- sr-users - kamailio.org

Re: [SR-Users] NAT Contact:
by Slava Bendersky 29 Mar '15

29 Mar '15

Hello Alex, This is debug might be helpful. Mar 29 12:24:05 canlvprx01 kamailio: 11(31941) DEBUG: <core> [receive.c:154]: receive_msg(): After parse_msg... Mar 29 12:24:05 canlvprx01 kamailio: 11(31941) DEBUG: <core> [receive.c:197]: receive_msg(): preparing to run routing scripts... Mar 29 12:24:05 canlvprx01 kamailio: 11(31941) DEBUG: sl [sl_funcs.c:412]: sl_filter_ACK(): DEBUG : sl_filter_ACK: to late to be a local ACK! Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, March 29, 2015 11:56:58 AM Subject: Re: [SR-Users] NAT Contact: Hello Alex, I switched to aliases method, but in case like this, AU ----> Internet ---> Kamailio ----> Asterisk AU2 <--Internet <---------| <------------| Call between two extensions not working, because kamailio mark as external domain and LOCATION not saved, but even if I save location I see in kamctl ul show more then 5 records for each AOR. Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, March 29, 2015 9:52:18 AM Subject: Re: [SR-Users] NAT Contact: Hello Alex, I switched to alias method, but there some CRITICAL message. Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=905 a=16 n=if Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=901 a=24 n=is_first_hop Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) DEBUG: siputils [sipops.c:161]: w_is_first_hop(): no 2nd via found - first hop Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) CRITICAL: <core> [route.c:1237]: comp_num(): Invalid right operand (1) Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=902 a=5 n=route Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) ERROR: *** cfgtrace:request_route=[ADD_CONTACT_ALIAS] c=[/etc/kamailio/kamailio-asterisk.cfg] l=976 a=16 n=if Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) ERROR: *** cfgtrace:request_route=[ADD_CONTACT_ALIAS] c=[/etc/kamailio/kamailio-asterisk.cfg] l=971 a=24 n=add_contact_alias Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) DEBUG: nathelper [nathelper.c:970]: add_contact_alias_0_f(): no need to add alias param From: "Alex Balashov" <abalashov(a)evaristesys.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, March 28, 2015 7:31:54 PM Subject: Re: [SR-Users] NAT Contact: On 03/28/2015 07:26 PM, Slava Bendersky wrote: > Is for mix of tcp transport and alias is required add any additional > parameters in global configuration ? I wouldn't think so. -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] NAT Contact:
by Slava Bendersky 29 Mar '15

29 Mar '15

Hello Alex, I switched to aliases method, but in case like this, AU ----> Internet ---> Kamailio ----> Asterisk AU2 <--Internet <---------| <------------| Call between two extensions not working, because kamailio mark as external domain and LOCATION not saved, but even if I save location I see in kamctl ul show more then 5 records for each AOR. Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, March 29, 2015 9:52:18 AM Subject: Re: [SR-Users] NAT Contact: Hello Alex, I switched to alias method, but there some CRITICAL message. Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=905 a=16 n=if Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=901 a=24 n=is_first_hop Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) DEBUG: siputils [sipops.c:161]: w_is_first_hop(): no 2nd via found - first hop Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) CRITICAL: <core> [route.c:1237]: comp_num(): Invalid right operand (1) Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=902 a=5 n=route Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) ERROR: *** cfgtrace:request_route=[ADD_CONTACT_ALIAS] c=[/etc/kamailio/kamailio-asterisk.cfg] l=976 a=16 n=if Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) ERROR: *** cfgtrace:request_route=[ADD_CONTACT_ALIAS] c=[/etc/kamailio/kamailio-asterisk.cfg] l=971 a=24 n=add_contact_alias Mar 29 09:54:45 canlvprx01 kamailio: 2(31248) DEBUG: nathelper [nathelper.c:970]: add_contact_alias_0_f(): no need to add alias param From: "Alex Balashov" <abalashov(a)evaristesys.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, March 28, 2015 7:31:54 PM Subject: Re: [SR-Users] NAT Contact: On 03/28/2015 07:26 PM, Slava Bendersky wrote: > Is for mix of tcp transport and alias is required add any additional > parameters in global configuration ? I wouldn't think so. -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ _______________________________________________ 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

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Re: [SR-Users] NAT Contact:
by Slava Bendersky 28 Mar '15

28 Mar '15

Hello Alex, Is for mix of tcp transport and alias is required add any additional parameters in global configuration ? Slava. From: "Alex Balashov" <abalashov(a)evaristesys.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, March 28, 2015 5:26:16 PM Subject: Re: [SR-Users] NAT Contact: Hello Slava, On 03/28/2015 02:26 PM, Slava Bendersky wrote: > How is possible add to Contact: received=ip ? Is this should be > handled by fix_nated_contact() This is not the function of fix_nated_contact(). Have a look at a add_rcv_param(): http://kamailio.org/docs/modules/4.2.x/modules/nathelper.html#nathelper.f.a… Someone might correct me if I'm wrong, but I believe the problem with the ;received approach is that it's only useful for registrations, since only the registrar module has a provision that allows it to be used automatically. So, for larger applications outside of registration, you should take a look at the new add_contact_alias() / handle_ruri_alias() facility: http://kamailio.org/docs/modules/4.2.x/modules/nathelper.html#nathelper.f.a… http://kamailio.org/docs/modules/4.2.x/modules/nathelper.html#nathelper.f.h… -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ _______________________________________________ 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

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NAT Contact:
by Slava Bendersky 28 Mar '15

28 Mar '15

Hello Everyone, How is possible add to Contact: received=ip ? Is this should be handled by fix_nated_contact() ? I am using default NATDETECT route, but I see in debug fix_nated_contact() executed, but in SDP Contact: stay Contact: <sip:102@private_ip:5065;transport=tcp>;audio. >From public side client use TCP inbound to kamailio and on local side to asterisk is use UDP. Kamailio have 2 interfaces one public one private. Any help thank you. Mar 28 14:28:28 canlvprx01 kamailio: 14(28868) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=895 a=55 n=force_rport Mar 28 14:28:28 canlvprx01 kamailio: 14(28868) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=905 a=16 n=if Mar 28 14:28:28 canlvprx01 kamailio: 14(28868) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=896 a=25 n=nat_uac_test Mar 28 14:28:28 canlvprx01 kamailio: 14(28868) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=901 a=16 n=if Mar 28 14:28:28 canlvprx01 kamailio: 14(28868) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=897 a=25 n=is_method Mar 28 14:28:28 canlvprx01 kamailio: 14(28868) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=900 a=24 n=fix_nated_contact Mar 28 14:28:28 canlvprx01 kamailio: 14(28868) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=902 a=39 n=setflag Mar 28 14:28:28 canlvprx01 kamailio: 14(28868) ERROR: *** cfgtrace:request_route=[NATDETECT] c=[/etc/kamailio/kamailio-asterisk.cfg] l=905 a=2 n=return

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DNS resolver.
by Slava Bendersky 28 Mar '15

28 Mar '15

Hello Everyone, How I can disable DNS resolution ? Currently I have in configuration file, but I still see in log DNS SRV resolution and forward. dns_cache_init=off use_dns_cache=off use_dns_failover=off dns=no rev_dns=no Mar 28 13:30:29 canlvprx01 kamailio: 12(28631) DEBUG: <core> [resolve.c:1239]: srv_sip_resolvehost(): srv_sip_resolvehost: domain.org:0 proto=0 Mar 28 13:30:29 canlvprx01 kamailio: 12(28631) DEBUG: <core> [resolve.c:1329]: srv_sip_resolvehost(): srv_sip_resolvehost: SRV(_sip._udp.domain.org) = canlpbx02.domain.org:5060 Mar 28 13:30:29 canlvprx01 kamailio: 12(28631) DEBUG: <core> [resolve.c:1360]: srv_sip_resolvehost(): srv_sip_resolvehost: returning 0x7f2bf81f9680 (domain.org:5060 proto=1) Mar 28 13:30:29 canlvprx01 kamailio: 12(28631) DEBUG: <core> [forward.c:211]: get_out_socket(): DEBUG: get_out_socket: socket determined: 0x7f2bf765e2d8 Mar 28 13:30:29 canlvprx01 kamailio: 12(28631) DEBUG: <core> [msg_translator.c:2893]: create_via_hf(): create_via_hf: id added: <;i=2>, rcv proto=2 Mar 28 13:30:29 canlvprx01 kamailio: 12(28631) DEBUG: tm [t_funcs.c:394]: t_relay_to(): SER: new transaction fwd'ed

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Asking for help in configuration Kamailio
by Mikael Sarkisyan 28 Mar '15

28 Mar '15

Hello Dear Sip Router Team I’m asking you to help me please with kamailio configuration I’ve made some laboratory system following this document http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb <http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb> and it works perfect. But this scenario is not good for my tests. What i really want is: UAC going to Kamailio IP address and Kamailio forwarding request to Asterisk IP without auth on Kamailio and auth with asterisk .conf files (i mean without mysql at all) But i wanna keep data about what to do will presence and other stuff Can you make changes in attached files please?

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Reload htable
by Grant Bagdasarian 28 Mar '15

28 Mar '15

Hello, I need to reload a htable from the database when a certain condition is met inside my kamailio script. I found the htable.reload htable RPC command, but is it possible to run this from within the script? Is there perhaps another way to force reload the htable inside the script? Thanks, Grant

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evapi_relay some problems
by dolphin 28 Mar '15

28 Mar '15

Hello, I test the evapi modules function , but not full operation ,only receive not send to client I capture the important setup parameters are as follows: debug=3 loadmodule "evapi.so" #----- evapi params ----- # modparam("evapi", "bind_addr", "0.0.0.0:8448") modparam("evapi", "netstring_format", 0) modparam("evapi", "workers", 4) event_route[evapi:connection-new] { evapi_relay("{ \"event\": \"test\",\n \"data\": { \"fU\": \"fU\" }\n}"); xlog("L_WARN", "new connection from $evapi(srcaddr):$evapi(srcport)\n"); } scene : One client connected kamailio server x.x.x.x:8448 use nc tools , I get garbled , kamailioserver not sent to me client , Because this is the right format to display josn ? sh-4.2# nc -n x.x.x.x 8448 ; øX±š^C any suggestions or missing something ? my syslog : Mar 28 14:31:37 SIP /home/pkg/kamailio-4.2.3/sbin/kamailio[297164]: DEBUG: evapi [evapi_dispatch.c:385]: evapi_accept_client(): new connection - pos[0] from: [x.x.x.x:56215] Mar 28 14:31:37 SIP /home/pkg/kamailio-4.2.3/sbin/kamailio[297164]: DEBUG: evapi [evapi_dispatch.c:531]: evapi_relay(): relaying event data [{ "event": "test",#012 "data": { "fU": "fU" }#012}] Mar 28 14:31:37 SIP /home/pkg/kamailio-4.2.3/sbin/kamailio[297164]: DEBUG: evapi [evapi_dispatch.c:561]: evapi_relay(): sent [0x7f939ab158f8] [{ "event": "test",#012 "data": { "fU": "fU" }#012}] (44) Mar 28 14:31:37 SIP /home/pkg/kamailio-4.2.3/sbin/kamailio[297164]: WARNING: <script>: new connection from x.x.x.x:56215 kamailio server : root@SIP:/etc/init.d# dpkg -l | grep libev-dev ii libev-dev 1:4.11-1 amd64 static library, header files, and docs for libev root@SIP:/etc/init.d# uname -a Linux SIP 3.2.0-4-amd64 #1 SMP Debian 3.2.46-1+deb7u1 x86_64 GNU/Linux root@SIP:/home/pkg/kamailio-4.2.3/sbin# ./kamailio -V version: kamailio 4.2.3 (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, F_MALLOC, DBG_F_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: unknown compiled on 15:13:09 Mar 27 2015 with gcc 4.7.2 download : http://www.kamailio.org/pub/kamailio/latest/src/kamailio-4.2.3_src.tar.gz

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Routing PSTN calls from Asterisk through Kamailio to UAC & loose_route
by Anthony Messina 27 Mar '15

27 Mar '15

I've been working on integration of Asterisk and Kamailio, currently on the same host with different ports, and have come across a problem with calls that originate from the Asterisk side (PSTN/DAHDI) and route through Kamailio to a SIP UAC. In short, when the SIP UAC (10.1.1.9) sends the BYE, loose_route() is returning -1 and the BYE is routed back to Kamailio (10.1.1.1:5060) instead of Asterisk (10.1.1.1:5080). I am using the stock WITHINDLG route configuration. RR module settings are as follows: modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 1) The BYE from the SIP UAC contains the following Route header which only contains the contents of Kamailio's Record-Route header. I have attached the full sip trace for review as well. Route: <sip:10.1.1.1;lr=on;ftag=2400d939-de0b-4456-9e01-f9a3302f3e25;nat=yes>. What would be the best method to resolve this issue in either Asterisk or Kamailio? Should I manually add a Record-Route header for the Asterisk host:port to Kamailio config? Is there something to be done in Asterisk? Thanks. -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E

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Kamailio - Used By Directory
by Daniel-Constantin Mierla 27 Mar '15

27 Mar '15

Hello, in order to facilitate easier interaction and updates, references (http://www.kamailio.org/w/references/) and some of the business use cases already listed on kamailio.org are in the process of being moved to a new page: - http://www.kamailio.org/w/used-by/ This offers now the possibility to submit new entries without the interaction with Kamailio admins that could lead to delays. The form is available at: - http://www.kamailio.org/w/used-by-form/ The goals is to show where Kamailio is used as well as increase the visibility of the products, solutions and services offered by the Kamailio community. Anyone is welcome to add own references. Besides a spambot protection of the form, we rely for now on the good manners of human nature, if you notice anything inadequate there, please report to be corrected. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

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tsilo module
by Alex Balashov 27 Mar '15

27 Mar '15

Hello, I don't understand the tsilo module. What is the practical use case for it? Thanks, -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

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Kamailio BLF Issue TCP vs UDP
by Fred Posner 27 Mar '15

27 Mar '15

Greetings all: Having a weird issue with BLF relay from Kamailio <-> FreeSWITCH on the EC2 network. Set-up: Natted Client <-> Natted Kamailio <-> Natted FreeSWITCH Kamailio has a public advertised IP (Amazon Cloud) and sends to FreeSWITCH's public IP (Amazon Cloud). Clients register to Kamailio and the registration sent to FreeSWITCH (via uac). What works: When a client registers to kamailio via UDP, all works beautifully. What doesn't work: When a client registers to kamailio via TCP, a NOTIFY is sent with almost no information. Example of UDP traffic: U 2015/03/26 22:12:30.483767 FREESWITCH_IPADDR:5060 -> KAMAILIO_NAT_IPADDR:5060 NOTIFY sip:3000@CLIENT_PUBLIC_IPADDR:42477;line=51w5wsns SIP/2.0. Via: SIP/2.0/UDP FREESWITCH_IPADDR;rport;branch=z9hG4bKv7X88FmKg04UF. Route: <sip:KAMAILIO_PUBLIC_IPADDR>;lr=on;ftag=xew5nmpr9q;nat=yes. Record-Route: <sip:KAMAILIO_PUBLIC_IPADDR>;lr=on;ftag=xew5nmpr9q;nat=yes. Max-Forwards: 70. From: <sip:3000@CLIENT_DOMAIN;user=phone>;tag=YF4HMSQNlOOt. To: <sip:3000@CLIENT_DOMAIN>;tag=xew5nmpr9q. Call-ID: 3134323734303730333531303039-ji1sw7gyd7uj. CSeq: 366877453 NOTIFY. Contact: <sip:3000@FREESWITCH_IPADDR:5060>. User-Agent: FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, path, replaces. Event: dialog. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: active;expires=2686. Content-Type: application/dialog-info+xml. Content-Length: 546. . <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="14" state="full" entity="sip:3000@CLIENT_DOMAIN"> <dialog id="312b3bee-d405-11e4-a8b6-41b65d0bd045" direction="initiator"> <state>confirmed</state> <local> <identity display="3000">sip:3000@CLIENT_DOMAIN</identity> <target uri="sip:3000@CLIENT_DOMAIN"> <param pname="+sip.rendering" pvalue="yes"/> </target> </local> <remote> <identity display="*98">sip:*98@CLIENT_DOMAIN</identity> <target uri="sip:**3000@CLIENT_DOMAIN"/> </remote> </dialog> </dialog-info> U 2015/03/26 22:12:30.483877 KAMAILIO_NAT_IPADDR:5060 -> CLIENT_PUBLIC_IPADDR:42477 NOTIFY sip:3000@CLIENT_PUBLIC_IPADDR:42477;line=51w5wsns SIP/2.0. Via: SIP/2.0/UDP KAMAILIO_PUBLIC_IPADDR:5060;branch=z9hG4bKfabf.4908255bf1623d250ddf7d51a3b49fb8.0. Via: SIP/2.0/UDP FREESWITCH_IPADDR;received=FREESWITCH_IPADDR;rport=5060;branch=z9hG4bKv7X88FmKg04UF. Record-Route: <sip:KAMAILIO_PUBLIC_IPADDR>;lr=on;ftag=xew5nmpr9q;nat=yes. Max-Forwards: 69. From: <sip:3000@CLIENT_DOMAIN;user=phone>;tag=YF4HMSQNlOOt. To: <sip:3000@CLIENT_DOMAIN>;tag=xew5nmpr9q. Call-ID: 3134323734303730333531303039-ji1sw7gyd7uj. CSeq: 366877453 NOTIFY. Contact: <sip:3000@FREESWITCH_IPADDR:5060>. User-Agent: FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, path, replaces. Event: dialog. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: active;expires=2686. Content-Type: application/dialog-info+xml. Content-Length: 546. . <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="14" state="full" entity="sip:3000@CLIENT_DOMAIN"> <dialog id="312b3bee-d405-11e4-a8b6-41b65d0bd045" direction="initiator"> <state>confirmed</state> <local> <identity display="3000">sip:3000@CLIENT_DOMAIN</identity> <target uri="sip:3000@CLIENT_DOMAIN"> <param pname="+sip.rendering" pvalue="yes"/> </target> </local> <remote> <identity display="*98">sip:*98@CLIENT_DOMAIN</identity> <target uri="sip:**3000@CLIENT_DOMAIN"/> </remote> </dialog> </dialog-info> Example of TCP failure: U 2015/03/26 22:19:45.963196 FREESWITCH_IPADDR:5060 -> KAMAILIO_NAT_IPADDR:5160 NOTIFY sip:3001@CLIENT_PUBLIC_IPADDR:5065;transport=tcp SIP/2.0. Via: SIP/2.0/UDP FREESWITCH_IPADDR;rport;branch=z9hG4bKjH5Nt8QKUU9Sp. Route: <sip:KAMAILIO_PUBLIC_IPADDR:5160;r2=on;lr=on;ftag=9jSFr02tgvHym;nat=yes>. Route: <sip:KAMAILIO_PUBLIC_IPADDR:5160;transport=tcp;r2=on;lr=on;ftag=9jSFr02tgvHym;nat=yes>. Max-Forwards: 70. From: "3000" <sip:3000@CLIENT_DOMAIN>;tag=9jSFr02tgvHym. To: <sip:3001@CLIENT_DOMAIN>;tag=1dba0f3ceb32436bi0. Call-ID: 0882b1bc-4ea9-1233-f3a2-12e513dfd8f0. CSeq: 73368127 NOTIFY. Contact: <sip:mod_sofia@FREESWITCH_IPADDR:5060>. User-Agent: FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, path, replaces. Event: refer;id=101. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: terminated;reason=noresource. Content-Type: message/sipfrag;version=2.0. Content-Length: 16. . SIP/2.0 200 OK. T 2015/03/26 22:19:45.963407 KAMAILIO_NAT_IPADDR:5160 -> CLIENT_PUBLIC_IPADDR:5065 [AP] NOTIFY sip:3001@CLIENT_PUBLIC_IPADDR:5065;transport=tcp SIP/2.0. Via: SIP/2.0/TCP KAMAILIO_PUBLIC_IPADDR:5160;branch=z9hG4bKe961.8d1f19f1acc0295e5e397afa76910115.0. Via: SIP/2.0/UDP FREESWITCH_IPADDR;received=FREESWITCH_IPADDR;rport=5060;branch=z9hG4bKjH5Nt8QKUU9Sp. Max-Forwards: 69. From: "3000" <sip:3000@CLIENT_DOMAIN>;tag=9jSFr02tgvHym. To: <sip:3001@CLIENT_DOMAIN>;tag=1dba0f3ceb32436bi0. Call-ID: 0882b1bc-4ea9-1233-f3a2-12e513dfd8f0. CSeq: 73368127 NOTIFY. Contact: <sip:mod_sofia@FREESWITCH_IPADDR:5060>. User-Agent: FreeSWITCH-mod_sofia/1.4.18+git~20150312T185523Z~4eed221b69~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, path, replaces. Event: refer;id=101. Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Subscription-State: terminated;reason=noresource. Content-Type: message/sipfrag;version=2.0. Content-Length: 16. . SIP/2.0 200 OK. My thoughts is that this is on FreeSWITCH, just not sending a NOTIFY with anything of value. That being said, I'd love to know if (a) anyone else has had this issue or (b) if someone has a different theory. Thanks! -- Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct)

4 11

rtpengine DTLS
by Marc Soda 25 Mar '15

25 Mar '15

I'm running a new kamailio/rtpengine instance and, using webrtc, I can't get audio to flow between the browser and rtpengine. Kamailio seems to be handing it off properly. Based on the rtpengine logs and a packet capture it doesn't look like DTLS is being negotiated properly: https://gist.github.com/marcantonio/ea0077a5884e4e4b6b45 I see "SRTCP output wanted, but no crypto suite was negotiated" and I've never seen that one before (the SRTCP part). I have this exact config working elsewhere, but this new setup is not working? Any advice? Thanks, Marc

1 1

Siremis error Call to a member function getQuoted()
by canuck15 25 Mar '15

25 Mar '15

When you add/update/view Administrator usernames To reproduce Administration > User Management > Click on any existing username. The browser goes blank. Error in apache logs is: PHP Fatal error: Call to a member function getQuoted() on a non-object in /var/www/html/openbiz/bin/data/BizDataSql.php on line 237 This is on the latest Siremis v4.2.0 downloaded today I am using CentOS v6.6, php v5.3.3

3 11

active_watchers - dbt_table_check_row(): null value not allowed - field 17
by Alexandru Raileanu 25 Mar '15

25 Mar '15

Hi, I have kamailio 4.2 with presence, auth, textdb activated. The scenario is that Kamailio receives a SUBSCRIBE, sends 200ok then a NOTIFY. (like it should do) The problem is what when I kill kamailio and run it again, it gives me this error: 0(30580) ERROR: db_text [dbt_tb.c:474]: dbt_table_check_row(): null value not allowed - field 17 0(30580) DEBUG: db_text [dbt_file.c:480]: dbt_load_file(): error at row=1 col=26 c=1 0(30580) ERROR: db_text [dbt_lib.c:307]: dbt_db_get_table(): could not load database from file [active_watchers] 0(30580) ERROR: db_text [dbt_base.c:460]: dbt_delete(): failed to load table <active_watchers>! The field with null values is record_route, so if I put some value I manage to run kamailio. my conf, maybe I missed some field? #!define DBURL "text:///etc/kamailio/dbtext" #!ifdef WITH_PRESENCE modparam("presence", "db_url", DBURL) modparam("presence", "max_expires", 3600) modparam("presence", "server_address", "sip:10.0.37.1:5060" ) modparam("pua_reginfo", "default_domain", "10.0.37.1") modparam("pua_reginfo", "server_address", "sip:10.0.37.1") modparam("pua_reginfo", "outbound_proxy", "sip:10.0.37.1") modparam("pua_reginfo", "publish_reginfo", 1) modparam("pua", "db_mode", 2) modparam("pua", "hash_size", 11) modparam("pua", "db_url", DBURL) modparam("pua", "db_table", "pua") modparam("pua", "min_expires", 0) modparam("pua", "default_expires", 3600) modparam("pua", "update_period", 1000) modparam("pua", "dlginfo_increase_version", 1) modparam("pua", "reginfo_increase_version", 1) modparam("pua", "check_remote_contact", 0) modparam("pua", "fetch_rows", 1000) modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif Or as a workaround, can I reset the active_watchers table every time I start kamailio? because *kamdbctl create/drop/init* wipes all the other tables... thanks a lot! Alex

2 2

File transferring issue
by Wingsravi R 25 Mar '15

25 Mar '15

Dear Kamailio'ns, I am working on kamailio Server (V 4.0) installed on Ubuntu 12.04 system. Currently i am trying file transfer feature using IMSDroid SIP clients. file transferring is fine between two IMSDroid clients but at the receiving end (file recipient) i cont see the sender identity/ sender name. it is just displaying as 'null' (Screen shot is attached). But the full file is transferring to destination address without any problem. I have configured MSRP module related script in my kamailio configuration file (find the attachment). but i guess it is not invoking anyway when file transferring and the file transferring is negotiating client-to-client basis. And also find Sip capture of file transfer (ngrep based) for better understanding. What could be the problem? Why at the file recipient side it is not displaying sender's identity (instead it is 'null')? how can i resolve this issue? Anyone please help me in resolving this issue. Any help will greatly appreciate. Regards, Ravi.

5 6

[4.2.x] "socket" feature missing in dispatcher module
by Sven Neuhaus 25 Mar '15

25 Mar '15

Hello, in 4.2.2, the "socket" module was added to the documentation of the dispatcher module (see http://kamailio.org/docs/modules/4.2.x/modules/dispatcher.html#dispatcher.e… ), however the code is missing, the feature isn't implemented in 4.2.x. This was in commit 9bb59f922. Please include the patch that adds this feature from the trunk in the 4.2.x branch, as documented. A brief search turned up the commits aae775859f54f84734ba2cfc2945c02c52b95e1f and 2a605383d5c524d9875256c7f95f64216b970331 with this feature, perhaps there are more. Thanks, -Sven Neuhaus

2 1

Re: [SR-Users] kamailio asterisk NOTIFY
by Slava Bendersky 24 Mar '15

24 Mar '15

Hello Daniel, Look like transactions are not match when it send reply for NOTIFY. I think it should reuse connection. Here SDP output where cseq are match and asterisk does retransmissions until time out. Also I wonder where Path route is reused I see %* instead ; as delimiter between transport and port. Retransmitting #8 (NAT) to 10.18.130.46:5060: NOTIFY sip:191@192.168.88.246:5062;transport=tcp SIP/2.0 Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK6939f20d;rport Max-Forwards: 70 Route: <sip:inbound@10.18.130.46;lr;received=sip:client_public_ip:5062%3Btransport%3Dtcp> From: "canlpbx02" <sip:canlpbx02@10.18.130.51>;tag=as4b212049 To: <sip:191@192.168.88.246:5062;transport=tcp> Contact: <sip:canlpbx02@10.18.130.51:5060> Call-ID: 14c7c8cc7335adf60f2be634132d3b2f@10.18.130.51:5060 CSeq: 102 NOTIFY User-Agent: FPBX-12.0.43(12.8.1) Event: message-summary Content-Type: application/simple-message-summary Content-Length: 89 Messages-Waiting: yes Message-Account: sip:*97@10.18.130.51 Voice-Message: 24/0 (0/0) --- Retransmitting #9 (NAT) to 10.18.130.46:5060: NOTIFY sip:191@192.168.88.246:5062;transport=tcp SIP/2.0 Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK6939f20d;rport Max-Forwards: 70 Route: <sip:inbound@10.18.130.46;lr;received=sip:99.251.238.38:5062%3Btransport%3Dtcp> right here From: "canlpbx02" <sip:canlpbx02@10.18.130.51>;tag=as4b212049 To: <sip:191@192.168.88.246:5062;transport=tcp> Contact: <sip:canlpbx02@10.18.130.51:5060> Call-ID: 14c7c8cc7335adf60f2be634132d3b2f@10.18.130.51:5060 CSeq: 102 NOTIFY User-Agent: FPBX-12.0.43(12.8.1) Event: message-summary Content-Type: application/simple-message-summary Content-Length: 89 Messages-Waiting: yes Message-Account: sip:*97@10.18.130.51 Voice-Message: 24/0 (0/0) --- <--- SIP read from UDP:10.18.130.46:5060 ---> SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK6939f20d;rport=5060;received=10.18.130.51 From: "canlpbx02" <sip:canlpbx02@10.18.130.51>;tag=as4b212049 To: <sip:191@192.168.88.246:5062;transport=tcp>;tag=dc299d9ac16f3377ac0b8dfa22eaf0a6-2fe2 Call-ID: 14c7c8cc7335adf60f2be634132d3b2f@10.18.130.51:5060 CSeq: 102 NOTIFY Server: kamailio (4.2.3 (x86_64/linux)) Content-Length: 0 <-------------> Mar 23 08:45:47 canlvprx01 /usr/sbin/kamailio[20143]: ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect client_pub_ip:5063 failed (timeout) Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Monday, March 23, 2015 4:21:47 AM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, as you can see the 408 is for another request, because the cseq is 103. The first one has cseq 102, but has no Record-Route header, which is mandated by new RFCs. In kamailio, if you added in request_route when handling the NOTIFY, you can try to add record route via reply_route if the client is not keeping it in reply, kind of following fix: reply_route { if(is_method("NOTIFY")) { if(!is_present_hf("Record-Route")) { insert_hf("Record-Route: < sip:yourserverip:port;transport=...\r\n "); } } Cheers, Daniel On 22/03/15 18:11, Slava Bendersky wrote: Hello Daniel, I still can't understand clearly why when asterisk send NOTIFY to client through kamailio it coming back with with 408. I I tried different case event change Contact:, but no luck can't I see what the issue all routes in first SUBSCRIBE looks ok . Any though or help will be really appreciate. This what I see on asterisk debug Reliably Transmitting (NAT) to 10.18.130.46:5060: NOTIFY sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK5dd3f266 Max-Forwards: 70 Route: <sip:10.18.130.46:5060;lr=on;ftag=3b27a2a409;nat=yes> --->Kamailio route From: "canlpbx02" <sip:canlpbx02@10.18.130.51> ;tag=as56402f92 To: <sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2> ;tag=3b27a2a409 Contact: <sip:canlpbx02@10.18.130.51:5060> Call-ID: b1edde040d695988 CSeq: 102 NOTIFY User-Agent: FPBX-12.0.43(12.8.1) Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 88 Messages-Waiting: yes Message-Account: sip:*97@10.18.130.51 Voice-Message: 5/0 (0/0) --- <--- SIP read from UDP:10.18.130.46:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.51:5060;rport=5060;branch=z9hG4bK5dd3f266 From: "canlpbx02" <sip:canlpbx02@10.18.130.51> ;tag=as56402f92 To: <sip:102@192.168.88.243:5068;alias=client_pub_ip~5068~2> ;tag=3b27a2a409 Call-ID: b1edde040d695988 CSeq: 102 NOTIFY Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, REFER, SUBSCRIBE, UPDATE Server: Media5-fone/4.1.6.3283 Android/5.1 Supported: eventlist, replaces, timer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:10.18.130.46:5060 ---> SIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 10.18.130.51:5060;branch=z9hG4bK6386387f;rport=5060 From: "canlpbx02" <sip:canlpbx02@10.18.130.51> ;tag=as00936a6d To: <sip:102@192.168.88.243:5066;alias=client_pub_ip~5066~2> ;tag=a3aea18741 Call-ID: 5a66b87a618d1042 CSeq: 103 NOTIFY Server: kamailio (4.2.3 (x86_64/linux)) Content-Length: 0 <-------------> Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, March 13, 2015 1:28:33 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, that function might be in textopsx module. Cheers, Daniel On 13/03/15 14:25, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Where possible find updated version of options for textops. [sr_module.c:701]: find_mod_export_record(): find_export_record: <remove_hf_value> not found Specially Record-Route manipulation. Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: miconda(a)gmail.com Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, March 13, 2015 8:54:26 AM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello Daniel, Here paste for first SUBSCRIBE. http://fpaste.org/197545/25118514/ Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, March 12, 2015 5:09:17 AM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, can you get the sip traffic from the network (from the first subscribe of this presence dialog)? It will help to see the Route set. Cheers, Daniel On 12/03/15 00:36, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Here debug of the notify which failing. Look like transaction miss match http://fpaste.org/196896/61168591/ Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 5:44:05 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, it doesn't matter if there is a mix of transports. If the UA is mirroring the Record-Route headers in the response to the NOTIFY, it should work fine. Cheers, Daniel On 11/03/15 21:40, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Is this matter that UDP and TCP transport in mix ? Slava. From: "Slava Bendersky" <volga629(a)networklab.ca> To: miconda(a)gmail.com Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 4:21:39 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello Daniel, I have record route in WITHINDLG for NOTIFY, but it not helped behaviour is the same it loosing route for NOTIFY. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "Slava Bendersky" <volga629(a)networklab.ca> , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, March 11, 2015 2:14:02 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, you have to do record_route() for all NOTIFY requests -- there are new specs requiring that (RFC 6665) and perhaps asterisk is following them. If you look in the default kamailio.cfg for v4.2, inside route[WITHINDLG], then you will see how it is done. Cheers, Daniel On 11/03/15 14:22, Slava Bendersky wrote: BQ_BEGIN Hello Daniel, Here fresh paste http://fpaste.org/196616/42608006/ I see proper routing headers on SUBSCRIBE, but then NOTIFY loose it. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Tuesday, March 10, 2015 5:31:51 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Hello, fpaste link doesn't show anything. Can you check if the NOTIFY coming to Kamailio has two Route headers? Cheers, Daniel On 10/03/15 18:44, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, Having issue with NOTIFY, is not forwarded to client properly. asterisk ---> private kamailio public ----> soft client http://fpaste.org/196257/99948714/ I see kamailio trying connect always with private IP to the client when NOTIFY is come is in. Is this something to do with rr_advertise_address ? Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5064 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64e2800 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 33, -1, 0x10) fd_no=24 called Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4320]: tcpconn_main_timeout(): tcp_main: entering timer for 0x7fbbb64eb6c0 (ticks=839577436, timeout=839634956 (3595 s), wr_timeout=839577436 (0 s)), write queue: 798 bytes Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect 192.168.88.246:5066 failed (timeout) Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [tcp_main.c:4362]: tcpconn_main_timeout(): tcp_main: timeout for 0x7fbbb64eb6c0 Mar 10 13:44:23 canlvprx01 kamailio: 16(24971) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9dbf00, 34, -1, 0x10) fd_no=23 called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: <core> [mem/shm_mem.c:111]: _shm_resize(): WARNING:vqm_resize: resize(0) called Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 Mar 10 13:44:42 canlvprx01 kamailio: 5(24960) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=0 I have in configuration in WITHINDLG if (is_method("NOTIFY")) { record_route(); } and in NATDETECT if(is_method("INVITE|SUBSCRIBE|UPDATE|NOTIFY")) { set_contact_alias(); } Slava. _______________________________________________ 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://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ 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 BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ 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 BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com _______________________________________________ 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 BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

5 8

msilo module
by Rajat Yadav 24 Mar '15

24 Mar '15

Hello i m very new to kamailio.I m trying to develop an application using webrtc and kamailio. In this i tried using msilo module so as to store offline messages but wasnt able to achieve it.After that i searched a lot on internet about working msilo codes but i didnt find it. Can someone please send me a working code of msilo which i can embed in the kamailio configuration file. Thank you.

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ERROR: sipcapture [hep.c:139]: hepv2_received(): ERROR: sipcapture:hep_msg_received: unknow protocol [1]
by Satish Patel 24 Mar '15

24 Mar '15

Hello, Recently i have installed Homer capture server with kamailio. It works! but in logs i am seeing many error like following, I am using Opensips 2.1 siptrace server to send all trace to Homer. ERROR: sipcapture [hep.c:139]: hepv2_received(): ERROR: sipcapture:hep_msg_received: unknow protocol [1] ERROR: sipcapture [hep.c:139]: hepv2_received(): ERROR: sipcapture:hep_msg_received: unknow protocol [1] ERROR: sipcapture [hep.c:139]: hepv2_received(): ERROR: sipcapture:hep_msg_received: unknow protocol [1]

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Compiling with db_unixodbc
by Grant Bagdasarian 24 Mar '15

24 Mar '15

Hello, I'm trying to compile Kamailio 4.2.3 on Debian 7.8 with db_unixodbc but the sql.h file can't be found. I've used the following command to compile and install unixODBC. tar -zxvf unixODBC-2.3.2.tar.gz && cd unixODBC-2.3.2 && ./configure --sysconfdir=/etc --prefix=/usr/local/unixODBC --disable-gui && make && make install #####################: locate odbc_config /usr/local/unixODBC/bin/odbc_config /usr/src/unixODBC-2.3.2/exe/odbc_config #####################: locate sql.h /usr/local/unixODBC/include/sql.h /usr/src/kamailio-4.2.3/modules/db_mysql/km_db_mysql.h /usr/src/kamailio-4.2.3/modules/db_postgres/pg_sql.h /usr/src/unixODBC-2.3.2/Drivers/Postgre7.1/isql.h /usr/src/unixODBC-2.3.2/Drivers/nn/isql.h /usr/src/unixODBC-2.3.2/Drivers/nn/nnsql.h /usr/src/unixODBC-2.3.2/exe/isql.h /usr/src/unixODBC-2.3.2/include/sql.h Kamailio Compile Error CC (gcc) [M db_unixodbc.so] connection.o In file included from connection.c:34:0: connection.h:42:17: fatal error: sql.h: No such file or directory compilation terminated. make[1]: *** [connection.o] Error 1 make: *** [modules] Error 1 Modules.lst include_modules= carrierroute db_unixodbc dialog_ng lcr tls uuid Removed these includes from the exclude_modules Any ideas?

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Inter Kamailio communication: UDP vs TCP vs SCTP
by Mickael Marrache 24 Mar '15

24 Mar '15

Hi, We are deploying a VoIP system based on Kamailio. The system is composed of load balancers, proxies, registrars, etc What is the recommended transport protocol to use for internal communication (i.e. load balancer <-> proxy/registrar)? We started with UDP but we also see that Kamailio supports TCP and SCTP. For information, at first stage, the UAs will communicate using UDP but we may need to use TCP in case messages are too large (we had such issues when using ICE). Thanks, Mickael

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Re: [SR-Users] kamailio asterisk NOTIFY
by Slava Bendersky 23 Mar '15

23 Mar '15

Hello Everyone, In my case I am using Path and I asterisk generate NOTIFY with route based on path, then I see on kamailio side tries reach on client on public IP side and open new connection not follow ESTABLISHED/RELATED link then it times out. Log: Mar 23 11:00:17 canlvprx01 /usr/sbin/kamailio[20193]: INFO: <script>: [NOTIFY], Inbound connection from <10.18.130.51> traffic from Asterisk Mar 23 11:00:17 canlvprx01 /usr/sbin/kamailio[20193]: INFO: <script>: [NOTIFY], Present Route: Client_public_IP. Mar 23 11:00:28 canlvprx01 /usr/sbin/kamailio[20204]: ERROR: <core> [tcp_main.c:4338]: tcpconn_main_timeout(): connect Client_public_IP:5060 failed (timeout) Here trace on kamailio side http://fpaste.org/201813/15287114/ Here trace from soft phone client side. http://fpaste.org/201814/14271530/ Slava. From: "Alex Balashov" <abalashov(a)evaristesys.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Monday, March 23, 2015 6:42:59 PM Subject: Re: [SR-Users] kamailio asterisk NOTIFY Anthony, The Contact presented by the subscriber in the initial subscription is: m: "Test User" <sip:172.16.4.7;line=sr-D8G7CE2.5PUeK-xuarl7NYDdNYDxNYlFUYoeUeQ8Cw.6DE2vDdyJDAa4TliwC84OC82LK-2ehwl7NYDdNYDxNYlFUAm6UYzm0gme>;+sip.ice And the subsequent NOTIFY is correctly targeted to this RURI: NOTIFY sip:172.16.4.7;line=sr-D8G7CE2.5PUeK-xuarl7NYDdNYDxNYlFUYoeUeQ8Cw.6DE2vDdyJDAa4TliwC84OC82LK-2ehwl7NYDdNYDxNYlFUAm6UYzm0gme SIP/2.0 The subscriber returns this Contact in the 200 OK response for the NOTIFY: m: "Test User" <sip:172.16.4.7;line=sr-D8G7CE2.5PUeK-xuarl7NYDdNYDxNYlFUYoeUeQ8Cw.6DE2vDdyJDAa4TliwC84O>;+sip.ice And subsequent NOTIFYs appear to use this RURI instead of the original Contact of the subscriber: NOTIFY sip:172.16.4.7;line=sr-D8G7CE2.5PUeK-xuarl7NYDdNYDxNYlFUYoeUeQ8Cw.6DE2vDdyJDAa4TliwC84O SIP/2.0 NOTIFY sip:172.16.4.7;line=sr-D8G7CE2.5PUeK-xuarl7NYDdNYDxNYlFUYoeUeQ8Cw.6DE2vDdyJDAa4TliwC84O SIP/2.0 etc. A SUBSCRIBE is a dialog-creating event, so in-dialog messages (e.g. NOTIFYs) should be targeted at a Request URI corresponding to the Contact URI of the SUBSCRIBE-er. It seems to me that this is the problem here, rather than truncation. -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ _______________________________________________ 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

Kamalio call issue
by Yogendra Gupta 23 Mar '15

23 Mar '15

Hello, We have setup kamalio at server and it is working at some DNS with call/chat. At some IP chat is working but when we call then it is not working. At some DNS(Net IP) it is working . What can be problem at server? Following is login for SIP test user: tester1(a)23.253.110.48 123456 Host IP : 23.253.110.48 I have attached mu kamalio config file. Let me know what we need to change at config file or server Thanks

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What is the best SIP trunk authentication strategy
by canuck15 23 Mar '15

23 Mar '15

Been struggling with this for awhile now. So far I am finding it rather difficult to come up with way to authenticate SIP trunks taking into account all possible scenarios. My setup is Kamailio combined with Asterisk realtime. Everything is in a MySQL database. All authentication is done by Kamailio. Kamailio handles SIP extension (user) authentication quite well but seems poorly equipped for SIP trunk (peer) authentication. The biggest problem I have run into so far is that SIP trunks typically use DNS names and Kamailio is not at all designed to used DNS names. Also SIP trunks do not use named "realms" whereas that is a key part of Kamailio authentication. For IP authentication about the only solution I have found is to DNS lookup and save all returned IP addresses in the ip_addr mysql table. Then I do: #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address() && $au == "") { # source IP allowed return; } I can run a cron job every hour to DNS lookup and update the ip_addr table as needed so I think this is a satisfactory solution for IP authentication. SIP trunk user/pass authentication is the one I am now struggling with. This standard Kamailio authentication section does not work for SIP trunks. if (is_method("REGISTER|INVITE") || from_uri==myself) { # authenticate requests #if (!auth_check("$fd", "subscriber", "1")) { if (!auth_check("$fd", "subscriber", "0")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); This is a multidomain setup and therefore: modparam("auth_db", "use_domain", 1) So the authentication section will try authenticate the realm of the trunk which will always be an IP address. However the SIP trunk realm will usually be saved in the "subscriber" database as the FQDN. There is no mechanism for auth_check() to directly or indirectly check against FQDN. The other problem is that even if I used the ip_addr table somehow it will only try use the user/pass of the first instance of a matching IP address. For SIP trunks it is possible I could have multiple subaccounts and therefore multiple instances of the same IP with different user/pass. So auth_check() is not designed for being used this way as far as I can tell. Seems to me like there should almost be a special kamailio module just for SIP trunks. I had a look a carrierroute module it's not designed for this either.

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