlowes, Hemel Hempstead, Herts, HP1 1BB

3 3

Core/ signal 11 Segmenation fault on kamailio 4.3.5
by Jonathan Hunter 17 Aug '16

17 Aug '16

Hi guys, We seem to have a regular occurrence where one of the child processes is crashing, and causing a coredump. Please see the output of the corefile; gdb kamailio /core.31568GNU gdb (GDB) Red Hat Enterprise Linux (7.2-83.el6)Copyright (C) 2010 Free Software Foundation, Inc.License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html>This is free software: you are free to change and redistribute it.There is NO WARRANTY, to the extent permitted by law. Type "show copying"and "show warranty" for details.This GDB was configured as "x86_64-redhat-linux-gnu".For bug reporting instructions, please see:<http://www.gnu.org/software/gdb/bugs/>...Reading symbols from /usr/sbin/kamailio...(no debugging symbols found)...done.[New Thread 31568]Reading symbols from /lib64/ld-linux-x86-64.so.2...Reading symbols from /usr/lib /debug/lib64/ld-2.12.so.debug...done.done.Loaded symbols for /lib64/ld-linux-x86-64.so.2Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio/kamailio.pid -f / /etc/kamailio//kamaili'.Program terminated with signal 11, Segmentation fault.#0 0x00000000004deefd in send_fd ()(gdb) bt full#0 0x00000000004deefd in send_fd ()No symbol table info available.#1 0x00000000004e0bc1 in pp_substdef_add ()No symbol table info available.#2 0x00000000004da13c in build_res_buf_from_sip_req ()No symbol table info available.#3 0x00007fc99099f4a1 in ?? ()No symbol table info available.#4 0x00000000921cd3c8 in ?? ()No symbol table info available.#5 0x00007fc9921cd3f8 in ?? ()No symbol table info available.#6 0x0000000000415eb0 in do_action ()No symbol table info available.#7 0x00007fc9909a2841 in ?? ()No symbol table info available.#8 0x00007fc991faf030 in ?? ()No symbol table info available.#9 0x0000000000000004 in ?? ()No symbol table info available.#10 0x0000000000000001 in ?? ()No symbol table info available.#11 0x0000000000000002 in ?? () The last transaction appears to be a REGISTER message from a User-Agent: SIPAUA/0.1.001 which runs on an android device. We are looking to reproduce in lab environment but I just wondered if anyone had seen this on this version? Or can advise on the core dump output? Thanks Jon

3 3

Stress Testing
by Jack Stevens 17 Aug '16

17 Aug '16

Hi Guys, I have been stress testing my Kamailio box but I am unable to get it upto 2000 concurrent calls it starts to fall over at 1300 have you got any ideas on how I can increase the performance of kamilio btw I am also using rtpengine Kind Regards -- CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED This email, and any attachments, is intended only for the above addressee. It may contain private and/or confidential information. If you have received this email in error you are on notice of its status, please immediately notify the sender by return email then delete this message and any attachments. If you are not the addressee, except to notify the sender, you must not use, disclose, copy or distribute this email and/or its attachments. Netcall Telecom accepts no responsibility for any changes made to this message after it has been sent by the original author. Opinions or views expressed in this email may be those of the individual sender and not Netcall Telecom. Nothing in this email shall bind Netcall Telecom in any contract or obligation Netcall Telecom Ltd Registered in England 2831215. Registered Office : 3rd Floor, Hamilton House, 111 Marlowes, Hemel Hempstead, Herts, HP1 1BB

5 7

Kamailio SIP 500
by Rodrigo Moreira 17 Aug '16

17 Aug '16

Hello, You can help me to due with the error: SIP 500 - Server error on UAR select next S-CSCF. What am I doing wrong? Thanks. Regards. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

3 3

t_set_fr() different behavior whether transport is udp or tcp
by Daniel Tryba 16 Aug '16

16 Aug '16

I'm seeing a different behavior of t_set_fr depending on transports. Scenario is that a endpoint has a failover defined in the registrat after 10s (t_set_fr(10000) and handling the locally generated 408 to the failover destination). This works fine when the request and response where delivered over UDP. When the Path is TCP the failover happends after 30s (even when using a different time t_set_fr(20000) so it is not a factor 3 or something like that). Setup: OK, 10s: Orig->UDP->loadbalancer->UDP->registrar->UDP->loadbalancer->TCP->Term Fail, 30s: Orig->UDP->loadbalancer->TCP->registrar->TCP->loadbalancer->TCP->Term Loadbalancer and registrar are kamailio machines (4.3.6). Communication between lb and registrar is based on dispatcher and path modules. 1 sip:registrar:5060;transport=udp 8 0 or via tcp: 1 sip:registrar:5060;transport=tcp 8 0 In the location database of the registrar the difference between the cases is: socket: udp:registrar:5060 or via tcp tcp:registrar:5060 path: <sip:lb@loadbalancer;lr;received=sip:1.2.3.4:5067%3Btransport%3Dtcp> or via tcp <sip:lb@loadbalancer;transport=tcp;lr;received=sip:1.2.3.4:5067%3Btransport%3Dtcp> Looking at debug(=3) nothing happens between the initial INVITE and the local 408 as far as I can see.

2 2

302 - stop sending once ACKed
by jaflong jaflong 16 Aug '16

16 Aug '16

I have kamailio as a redirect server. It responds with 302 to invites. The problem I am having is it continues to send the 302 multiple times. How can i get it to stop sending once it gets the first ACK

2 1

Is ACK required in a re-invite?
by Jay Li 15 Aug '16

15 Aug '16

Dear All, I just have a basic SIP question on re-invite. I wonder if "ACK" is required in a re-invite scenario, like re-invite -> 200 OK -> ACK. What I've seen is for some reason, my PSTN doesn't send a "ACK" back to me after my 200 OK response to their re-invite. I don't know if not sending an ACK in a re-invite is an allowed behavior. Thanks. Regards,Jay

2 3

kamailio TLS mod_register failed
by wanmon wanmon 15 Aug '16

15 Aug '16

4 14

Via in locally generated PUBLISH (using different presence server address)
by Asgaroth 15 Aug '16

15 Aug '16

Hi All, I have a configuration where we run a multi-homed proxy/loadbalancer and a seperate internal presence server. We then use pua/pua-dialoginfo on the proxies to publish dialog state to the internal presence server. The proxies listen on udp:external_ip:5062 and udp:internal_ip1:5060. The presence server listens on udp:internal_ip2:5060 on a seperate server. The problem I am coming accross is that when kamailio generates a PUBLISH message for some dialogs it creates the via header with address external:ip:5062, however the pua module is configured to send to the presence server address on internal_ip2:5060. I have a route which forces the sending socket ($fs) to be udp:internal_ip1:5060. What happens then is that the presence server tries to send the 200 reply to the PUBLISH message back to internal_ip1:5062 (when it should be internal_ip1:5060), I'm thinking this is because the Via header has host as external_ip:5062. Is it possible, from event-route[tm:local-request] to rewrite the via header so that it reflects internal_ip1:5060 instead of external_ip:5062? I did try the following snippet from event-route[tm:local-request], but I got module parse errors on the Via header: if (is_method("PUBLISH")) { subst_hf("Via", "/ 1\.2\.3\.[0-9]+:5062;/ $sel(cfg_get.socket.internal_ip);/", "f"); } [7370]: ERROR: <core> [parser/parse_via.c:2607]: parse_via(): ERROR:parse_via on <1> state 5 (default) [7370]: ERROR: <core> [parser/parse_via.c:2708]: parse_via(): ERROR: parse_via on: <SIP/2.0/UDP external_ip:5062;branch=z9hG4bK96eb.0f014ae4000000000000000000000000.0#015#012 internal_ip1; [7370]: ERROR: <core> [parser/parse_via.c:2712]: parse_via(): ERROR: parse_via parse error, parsed so far:<SIP/2.0/UDP external_ip:5062;branch=z9hG4bK96eb.0f014ae4000000000000000000000000.0#015#012 > [7370]: ERROR: <core> [parser/msg_parser.c:125]: get_hdr_field(): ERROR: get_hdr_field: bad via [7370]: INFO: <core> [parser/msg_parser.c:338]: parse_headers(): ERROR: bad header field [Via: SIP/2.0/UDP 1] [7370]: ERROR: <core> [parser/msg_parser.c:690]: parse_msg(): ERROR: parse_msg: message=<PUBLISH sip:user@domain;transport=UDP SIP/2.0#015#012Via: SIP/2.0/UDP external_ip:5062;branch=z9hG4bK96eb.0f014ae4000000000000000000000000.0#015#012 internal_ip1;To: <sip:user@domain;transport=UDP>#015#012From: <sip:user@domain;transport=UDP>;tag=3e8fdaca277026bf3e6d75c5e5313096-f162#015#012CSeq: 10 PUBLISH#015#012Call-ID: 5634294b786134b3-7370@external_ip#015#012Content-Length: 692#015#012User-Agent: kamailio (bfievklb04)#015#012Max-Forwards: 70#015#012Event: dialog#015#012Expires: 10901#015#012Content-Type: application/dialog-info+xml#015#012#015#012<?xml version="1.0"?>#012<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:user@domain;transport=UDP">#012 <dialog id="mvqXXaAr24DHdfsffYKZeA.." call-id="mvqXXaAr24DHdfsffYKZeA.." direction="initiator">#012 <state>Trying</state>#012 <remote>#012 <identity>sip:1234567@domain;transport=UDP</identity>#012 <target uri="sip:1234567@domain;transport=UDP"/>#012 </remote>#012 <local>#012 <identity>sip:user@domain;transport=UDP</identity>#012 <target uri="sip:user@domain;transport=UDP"/>#012 </local>#012 </dialog>#012</dialog-info>#012> [7370]: ERROR: <core> [msg_translator.c:3057]: build_sip_msg_from_buf(): parsing failed [7370]: ERROR: tm [uac.c:163]: uac_refresh_hdr_shortcuts(): failed to parse msg buffer [7370]: ERROR: tm [uac.c:487]: t_uac_prepare(): failed to refresh header shortcuts [7370]: ERROR: pua [send_publish.c:700]: send_publish(): in t_request tm module function [7370]: ERROR: pua_dialoginfo [dialog_publish.c:358]: dialog_publish(): while sending publish To me, it looks like the replace section of the subst_hf sub expression is being appended to the end of the via header instead of replacing the address section found. I have sanitized the above output to not show internal details, but if you need it, I can send it on to you privately. Am I going about this issue in the correct way, and, if so, where am I going wrong with the subst_hf snippet above? Kamailio version is 4.3.6. Any pointers/tips/trciks are greatly appreciated. Thanks

2 5

Installation question on ubuntu xenial / undefined symbol: EVP_sha1
by Christian Lox 15 Aug '16

15 Aug '16

Hi. I am trying to install kamailio 4.4.2 on ubuntu xenial, for use with kazoo. Using source and compiling just went fine, but /usr/local/sbin/kamailio -c throws ERROR: <core> [sr_module.c:582]: load_module(): could not open module </usr/local/lib64/kamailio/modules/outbound.so>: /usr/local/lib64/kamailio/modules/outbound.so: undefined symbol: EVP_sha1 any hint appreciated! Thanks, Christian

2 1

Re: [SR-Users] undefined symbol: EVP_sha1 when using outbound module
by Christian Lox 15 Aug '16

15 Aug '16

Hi. Same problem here; undefined symbol : EVP_sha1 in outbound.so pkg-config was available when compiled from source. Tried installing several dev packages and so on but always the same error. How do i link it against libssl/libcrypto? Thanks, Christian

2 2

access header variables in failure route
by jaflong jaflong 13 Aug '16

13 Aug '16

On an 302 message caught in failure_route Is it possible to get to access $hdr(xxx) of the 302. When I do this $hdr(xxx) it is null.

2 1

Struggling at 1000 Concurrent Calls
by Jack Stevens 12 Aug '16

12 Aug '16

Hi Guys, I have built an SBC using kamailio with the dispatcher module and rtpengine and when I run a sip test I find it struggles to go past 1000 calls at 20cps would really like some help on how to resolve this. As I am very new to this I am stuggeling to work out where to start from. I have found when running this sort of load the debian box drops pings any ideas Kind Regards -- CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED This email, and any attachments, is intended only for the above addressee. It may contain private and/or confidential information. If you have received this email in error you are on notice of its status, please immediately notify the sender by return email then delete this message and any attachments. If you are not the addressee, except to notify the sender, you must not use, disclose, copy or distribute this email and/or its attachments. Netcall Telecom accepts no responsibility for any changes made to this message after it has been sent by the original author. Opinions or views expressed in this email may be those of the individual sender and not Netcall Telecom. Nothing in this email shall bind Netcall Telecom in any contract or obligation Netcall Telecom Ltd Registered in England 2831215. Registered Office : 3rd Floor, Hamilton House, 111 Marlowes, Hemel Hempstead, Herts, HP1 1BB

3 3

CDP Error S-CSCF Kamailio
by Rodrigo Moreira 12 Aug '16

12 Aug '16

Hello Carsten Bock, I got to advance in the event REGISTER, but now face the following problem: When I type the command: */etc/init.d/kamailio restart* Aug 10 20:40:52 scscf /usr/sbin/kamailio[2639]: INFO: cdp [diameter_peer.c:428]: diameter_peer_destroy(): destroy_diameter_peer(): Terminating all children... Aug 10 20:40:52 scscf /usr/sbin/kamailio[2639]: INFO: cdp [diameter_peer.c:447]: diameter_peer_destroy(): destroy_diameter_peer(): All processes terminated....aning up. Aug 10 20:40:52 scscf /usr/sbin/kamailio[2639]: INFO: cdp [worker.c:140]: worker_destroy(): Unlocking workers waiting on empty queue... Aug 10 20:40:52 scscf /usr/sbin/kamailio[2639]: INFO: cdp [worker.c:143]: worker_destroy(): Unlocking workers waiting on full queue... Aug 10 20:40:52 scscf /usr/sbin/kamailio[2639]: CRITICAL: cdp [diameter_peer.c:487]: diameter_peer_destroy(): destroy_diameter_peer(): Bye Bye from C Diameter Peer test Aug 10 20:40:52 scscf /usr/sbin/kamailio[2639]: INFO: cdp [mod.c:244]: cdp_exit(): ... CDiameterPeer child stoped Aug 10 20:40:52 scscf /usr/sbin/kamailio[2639]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Aug 10 20:40:52 scscf systemd[1]: kamailio.service: control process exited, code=exited status=255 Aug 10 20:40:52 scscf systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. Aug 10 20:40:52 scscf systemd[1]: Unit kamailio.service entered failed state. I can not start the Kamailio, I used the examples codes available in the repository (GIT). You suggest something I may be doing wrong? Can you help me? I attached my config files. Thank you very much. Regards. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

4 4

Kamailio 4.4 Dispatcher on Debian 8 with MongoDB
by Jonathan Hunter 12 Aug '16

12 Aug '16

Hi Guys, I am integrating kamailio with MongoDB, and it works well with subscriber/location tables. However when using the dispatcher module I am getting the following error even though the IP Im trying to get to is routeable; /usr/local/kamailio-4.4/sbin/kamailio[2998]: ERROR: dispatcher [dispatch.c:1403]: ds_get_index(): destination set [1] not foundAug 11 17:01:47 ip-172-31-43-8 /usr/local/kamailio-4.4/sbin/kamailio[2998]: ERROR: dispatcher [dispatch.c:1832]: ds_select_dst_limit(): destination set [1] not found I created the table as below; db.getCollection("version").insert({table_name: "dispatcher", table_version: NumberInt(4) }) And the entry Im routing to is; db.dispatcher.find().pretty(){ "_id" : ObjectId("57acadc81e7e803fa565bce3"), "setid" : "1", "destination" : "sip:1.1.1.1:5080;transport=tcp", "flags" : "0", "priority" : "100", "attrs" : "name=labfS", "description" : "FreeSWITCH1"} Is there something I am missing ? Has anyone else got it working with Mongo? Thanks Jon

2 1

Kamailio CDP Error - SCSCF
by Rodrigo Moreira 12 Aug '16

12 Aug '16

Hello, I can not make the S-CSCF module work, is giving CDP error. CRITICAL: cdp [diameter_peer.c: 485]: diameter_peer_destroy (): destroy_diameter_peer (): Bye Bye from C Diameter Peer test I am using the examples available in the repository still occur errors that prevent Kamailio to boot. Can someone help me? Someone has the kamailio.cfg, scscf.cfg and scscf.xml files to share? Regards. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

1 0

Process redirected message
by jaflong jaflong 11 Aug '16

11 Aug '16

kamailio A relays to kamailio B kamailio B send 300 redirect back to kamailio A In this setup how do I catch the redirect response back from kamailio B What conditions should I do to catch and further process the redirect message Regards

2 2

Re: [SR-Users] kamailio starts looping invites(some) back to itself.
by SamyGo 10 Aug '16

10 Aug '16

Hi Aqs, How about putting a loop protection of some sort additionally like $si != myself. Is there any chance the API http call is getting skipped to relay w/o getting any reply back and hence looping back to itself ? Regards, Sammy On Jul 22, 2016 14:17, "Aqs Younas" <aqsyounas(a)gmail.com> wrote: Hi, I am using http_async_client and dispatcher modules of Kamailio. Here is scenario! http_async_client makes calls to an API to get groupid upon which dispatcher sends calls. I am using SIPp to generate calls with 100 cps. I see some calls being failed with an error cause, 483 Too Many Hops. Taking the trace shows Kamailio routing some calls to itself. I think this is not a configuration issue since all calls are same(follow same logic) and some calls get failed. Any pointer is much appreciated. Best Regards. _______________________________________________ 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

5 13

Problem receiving large payloads from JSONRPC-S
by Alex Balashov 10 Aug '16

10 Aug '16

Hi, I'm requesting 'dlg.list' from JSONRPC-S via XHTTP on a rather busy server, i.e. event_route[xhttp:request] { xlog("L_INFO", 'action=ER-XHTTP-REQUEST | ret=LOG_HTTP_REQ | data={"method": "$rm", "path": "$hu", "source": "$si:$sp"}'); # Digest challenge omitted. if($hu =~ "^/rpc") { xlog("L_INFO", 'action=ER-XHTTP-REQUEST | ret=JSON_RPC_REQUEST | data={"path": "$hu"}'); jsonrpc_dispatch(); } else { xhttp_reply("404", "Not Found", "text/html", ""); } } # curl --digest -v \ -X POST -H 'Content-Type: application/json' \ -H 'Call-ID: abc12' \ -d '{"jsonrpc": "2.0", "method": "dlg.list", "id": "0deadb33f"}' \ 'http://sipus_adm:raSheinmet4@10.0.62.74:5060/csrp_rpc/' and finding that when there's more than a hundred or so dialogs in play, Kamailio abruptly closes the connection in the middle of returning the output: ----- "tag": "AF5DA9B4-6D5", "contact": "sip:14045551212@xxx.xxx.xxx.xxx:5060", "cseq": "0", "route_set": "", "socket": "udp:xxx.xxx.xxx.xxx:5060" }, * transfer closed with 151674 bytes remaining to read * Closing connection #0 curl: (18) transfer closed with 151674 bytes remaining to read "profiles": ["total",[root@proxy kamailio]# ------ I can't seem to figure out what parameters regulate this. I'm using the following TCP settings: tcp_async=yes tcp_connect_timeout=5 tcp_crlf_ping=no tcp_keepalive=yes tcp_keepidle=30 tcp_linger2=10 tcp_max_connections=4096 tcp_send_timeout=3 tcp_accept_no_cl=yes tcp_wq_blk_size=65535 tcp_wq_max=20971520 But this doesn't seem to have had any effect. Any help appreciated! Thanks, -- 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 6

Duplicate ACC records in db when using db_cluster
by Jan Hazenberg 09 Aug '16

09 Aug '16

Hi, I have a issue with the db_cluster module. All works fine but when i enable ACC records to the database i get duplicate entry's in the acc table. I have 2 nodes enabled: # ----- db_cluster params ----- modparam("db_cluster", "connection", "con1=>mysql://kamailio:password@10.1.1.1/kamailio") modparam("db_cluster", "connection", "con2=>mysql://kamailio:password@10.1.1.2/kamailio") modparam("db_cluster", "cluster", "cls1=>con1=9s8p;con2=9s8p") When i remove one node from the pool the acc records look fine. Could this be a bug? Or should i bypass the db_cluster for accounting? I'm running kamailio 4.4.2. Thanks, Jan Hazenberg

3 5

When Not Found
by Nahum Nir 09 Aug '16

09 Aug '16

Hi List, When sending an invite and the calle not found or responding with an ACK I want my script do some costume work. When can I do it (using regular script)? Thanks, Nir

3 2

Kamailio Regiser
by Rodrigo Moreira 09 Aug '16

09 Aug '16

Hello everyone, I am deploying an IMS core, however I can not register the user (Bob). The entities are all working PCSCF, ICSCF SCSCF and HSS, I suppose. However after utiliizar Wireshark is returned the following error: 604 Does not exist anywhere - HSS User Unknow 400 Bad Request - P-visited-Network-ID header missing. What am I doing wrong? I ask for help, the great community. Or methods to better understand what is causing the problem. Is it DNS? Regards. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

2 1

open websocket connction from kamailio
by Uri Shacked 08 Aug '16

08 Aug '16

Hi, Is it possible to open a websocket connection from kamailio to another websocket server. not something based on sip, just a websocket to send to the websocket server messages from cfg file. (like http_query) ?? Uri

1 0

Feature Request CarrierRoute To and Contact header Strip
by Yasin CANER 08 Aug '16

08 Aug '16

2 1

app_python access to message body content
by Anthony Joseph Messina 08 Aug '16

08 Aug '16

I'm trying to play around a bit with app_python using the examples in the source. I'd like to access the content of the body of the SIP message, but can't figure out how to do that. I can get headers, method, etc. How does one access body content with app_python? Thanks. -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6

3 4

TCP message parsing & connection close issue
by Alex Balashov 08 Aug '16

08 Aug '16

Hello, I've got an odd issue where: 1. TCP client makes call out through Kamailio. 2. Call is answered, TCP client sends e2e ACK. 3. Parser chokes on ACK, closes TCP connection (SYN+FIN). 4. Subsequent messages from the client, immediately following the ACK (about 240 ms later), come through because the client is halfway around the world - about 250 ms away - and is not yet aware the connection has been closed. Otherwise, it would presumably try to re-establish the connection before sending anything. Anyway, messages from the parser concerning the ACK are: --- Aug 4 21:55:57 sip-proxy /usr/local/kamailio/sbin/kamailio[14680]: INFO: <core> [parser/parse_fline.c:144]: parse_first_line(): ERROR:parse_first_line: method not followed by SP Aug 4 21:55:57 sip-proxy /usr/local/kamailio/sbin/kamailio[14680]: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 0) Aug 4 21:55:57 sip-proxy /usr/local/kamailio/sbin/kamailio[14680]: ERROR: <core> [parser/msg_parser.c:690]: parse_msg(): ERROR: parse_msg: message=<0#015#012a=rtcp:4007 IN IP4 1.1.1.1:1071#015#012a=sendrecv#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:96 telephone-event/8000#015#012a=fmtp:96 0-16#015#012ACK sip:*98*0b3ff883-a89d-4317-b758-8979682f5357@10.0.2.53:5060;transport=udp SIP/2.0#015#012Via: SIP/2.0/TCP 1.1.1.1:61979;branch=z9hG4bKPjc15bf9fdded940cab7f859634b9f036e;alias#015#012Max-Forwards: 70#015#012From: sip:4b57be8d-267d-48c4-aa82-0f3a269d74cb@sip.evaristesys.com;tag=dd1d7fd0344d47b7b22c4017091df17a#015#012To: sip:*98*0b3ff883-a89d-4317-b758-8979682f5357@sip.evaristesys.com;tag=2rZDKcBFmU65S#015#012Call-ID: 5ba5a84451074159aaaa931eb1f98fe5#015#012CSeq: 3071 ACK#015#012Route: <sip:2.2.2.2;transport=tcp;lr;r2=on;ftag=dd1d7fd0344d47b7b22c4017091df17a;rtp_set_id=1>#015#012Route: <sip:10.0.2.119;lr;r2=on;ftag=dd1d7fd0344d47b7b22c4017091df17a;rtp_set_id=1>#015#012Content-Length: 0#015#012#015#012> Aug 4 21:55:57 sip-proxy /usr/local/kamailio/sbin/kamailio[14680]: ERROR: <core> [receive.c:173]: receive_msg(): core parsing of SIP message failed (1.1.1.1:61979/2) --- That doesn't look like a clean message buffer to me. The message should start here: ACK sip:*98*0b3ff883-a89d-4317-b758-8979682f5357@10.0.2.53:5060;transport=udp SIP/2.0#015#012Via: SIP/2.0/TCP ... But instead it's got these SDP attributes prepended: 0#015#012a=rtcp:4007 IN IP4 1.1.1.1:1071#015#012a=sendrecv#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:96 telephone-event/8000#015#012a=fmtp:96 0-16#015#012 These SDP attribute appear to have fallen off the back of the SDP offer in the initial INVITE: v=0 o=- 3679365582 3679365582 IN IP4 10.0.0.200 s=pjmedia t=0 0 m=audio 4006 RTP/AVP 0 8 96 c=IN IP4 1.1.1.1 The relationship between the 4006 RTP port and the 4007 RTCP port suggests that these two things belong together. This is on kamailio 4.4.0 (x86_64/linux) d4f23c. Any help would be greatly appreciated! The issue is consistently reproducible and this happens every time. Cheers, -- 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 6

Synonyms User Name
by Nahum Nir 07 Aug '16

07 Aug '16

Hello List, I wonder if there is a way to define synonyms. For example, I there is a person by the name of John Smith with extention 5555 I would like to have a user john_smith that will also be available if some INVITE 5555. Thanks, Nahum

2 3

Prepaid system using dialog and sqlops
by Infinicalls Infinicalls 06 Aug '16

06 Aug '16

Hi, I am still struggling to get a prepaid system for my service. After installing SIREMIS, I am able to get the CDRs updated. But even after adding the cron settings, the call duration doesn't get updated. And also in http://lists.sip-router.org/pipermail/sr-users/2012-January/071517.html it has been mentioned that "If it is the case of 1 call per user at a time, then practically can be done from kamailio config only. When call starts, compute the max duration for that call based on caller/destination (e.g., call a stored procedure via sqlops) and set the dialog timeout to that duration. When call ends (BYE in main route or timeout route) update the credit accordingly." Can somebody let me know if my understanding is correct. 1) Add a field called "credits" in the database. And topup that field. 2) First compute max permitted duration for that user by multiplying the cost for number prefix (say +91) and pulse. 3) And finally add this value to timeout of dialog. modparam("dialog", "default_timeout", 100000) $dlg_ctx(timeout_bye) = 1; Will this work? Any help would be appreciated. regards Ganesh Kumar

3 4

segfault when using sipt modifications
by david 05 Aug '16

05 Aug '16

hello all i'm trying to add a ISUP multipart in a regular call using the textops module, and despite seeing one previous issue int he mail list, i'm finding some issue here i'm using this commands msg_apply_changes(); set_body_multipart(,"2123894789_1257887457"); msg_apply_changes(); append_body_part("\x01\x12\x49\x00\x0a\x03\x02\x0a\x08\x84\x90\x33 \x41\x72\x17\x00\x06\x0a\x08\x04\x13\x93\x70\x21\x73\x23\x10\x08\x01\x00 \x00","application/isup;version=itu-t92+","signal;handling=optional"); msg_apply_changes(); sipt_destination($rU, 31, 4); sipt_set_calling($fU, 4, 0, 3); msg_apply_changes(); before the record_route command, and when using sipt_destination($rU, 31, 4); i'm getting asegfault here you have the logs could you please give me any clue about something i'm missing or doing wrong? thanks alot and regards david 5(24552) DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): SIP Request: 5(24552) DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): method: <INVITE> 5(24552) DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): uri: <sip:34666988297@79.170.71.132:5060> 5(24552) DEBUG: <core> [parser/msg_parser.c:614]: parse_msg(): version: <SIP/2.0> 5(24552) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 235, <rport> = <n/a>; state=6 5(24552) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK7ma5t4FKS0SKa>; state=16 5(24552) DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 5(24552) DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 5(24552) DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via 5(24552) DEBUG: <core> [receive.c:178]: receive_msg(): After parse_msg... 5(24552) DEBUG: <core> [receive.c:221]: receive_msg(): preparing to run routing scripts... 5(24552) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 65 5(24552) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=10 5(24552) DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [49]; uri=[sip:34666988297@79.170.71.132:5060;user=phone] 5(24552) DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [<sip:34666988297@79.170.71.132:5060;user=phone> ] 5(24552) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <94568362> <INVITE> 5(24552) DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=104 5(24552) DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header 5(24552) DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=5jBBFae4D90rg 5(24552) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 5(24552) DEBUG: sanity [mod_sanity.c:248]: w_sanity_check(): sanity checks result: 1 5(24552) DEBUG: siputils [checks.c:97]: has_totag(): no totag 5(24552) DEBUG: <core> [socket_info.c:564]: grep_sock_info(): checking if host==us: 13==13 && [79.170.71.132] == [79.170.71.132] 5(24552) DEBUG: <core> [socket_info.c:567]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 5(24552) DEBUG: tm [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=1 global id=0 T start=0xffffffffffffffff 5(24552) DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=46859, isACK=0 5(24552) DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed 5(24552) DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found 5(24552) DEBUG: tm [t_lookup.c:1080]: t_check_msg(): DEBUG: t_check_msg: msg id=1 global id=1 T end=(nil) 5(24552) DEBUG: tm [t_lookup.c:1312]: t_newtran(): DEBUG: t_newtran: msg id=1 , global msg id=1 , T on entrance=(nil) 5(24552) DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=46859, isACK=0 5(24552) DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed 5(24552) DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found 5(24552) DEBUG: tm [t_hooks.c:358]: run_reqin_callbacks_internal(): DBG: trans=0x7fb3fb0c64e0, callback type 1, id 0 entered 5(24552) DEBUG: <core> [md5utils.c:67]: MD5StringArray(): MD5 calculated: 5ea9736b3bdcf90a4c3043a5a492b31f 5(24552) DEBUG: <core> [select.c:412]: run_select(): Calling SELECT 0x7fb404252ef8 5(24552) INFO: Theseus-Test Call 69672589-67f5-4edd-8844-f6ad2ee5b208 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: Call-ID = 69672589-67f5-4edd-8844-f6ad2ee5b208 5(24552) DEBUG: app_lua [app_lua_api.c:643]: app_lua_run(): reload deactivated 5(24552) DEBUG: app_lua [app_lua_api.c:644]: app_lua_run(): executing Lua function: [[allow_src_uri]] 5(24552) DEBUG: app_lua [app_lua_api.c:645]: app_lua_run(): lua top index is: 4 5(24552) DEBUG: app_lua [app_lua_sr.c:979]: lua_sr_pv_get(): pv get: $avp(s:call_obj) 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$avp(s:call_obj)] found in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:979]: lua_sr_pv_get(): pv get: $ci 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$ci] found in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:1101]: lua_sr_pv_sets(): pv set: $var(authorized) 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$var(authorized)] found in cache 5(24552) INFO: Theseus-Test Call 69672589-67f5-4edd-8844-f6ad2ee5b208 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: Method INVITE from 79.170.64.130:5080 authorized 5(24552) INFO: Theseus-Test Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: Object 58825 assigned to Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208 5(24552) INFO: Theseus-Test Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: The call has been sent to RG 106406 5(24552) DEBUG: app_lua [app_lua_api.c:643]: app_lua_run(): reload deactivated 5(24552) DEBUG: app_lua [app_lua_api.c:644]: app_lua_run(): executing Lua function: [[get_rg_info]] 5(24552) DEBUG: app_lua [app_lua_api.c:645]: app_lua_run(): lua top index is: 4 5(24552) DEBUG: app_lua [app_lua_sr.c:979]: lua_sr_pv_get(): pv get: $ci 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$ci] found in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:979]: lua_sr_pv_get(): pv get: $avp(s:call_obj) 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$avp(s:call_obj)] found in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:979]: lua_sr_pv_get(): pv get: $shv(flag_list) 5(24552) DEBUG: <core> [pvapi.c:293]: pv_cache_add(): pvar [$shv(flag_list)] added in cache 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG 106406 attributes: 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG OUT 106406 description = PBX--NOC-TEST 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG OUT 106406 admin_status = 1 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG OUT 106406 cir_direction = 2 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG OUT 106406 ani_posttranslation = 0 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG OUT 106406 ani_posttranslation_table = 0 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG OUT 106406 posttranslation = 1 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG OUT 106406 posttranslation_table = 106384 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG OUT 106406 pound = 2 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG OUT 106406 gw_port = 5060 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: RG OUT 106406 custom_flags = None 5(24552) DEBUG: app_lua [app_lua_sr.c:1101]: lua_sr_pv_sets(): pv set: $avp(s:description) 5(24552) DEBUG: <core> [usr_avp.c:882]: parse_avp_ident(): Parsing 's:description' 5(24552) DEBUG: <core> [pvapi.c:293]: pv_cache_add(): pvar [$avp(s:description)] added in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:1101]: lua_sr_pv_sets(): pv set: $avp(s:admin_status) 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$avp(s:admin_status)] found in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:1101]: lua_sr_pv_sets(): pv set: $avp(s:cir_direction) 5(24552) DEBUG: <core> [usr_avp.c:882]: parse_avp_ident(): Parsing 's:cir_direction' 5(24552) DEBUG: <core> [pvapi.c:293]: pv_cache_add(): pvar [$avp(s:cir_direction)] added in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:1101]: lua_sr_pv_sets(): pv set: $avp(s:ani_posttranslation_table) 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$avp(s:ani_posttranslation_table)] found in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:1101]: lua_sr_pv_sets(): pv set: $avp(s:posttranslation_table) 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$avp(s:posttranslation_table)] found in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:1101]: lua_sr_pv_sets(): pv set: $avp(s:pound) 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$avp(s:pound)] found in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:1101]: lua_sr_pv_sets(): pv set: $avp(s:gw_ip) 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$avp(s:gw_ip)] found in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:1101]: lua_sr_pv_sets(): pv set: $avp(s:gw_port) 5(24552) DEBUG: <core> [pvapi.c:321]: pv_cache_lookup(): pvar [$avp(s:gw_port)] found in cache 5(24552) INFO: app_lua [app_lua_sr.c:96]: lua_sr_log(): Call 58825 / Call-ID 69672589-67f5-4edd-8844-f6ad2ee5b208: Custom flags = None 5(24552) DEBUG: app_lua [app_lua_sr.c:979]: lua_sr_pv_get(): pv get: $shv(custom_flag_list) 5(24552) DEBUG: <core> [pvapi.c:293]: pv_cache_add(): pvar [$shv(custom_flag_list)] added in cache 5(24552) DEBUG: app_lua [app_lua_sr.c:979]: lua_sr_pv_get(): pv get: $shv(custom_flag_list_branch) 5(24552) DEBUG: <core> [pvapi.c:293]: pv_cache_add(): pvar [$shv(custom_flag_list_branch)] added in cache 5(24552) DEBUG: textopsx [textopsx.c:231]: msg_apply_changes_f(): SIP message content updated - reparsing 5(24552) DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): SIP Request: 5(24552) DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): method: <INVITE> 5(24552) DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): uri: <sip:34666988297@79.170.71.132:5060> 5(24552) DEBUG: <core> [parser/msg_parser.c:614]: parse_msg(): version: <SIP/2.0> 5(24552) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 235, <rport> = <n/a>; state=6 5(24552) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK7ma5t4FKS0SKa>; state=16 5(24552) DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 5(24552) DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 5(24552) DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via 5(24552) DEBUG: textops [textops.c:1577]: set_multibody_helper(): delimiter<21>:[2123894789_1257887457] 5(24552) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=10 5(24552) DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [49]; uri=[sip:34666988297@79.170.71.132:5060;user=phone] 5(24552) DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [<sip:34666988297@79.170.71.132:5060;user=phone> ] 5(24552) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <94568362> <INVITE> 5(24552) DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=104 5(24552) DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header 5(24552) DEBUG: textops [textops.c:1486]: generate_boundary(): adding final CRLF+CRLF 5(24552) DEBUG: textops [textops.c:1714]: set_multibody_helper(): content-type<48>:[multipart/mixed;boundary="2123894789_1257887457"] 5(24552) DEBUG: textops [textops.c:1768]: set_multibody_helper(): set flag FL_BODY_MULTIPART 5(24552) INFO: <core> [msg_translator.c:1692]: get_boundary(): Content-Type hdr has no params <application/sdp> 5(24552) WARNING: <core> [msg_translator.c:1958]: build_req_buf_from_sip_req(): check_boundaries error 5(24552) DEBUG: <core> [msg_translator.c:422]: clen_builder(): content-length: 189 (189) 5(24552) DEBUG: textopsx [textopsx.c:231]: msg_apply_changes_f(): SIP message content updated - reparsing 5(24552) DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): SIP Request: 5(24552) DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): method: <INVITE> 5(24552) DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): uri: <sip:34666988297@79.170.71.132:5060> 5(24552) DEBUG: <core> [parser/msg_parser.c:614]: parse_msg(): version: <SIP/2.0> 5(24552) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 235, <rport> = <n/a>; state=6 5(24552) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK7ma5t4FKS0SKa>; state=16 5(24552) DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 5(24552) DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 5(24552) DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via 5(24552) ERROR: <core> [lvalue.c:345]: lval_pvar_assign(): non existing right pvar 5(24552) ERROR: <core> [lvalue.c:405]: lval_assign(): assignment failed at pos: (390,18-390,44) 5(24552) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=10 5(24552) DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [49]; uri=[sip:34666988297@79.170.71.132:5060;user=phone] 5(24552) DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [<sip:34666988297@79.170.71.132:5060;user=phone> ] 5(24552) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <94568362> <INVITE> 5(24552) DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=189 5(24552) DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header 5(24552) DEBUG: textops [textops.c:1486]: generate_boundary(): adding final CRLF+CRLF 5(24552) DEBUG: <core> [msg_translator.c:1719]: get_boundary(): boundary is <--2123894789_1257887457> 5(24552) DEBUG: <core> [msg_translator.c:1839]: check_boundaries(): last bondary without -- at the end 5(24552) DEBUG: <core> [msg_translator.c:1639]: replace_body(): old size body[189] actual[319] 5(24552) DEBUG: <core> [msg_translator.c:422]: clen_builder(): content-length: 319 (319) 5(24552) DEBUG: textopsx [textopsx.c:231]: msg_apply_changes_f(): SIP message content updated - reparsing 5(24552) DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): SIP Request: 5(24552) DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): method: <INVITE> 5(24552) DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): uri: <sip:34666988297@79.170.71.132:5060> 5(24552) DEBUG: <core> [parser/msg_parser.c:614]: parse_msg(): version: <SIP/2.0> 5(24552) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 235, <rport> = <n/a>; state=6 5(24552) DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK7ma5t4FKS0SKa>; state=16 5(24552) DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 5(24552) DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 5(24552) DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via 5(24552) DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=10 5(24552) DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [49]; uri=[sip:34666988297@79.170.71.132:5060;user=phone] 5(24552) DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [<sip:34666988297@79.170.71.132:5060;user=phone> ] 5(24552) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <94568362> <INVITE> 5(24552) DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=319 5(24552) DEBUG: <core> [parser/parse_body.c:87]: get_boundary_param(): boundary is "2123894789_1257887457" 5(24552) DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header 5(24552) DEBUG: <core> [parser/parse_body.c:87]: get_boundary_param(): boundary is "2123894789_1257887457" 5(24552) CRITICAL: <core> [data_lump.c:297]: del_lump(): offset + len exceeds message size (1176 + 51 > 1201) 35(24582) CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 17 35(24582) DEBUG: <core> [tcp_main.c:3448]: handle_ser_child(): dead child 5, pid 24552 (shutting down?) 35(24582) DEBUG: <core> [io_wait.h:598]: io_watch_del(): DBG: io_watch_del (0xa2ae40, 17, -1, 0x0) fd_no=39 called 0(24547) ALERT: <core> [main.c:739]: handle_sigs(): child process 24552 exited by a signal 6 0(24547) ALERT: <core> [main.c:742]: handle_sigs(): core was not generated 0(24547) INFO: <core> [main.c:754]: handle_sigs(): terminating due to SIGCHLD 0(24547) DEBUG: <core> [main.c:756]: handle_sigs(): terminating due to SIGCHLD 34(24581) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 33(24580) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 32(24579) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 31(24578) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 30(24577) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 29(24576) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 28(24575) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 27(24574) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 26(24573) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 25(24572) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 24(24571) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 23(24570) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 22(24569) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 21(24568) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 20(24567) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 19(24566) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 18(24565) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 17(24564) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 16(24563) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 15(24562) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 14(24561) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 13(24560) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 12(24559) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 11(24558) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 10(24557) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 9(24556) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 8(24555) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 7(24554) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 6(24553) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 4(24551) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 3(24550) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 2(24549) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 1(24548) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 35(24582) INFO: <core> [main.c:809]: sig_usr(): signal 15 received 0(24547) DEBUG: rtpengine [rtpengine.c:2014]: mod_destroy(): rtpengine_hash_table_destroy() success! 0(24547) DEBUG: db_text [dbtext.c:110]: destroy(): destroy ... 0(24547) DEBUG: tm [t_funcs.c:86]: tm_shutdown(): DEBUG: tm_shutdown : start 0(24547) DEBUG: tm [t_funcs.c:89]: tm_shutdown(): DEBUG: tm_shutdown : emptying hash table 0(24547) DEBUG: tm [h_table.c:127]: free_cell_helper(): freeing transaction 0x7fb3fb0c64e0 from h_table.c:449 0(24547) DEBUG: tm [t_funcs.c:91]: tm_shutdown(): DEBUG: tm_shutdown : removing semaphores 0(24547) DEBUG: tm [t_funcs.c:93]: tm_shutdown(): DEBUG: tm_shutdown : destroying tmcb lists 0(24547) DEBUG: tm [t_funcs.c:96]: tm_shutdown(): DEBUG: tm_shutdown : done 0(24547) INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized 0(24547) DEBUG: <core> [mem/shm.c:174]: shm_core_lock_destroy(): destroying the shared memory lock 0(24547) DEBUG: <core> [mem/shm.c:266]: shm_destroy_manager(): destroying memory manager: f_malloc 0(24547) DEBUG: <core> [mem/pkg.c:89]: pkg_destroy_manager(): destroying memory manager: f_malloc

3 13

e2e ACK. Kamailio. Cantact. Route.
by pavel＠eremina.net 05 Aug '16

05 Aug '16

Hello, I have this scheme: SIP Carrier (external IP 22.203) < > (local ip 1.21) mikrotik < > (local ip 1.21) Kamailio - (local ip 1.15) Asterisk. Carrier does not know what is Route header. that is why Carrier send ACK directly to 1.15 and fail. I change Contact at Kamailio to 1.21 then ACK is coming to Kamailio. But i think it's wrong way... How can i get ACK from CArrier to Kamalio and route it in right way? Thanks

1 0

e2e ACK
by Pavel Eremin 05 Aug '16

05 Aug '16

Hello, I have this scheme: SIP Carrier (external IP 22.203) < > (local ip 1.21) mikrotik < > (local ip 1.21) Kamailio - (local ip 1.15) Asterisk. Carrier does not know what is Route header. that is why Carrier send ACK directly to 1.15 and fail. I change Contact at Kamailio to 1.21 then ACK is coming to Kamailio. But i think it's wrong way... How can i get ACK from CArrier to Kamalio and route it in right way? Thanks

1 0

Re: [SR-Users] Textops and Multipart Body - adding ISUP segfaults Kamailio
by david 05 Aug '16

05 Aug '16

Hello all i'm having the same error on kam 4.4 when doing set_body_multipart(); msg_apply_changes(); i see in the logs 5(17797) DEBUG: textops [textops.c:1577]: set_multibody_helper(): delimiter<17>:[unique-boundary-1] 5(17797) DEBUG: textops [textops.c:1486]: generate_boundary(): adding final CRLF+CRLF 5(17797) DEBUG: textops [textops.c:1714]: set_multibody_helper(): content-type<44>:[multipart/mixed;boundary="unique-boundary-1"] 5(17797) DEBUG: textops [textops.c:1768]: set_multibody_helper(): set flag FL_BODY_MULTIPART 5(17797) INFO: <core> [msg_translator.c:1692]: get_boundary(): Content-Type hdr has no params <application/sdp> 5(17797) WARNING: <core> [msg_translator.c:1958]: build_req_buf_from_sip_req(): check_boundaries error 5(17797) DEBUG: <core> [msg_translator.c:422]: clen_builder(): content-length: 191 (191) the original INVITE is like .... Content-Type: application/sdp. Content-Disposition: session. Content-Length: 104. P-Asserted-Identity: "10707334" <sip:10707334@3.3.3.3>. . v=0. o=user1 53655765 23536 IN IP4 79.170.68.171. s=-. c=IN IP4 2.2.2.2. t=0 0. m=audio 6001 RTP/AVP 8. .... do you have any idea why these messages appear thanks a lot and regards david

2 1

Re: [SR-Users] Kamailio on multiple interface
by Daniel-Constantin Mierla 05 Aug '16

05 Aug '16

Hello, if you know the ip address associated with the network interfaces, you can use keyword dst_ip to check which of the ip addresses was used to receive the packet (or use $Ri instead of dst_ip). Cheers, Daniel On 05/08/16 08:41, Marino Mileti wrote: > > Hi guys, > > > > I've my Kamailio box that works great J > > > > Now I would like to enable wifi on the board (in AP mode) in order to > allow mobile devices to connect and receive and place calls. > > Is it possible? Is it possible to configure Kama like a "gateway" > between two network cards? > > > > I've find a guide regarding gateway between IPv4 and IPV6 but i can't > use "af" to discriminate message between eth0 or wlan0. Is there a > command to detect if a message is received from a specific interface card? > > > > Any help or any cfg example is very appreciate J > > > > /Marino Maria Mileti/// > > /marino.mileti(a)alice.it <mailto:marino.mileti@alice.it>/ > > / / > > /cid:006a01cb6b0e$67eecdae$_CDOSYS2.0//Reduce your energy consumption > and keep polar bears on ice!/ > > > > > > ------------------------------------------------------------------------ > Avast logo > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > > > > _______________________________________________ > 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 -- Daniel-Constantin Mierla http://www.asipto.com - http://www.kamailio.org http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Kamailio on multiple interface
by Marino Mileti 05 Aug '16

05 Aug '16

Hi guys, I've my Kamailio box that works great J Now I would like to enable wifi on the board (in AP mode) in order to allow mobile devices to connect and receive and place calls. Is it possible? Is it possible to configure Kama like a "gateway" between two network cards? I've find a guide regarding gateway between IPv4 and IPV6 but i can't use "af" to discriminate message between eth0 or wlan0. Is there a command to detect if a message is received from a specific interface card? Any help or any cfg example is very appreciate J Marino Maria Mileti <mailto:marino.mileti@alice.it> marino.mileti(a)alice.it cid:006a01cb6b0e$67eecdae$_CDOSYS2.0Reduce your energy consumption and keep polar bears on ice! --- Questa e-mail è stata controllata per individuare virus con Avast antivirus. https://www.avast.com/antivirus

1 0

Kamailio RTP Engine
by Jack Stevens 05 Aug '16

05 Aug '16

Hi Guys, Looking for some help on configuring RTP Engine with Dispatcher on my Kamailio Box Please see this paste for the kamailio configuration = http://pastebin.com/yZzSEJmK When I run a call the dispatcher does it job perfectly but the rtpengine is not been used does anyone know why -- CONFIDENTIAL EMAIL FROM NETCALL TELECOM LIMITED This email, and any attachments, is intended only for the above addressee. It may contain private and/or confidential information. If you have received this email in error you are on notice of its status, please immediately notify the sender by return email then delete this message and any attachments. If you are not the addressee, except to notify the sender, you must not use, disclose, copy or distribute this email and/or its attachments. Netcall Telecom accepts no responsibility for any changes made to this message after it has been sent by the original author. Opinions or views expressed in this email may be those of the individual sender and not Netcall Telecom. Nothing in this email shall bind Netcall Telecom in any contract or obligation Netcall Telecom Ltd Registered in England 2831215. Registered Office : 3rd Floor, Hamilton House, 111 Marlowes, Hemel Hempstead, Herts, HP1 1BB

2 1

Kamailio 4.4 wont start on CentOS 7 using script
by Fabian Borot 04 Aug '16

04 Aug '16

Hello I installed/compiled version 4.4 on CentOS 7. When I start the server like this: "./kamctl start" it works but when I try to use the script: "/etc/init.d/kamailio" it fails [root@ar-mia-proxy-01 sbin]# /etc/init.d/kamailio start Starting kamailio (via systemctl): Job for kamailio.service failed because the control process exited with error code. See "systemctl status kamailio.service" and "journalctl -xe" for details. [FAILED] [root@ar-mia-proxy-01 sbin]# when running the "status" I see this: [root@ar-mia-proxy-01 sbin]# /etc/init.d/kamailio status â kamailio.service - LSB: Kamailio (OpenSER) - the Open Source SIP Server Loaded: loaded (/etc/rc.d/init.d/kamailio) Active: failed (Result: exit-code) since Thu 2016-08-04 15:25:59 GMT; 14s ago Docs: man:systemd-sysv-generator(8) Process: 5556 ExecStart=/etc/rc.d/init.d/kamailio start (code=exited, status=1/FAILURE) Aug 04 15:25:59 ar-mia-proxy-01.dom.com systemd[1]: Starting LSB: Kamailio (OpenSER) - the Open Source SIP Server... Aug 04 15:25:59 ar-mia-proxy-01.dom.com kamailio[5556]: Starting kamailio: Aug 04 15:25:59 ar-mia-proxy-01.dom.com systemd[1]: kamailio.service: control process exited, code=exited status=1 Aug 04 15:25:59 ar-mia-proxy-01.dom.com systemd[1]: Failed to start LSB: Kamailio (OpenSER) - the Open Source SIP Server. Aug 04 15:25:59 ar-mia-proxy-01.dom.com systemd[1]: Unit kamailio.service entered failed state. Aug 04 15:25:59 ar-mia-proxy-01.dom.com systemd[1]: kamailio.service failed. [root@ar-mia-proxy-01 sbin]# I welcome all ideas, thank you in advance

3 4

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