- sr-users - kamailio.org

Kamailio stopped, but kamctl online still get user info
by Eirc 12 May '14

12 May '14

Hello, all of users: I am confused here, my kamalio is stopped, but kamctl online and ul show still can see the user info: ============================================ root@kamailio:/home/admin_1# kamctl online 102 root@kamailio:/home/admin_1# kamctl online 102 root@kamailio:/home/admin_1# kamctl online 102 root@kamailio:/home/admin_1# service kamailio status Status of Kamailio: kamailio is not running. root@kamailio:/home/admin_1# kamctl ul show Domain:: location table=512 records=1 max_slot=1 AOR:: 102 Contact:: sip:102@192.168.1.105:5060 Q= Expires:: 313 Callid:: 363490571-5060-1(a)BJC.BGI.B.BAF Cseq:: 2117 User-agent:: Grandstream GXP2124 1.0.4.10 State:: CS_SYNC Flags:: 0 Cflag:: 0 Socket:: udp:192.168.1.116:5060 Methods:: 7135 Ruid:: uloc-536d9a68-6a5-1 Instance:: <urn:uuid:00000000-0000-1000-8000-000B824017A0> Reg-Id:: 1 root@kamailio:/home/admin_1# ========================================== Does kamctl only receive the data from mysql, no matter kamctl status? -- *Best regards*

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Kamailio Logged in users
by Neville D'Souza 12 May '14

12 May '14

Hi, I am having a slight issue with Kamilio server. How do I disconnect users without restarting the server. When I run the command kamctl ul users i get the list of users connected, some that are already disconnected. I am using the Linphone android client to coonect through 5060 port. Kindly advise Best Regards, Neville D'Souza Network Administrator r_gogopay_96x73 Gogopay Online Pvt Ltd. India +91 832 222 0119 ncd(a)gogopay.com www.gogopay.com

2 1

One way audio
by aft 12 May '14

12 May '14

Hi, The network topology is : SipSoftphone(caller)----->kamailio/rtpproxy---->Softswitch---->sipGateway-->mobile phone(callee) The rtpproxy is managed in kamailio script by : if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; if (is_request()) { rtpproxy_offer("co"); } if (is_reply()) { rtpproxy_answer("coz80"); } if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) set_contact_alias(); } } Now if in rtpproxy_answer(), i don't resize rtp packets, then everything works fine. If i put "z80" then only one way audio can be heard. In SipSoftphone(caller) side, nothing can be heard. I have sniffed the SIP packets and it looked fine to me. The rtp packets from callee arrives fine in kamailio/rtpproxy box. But it does not leave that box. I've sniffed the rtp packets. Before the call is picked up by the callee, there is an ivr played. The ivr arrives at the caller(softphone) end fine. But after picking up the phone, rtppackets are not delivered to the caller(softphone)'s side. This is the rtpproxy's log: May 11 23:54:29 65429 rtpproxy[24207]: INFO:rxmit_packets: callee's address latched in: 85.13.205.178:13716 (RTP) May 11 23:54:41 65429 rtpproxy[24207]: DBUG:get_command: received command "27802_12 Uc18,96 2b15632de8c94ff488e9cdc1cdbcb85d 182.16.156.25 4000 74d009e3a57246138e2e0124cbe6ee89;1" May 11 23:54:41 65429 rtpproxy[24207]: INFO:handle_command: adding strong flag to existing session, new=1/0/0 May 11 23:54:41 65429 rtpproxy[24207]: INFO:handle_command: lookup on ports 54920/64810, session timer restarted May 11 23:54:41 65429 rtpproxy[24207]: DBUG:doreply: sending reply "27802_12 54920 64.120.57.111#012" May 11 23:55:42 65429 rtpproxy[24207]: INFO:process_rtp: session timeout May 11 23:55:42 65429 rtpproxy[24207]: INFO:remove_session: RTP stats: 1523 in from callee, 300 in from caller, 396 relayed, 0 dropped May 11 23:55:42 65429 rtpproxy[24207]: INFO:remove_session: RTCP stats: 0 in from callee, 8 in from caller, 8 relayed, 0 dropped May 11 23:55:42 65429 rtpproxy[24207]: INFO:remove_session: session on ports 54920/64810 is cleaned up May 11 23:58:58 65429 rtpproxy[24207]: DBUG:get_command: received command "27802_13 Uc18,96 008e2697e52a4adda7be16358b318def 182.16.156.25 4002 907ba9691f6a47f3bfb0bb4143ef7dc7;1" May 11 23:58:58 65429 rtpproxy[24207]: INFO:handle_command: new session 008e2697e52a4adda7be16358b318def, tag 907ba9691f6a47f3bfb0bb4143ef7dc7;1 requested, type strong May 11 23:58:58 65429 rtpproxy[24207]: INFO:handle_command: new session on a port 53876 created, tag 907ba9691f6a47f3bfb0bb4143ef7dc7;1 May 11 23:58:58 65429 rtpproxy[24207]: INFO:handle_command: pre-filling caller's address with 182.16.156.25:4002 May 11 23:58:58 65429 rtpproxy[24207]: DBUG:doreply: sending reply "27802_13 53876 64.120.57.111#012" May 11 23:58:59 65429 rtpproxy[24207]: DBUG:get_command: received command "27803_10 LZ80c18 008e2697e52a4adda7be16358b318def 85.13.205.178 26728 907ba9691f6a47f3bfb0bb4143ef7dc7;1 6611E63499885;1" May 11 23:58:59 65429 rtpproxy[24207]: INFO:handle_command: lookup on ports 53876/42866, session timer restarted May 11 23:58:59 65429 rtpproxy[24207]: INFO:handle_command: pre-filling callee's address with 85.13.205.178:26728 May 11 23:58:59 65429 rtpproxy[24207]: INFO:handle_command: RTP packets from callee will be resized to 80 milliseconds May 11 23:58:59 65429 rtpproxy[24207]: DBUG:doreply: sending reply "27803_10 42866 64.120.57.111#012" May 11 23:58:59 65429 rtpproxy[24207]: INFO:rxmit_packets: callee's address filled in: 85.13.217.158:26728 (RTP) May 11 23:58:59 65429 rtpproxy[24207]: INFO:rxmit_packets: guessing RTCP port for callee to be 26729 May 11 23:58:59 65429 rtpproxy[24207]: INFO:rxmit_packets: caller's address latched in: 182.16.156.25:4003 (RTCP) May 11 23:58:59 65429 rtpproxy[24207]: INFO:rxmit_packets: caller's address filled in: 182.16.156.25:1036 (RTP) May 11 23:58:59 65429 rtpproxy[24207]: DBUG:get_command: received command "27801_10 LZ80c18 008e2697e52a4adda7be16358b318def 85.13.205.178 26728 907ba9691f6a47f3bfb0bb4143ef7dc7;1 6611E63499885;1" May 11 23:58:59 65429 rtpproxy[24207]: INFO:handle_command: lookup on ports 53876/42866, session timer restarted May 11 23:58:59 65429 rtpproxy[24207]: INFO:handle_command: RTP packets from callee will be resized to 80 milliseconds May 11 23:58:59 65429 rtpproxy[24207]: DBUG:doreply: sending reply "27801_10 42866 64.120.57.111#012" May 11 23:59:00 65429 rtpproxy[24207]: DBUG:get_command: received command "27802_14 LZ80c18 008e2697e52a4adda7be16358b318def 85.13.205.178 26728 907ba9691f6a47f3bfb0bb4143ef7dc7;1 6611E63499885;1" May 11 23:59:00 65429 rtpproxy[24207]: INFO:handle_command: lookup on ports 53876/42866, session timer restarted May 11 23:59:00 65429 rtpproxy[24207]: INFO:handle_command: RTP packets from callee will be resized to 80 milliseconds May 11 23:59:00 65429 rtpproxy[24207]: DBUG:doreply: sending reply "27802_14 42866 64.120.57.111#012" May 11 23:59:02 65429 rtpproxy[24207]: DBUG:get_command: received command "27800_13 LZ80c18 008e2697e52a4adda7be16358b318def 85.13.205.178 26728 907ba9691f6a47f3bfb0bb4143ef7dc7;1 6611E63499885;1" May 11 23:59:02 65429 rtpproxy[24207]: INFO:handle_command: lookup on ports 53876/42866, session timer restarted May 11 23:59:02 65429 rtpproxy[24207]: INFO:handle_command: RTP packets from callee will be resized to 80 milliseconds May 11 23:59:02 65429 rtpproxy[24207]: DBUG:doreply: sending reply "27800_13 42866 64.120.57.111#012" May 11 23:59:06 65429 rtpproxy[24207]: DBUG:get_command: received command "27801_11 LZ80c18 008e2697e52a4adda7be16358b318def 85.13.205.178 26728 907ba9691f6a47f3bfb0bb4143ef7dc7;1 6611E63499885;1" May 11 23:59:06 65429 rtpproxy[24207]: INFO:handle_command: lookup on ports 53876/42866, session timer restarted May 11 23:59:06 65429 rtpproxy[24207]: INFO:handle_command: RTP packets from callee will be resized to 80 milliseconds May 11 23:59:06 65429 rtpproxy[24207]: DBUG:doreply: sending reply "27801_11 42866 64.120.57.111#012" May 11 23:59:10 65429 rtpproxy[24207]: DBUG:get_command: received command "27802_15 LZ80c18 008e2697e52a4adda7be16358b318def 85.13.205.178 26728 907ba9691f6a47f3bfb0bb4143ef7dc7;1 6611E63499885;1" May 11 23:59:10 65429 rtpproxy[24207]: INFO:handle_command: lookup on ports 53876/42866, session timer restarted May 11 23:59:10 65429 rtpproxy[24207]: INFO:handle_command: RTP packets from callee will be resized to 80 milliseconds May 11 23:59:10 65429 rtpproxy[24207]: DBUG:doreply: sending reply "27802_15 42866 64.120.57.111#012" May 11 23:59:20 65429 rtpproxy[24207]: INFO:rxmit_packets: callee's address latched in: 85.13.217.158:26728 (RTP) I don't what is going wrong here, but somehow rtpproxy is not relaying the packets from callee's end. But the addresses seems fine and whenever i remove the "z80" inside rtpproxy_answer(), It works fine. At first i thought its just that the resizer is broken. But when the ivr is played, the rtppackets of the ivr works just fine. -- -Cheers -Arif

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Help diagnosing non-response from incoming call
by Alex Villacís Lasso 09 May '14

09 May '14

I have this setup for kamailio + asterisk, in which kamailio is supposed to listen on all ethernet interfaces on UDP port 5060, and will forward traffic from/to asterisk running on the same machine, and listening on localhost, udp port 5080. The scenario for the problematic call is somewhat like this: SIP-PROVIDER<--->NAT<--192.168.0.0/16-->eth0:192.168.10.10<--KAMAILIO-->127.0.0.1<--ASTERISK Our firewall/NAT has been configured to redirect SIP traffic from the SIP provider to the kamailio+asterisk machine at IP 192.168.10.10. The attached file sip-traffic-from-eth0.txt shows a tcpdump capture of an incoming call at eth0. Then, kamailio is supposed to redirect this traffic to asterisk. The attached file sip-traffic-from-localhost.txt shows a tcpdump capture of the same call at 127.0.0.1. The issue is that the INVITE is received, then routed to asterisk, which sends back the 200 OK, but then there is no ACK from the SIP provider. From the point of view of the caller of the SIP provider, the destination just keeps ringing until timeout. Am I right in assuming that the two Record-Route headers should not appear on the traffic as seen from eth0, and that they are the source of the trouble? Can you see any additional issues I might have not seen in the traffic?

1 0

Multiple port configuration
by Alex Villacís Lasso 09 May '14

09 May '14

How do I configure kamailio to listen to multiple ports, on all known interfaces? I have tried "listen=udp:*:5060", "listen=udp:*:5062" on separate lines, but I get the following errors: Not starting : invalid configuration file! 0(7806) : <core> [cfg.y:3411]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 190, column 12: syntax error 0(7806) : <core> [cfg.y:3411]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 190, column 12: ip address, interface name or hostname expected 0(7806) : <core> [cfg.y:3411]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 190, column 13: ERROR: bad config file (3 errors) I have also tried "port=5060", then "port=5062" on separate lines, but kamailio only listens on the port indicated by the last line. I do not want to have to enumerate every single interface on the configuration file.

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route(RELAY) - what exactly execute?
by Patrik Kristel 09 May '14

09 May '14

Hello, I would like to ask what exactly means route(RELAY)? I don't found some declaration of it. I{ found it in many method before exit; I think its probably last handle of SIP message, am I right? Thank you Patrik

1 0

/tmp/kamailio_fifo is missing
by Patrik Kristel 09 May '14

09 May '14

Hello all, I trying to debug Errorwith mi_fifo an d I found out that file kamailio_fifo is missing from the directory /tmp, also I not found it in my Debian by locate command. I have correctly following lines in my conf files: in kamailio.cfg: loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") in kamctlrc: OSER_FIFO="/tmp/kamailio_fifo" So, I dont know, it maybe my mysql crashed? Thank you for you help! Patrik

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kamailio load balancer despatch to asterisk12 plus PJSIP Stack Mediagateways
by Rainer Piper 09 May '14

09 May '14

Hi, I'm using kamailio as a load balancer to dispatch to 5 asterisk Media gateways with pjsip Stack The old asterisk sip Stack is turned off. hier my asterisk pjsip.conf to trust kamailios IP Addr. pjsip.conf [transport-udp] type=transport protocol=udp bind=0.0.0.0:5060 tos=cs3 cos=3 [kamailio] type=endpoint context=outgoing-kamailio disallow=all allow=g722,alaw,ulaw transport=transport-udp direct_media=no disable_direct_media_on_nat=yes aors=kamailio [kamailio] type=aor qualify_frequency=60 [kamailio] type=identify endpoint=kamailio match=[IP-Addr-of-your-kamailio] extensions.conf exten => _XXXX.,1,Dial(PJSIP/${EXTEN}@trunkprofil) od exten => _XXXX.,1,Dial(PJSIP/trunkprofil/${EXTEN}) works great and pjsip is very fast !!! love it !!! Regards Rainer -- *Rainer Piper* Integration engineer Koeslinstr. 56 53123 BONN GERMANY Phone: +49 228 97167161 <callto:004922897167161>

1 0

Starting Kamailio with init script
by VOIP Tests 09 May '14

09 May '14

Hello, can someone let me know how I can start Kamailio with the init script? In the /etc/default/kamailio file I have commented out RUN_KAMAILIO=yes, this starts Kamailio but my phones do not register. If I stop the init script and start kamailio with kamctl then my phones register, what am I doing wrong here? Also, if my understanding is correct, if I start Kamailio with the init script I should not start kamailio with kamctl, in that case how do I run the kamctl commands, for eg like kamctl ul show ? Thanks for the help. Arun

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Re: [SR-Users] issue with usr_loc db_mode
by Daniel-Constantin Mierla 09 May '14

09 May '14

Hello, can you enable query log in mysql and verify if they happen in such case? I checked and for db_mode=1, it should immediately do the query. For db_mode=3, th queries are done on time. Also, is there any other application that could potentially delete records? Because after the first registration of a phone (which does an INSERT), there will be UPDATE queries, and if the record is not there, there is nothing to update. Look at usrloc module parameters, there are some parameters to try to do insert if the update doesn't affect records, you can try with this one to see if there is a difference. Cheers, Daniel On 08/05/14 18:20, Henry Fernandes wrote: > There are no errors in the log about this. > > Yesterday, I tried db_mode=2 (with default 'timer_interval' of 60 seconds) and it's better now, although still not working properly. There are still some records that are not in the 'location' table in the database. > > For example, I have a phone that has a registration expirations are set to 1 hour (3600 seconds). Currently, this record is not in the database however "kamctl ul show" shows the following: > > AOR:: phoneUsername > Contact:: sip:phoneUsername@a.b.c.d:62544 Q= > Expires:: 3123 > Callid:: f1e0aa72-eb0d28e2-c3737132(a)10.1.0.14 > Cseq:: 52 > User-agent:: PolycomVVX-VVX_500-UA/4.1.6.4835 > State:: CS_SYNC > Flags:: 0 > Cflag:: 0 > Socket:: udp:a.b.c.d:5060 > Methods:: 8159 > Ruid:: uloc-536a9e57-3137-2 > Reg-Id:: 0 > > In other words, it's been more than 60 seconds but the database is not up to date. This is similar to what I see with db_mode=1. In that case, the database is often missing more than 50% of the registered phones. > > Here are the parameters I'm using for the USRLOC module. > > modparam("usrloc", "db_url", WRITE_DBURL) > modparam("usrloc", "db_mode", 2) > modparam("usrloc", "use_domain", 0) > > -H > > > On 2014-05-08, at 1:00 AM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > >> Hello, >> >> any db_mode>0 in 3.3 is safe for not losing registrations upon kamailio restart (e.g., db_mode=2 writes to db on time and at shutdown, so nothing is lost as well). db_mode=1 should do that in realtime, indeed. >> >> I'm not aware of any issue with db_mode=1, being used in quite some deployments. Can you check the syslog to see if there is any error reported there? >> >> If you use mysql, then you can enable logging the queries and then check if for each registration there is a mysql query to insert/update/delete a record. >> >> Cheers, >> Daniel >> >> On 08/05/14 00:08, Henry Fernandes wrote: >>> I'm using Kamailio 3.3 and the USRLOC module to track registrations. I have set "db_mode" to 1 because I want to store all registration info to the database and I want the database to be up to date (in case I need to restart Kamailio). >>> >>> Unfortunately, this doesn't work as I expect. When I query the database, it has far fewer entries in the "location" table than are registered to Kamailio (which I find with a "kamctl ul show"). Also, when I restart Kamailio, I lose some of the registrations for my phones. >>> >>> Am I misunderstanding the behaviour for db_mode=1? Isn't it supposed to keep the database up to date? >>> -H >>> _______________________________________________ >>> 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://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> >> >> _______________________________________________ >> 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://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

What do next?
by Jaafar Hamza 09 May '14

09 May '14

Hi, I am a student in Programming, I develop for project of the end of study a sip phone(Swing application). I use Sip Jain api. I installed kamailio in Ubuntu Server 14.04 using this command : apt-get install kamailio*. Now, I don't know what to do to configure this server like a registrar and I would like use the presence module. Please help me, I need it. Thanks Hamza Jaafar

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sip trunk help
by Joli Martinez 09 May '14

09 May '14

Hello, I am new to Kamailio. I need to get a SIP trunk registered to Kamailio. Can someone explain how to set this up. Thanks

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issue with usr_loc db_mode
by Henry Fernandes 08 May '14

08 May '14

I'm using Kamailio 3.3 and the USRLOC module to track registrations. I have set "db_mode" to 1 because I want to store all registration info to the database and I want the database to be up to date (in case I need to restart Kamailio). Unfortunately, this doesn't work as I expect. When I query the database, it has far fewer entries in the "location" table than are registered to Kamailio (which I find with a "kamctl ul show"). Also, when I restart Kamailio, I lose some of the registrations for my phones. Am I misunderstanding the behaviour for db_mode=1? Isn't it supposed to keep the database up to date? -H

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n00b question -- RADIUS authentication.
by Måns Nilsson 08 May '14

08 May '14

[first post to list] Greetings, I'm in the process of getting a Kamailio 3.3.2 installation authenticating its SIP accounts against a RADIUS database. There are -- at the moment -- no plans to do any fancy accounting nor any authorisation beyond simple authentication. I've set up and tested a FreeRadius 2.2.3_1 server on a dedicated server. After a fairly steep learning curve involving RADIUS dictionaries I've come so far that kamailio sends out a RADIUS Access-Request message that is received by FreeRadius, processed, and returned to Kamailio which promptly ignores it and continues to send 401 to the client; the SIP message exchange with the client being: (some identifing info has been redacted) > REGISTER sip:my.domain SIP/2.0 < SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.25.191.24:41688;branch=z9hG4bK-d8754z-eac09e6c626d4c4d-1---d8754z-;rport=41688 > REGISTER sip:my.domain SIP/2.0 Via: SIP/2.0/UDP 10.25.191.24:41688;branch=z9hG4bK-d8754z-4f25c643f4b93465-1---d8754z-;rport < SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.25.191.24:41688;branch=z9hG4bK-d8754z-4f25c643f4b93465-1---d8754z-;rport=41688 The RADIUS exchange: 10:07:10.861063 IP (tos 0x0, ttl 64, id 14964, offset 0, flags [none], proto UDP (17), length 270) 10.24.194.198.63712 > 10.24.194.196.1812: [udp sum ok] RADIUS, length: 242 Access Request (1), id: 0x05, Authenticator: 4215e95809551826eda76972be4106c4 Username Attribute (1), length: 18, Value: mtu-06(a)my.domain 0x0000: 6d74 752d 3036 4069 706b 2e73 722e 7365 Unknown Attribute (207), length: 10, Value: 0x0000: 0a08 6d74 752d 3036 Unknown Attribute (207), length: 13, Value: 0x0000: 010b 6970 6b2e 7372 2e73 65 Unknown Attribute (207), length: 36, Value: 0x0000: 0222 5532 6448 326c 4e6e 5271 3677 4353 0x0010: 6463 6775 5056 3050 516e 3936 324d 5635 0x0020: 6d34 Unknown Attribute (207), length: 17, Value: 0x0000: 040f 7369 703a 6970 6b2e 7372 2e73 65 Unknown Attribute (207), length: 12, Value: 0x0000: 030a 5245 4749 5354 4552 Unknown Attribute (207), length: 8, Value: 0x0000: 0506 6175 7468 Unknown Attribute (207), length: 12, Value: 0x0000: 090a 3030 3030 3030 3031 Unknown Attribute (207), length: 36, Value: 0x0000: 0822 3933 3832 3333 3333 3530 3162 3238 0x0010: 6439 3236 3739 3863 3964 3038 6539 3134 0x0020: 3733 Unknown Attribute (206), length: 34, Value: 0x0000: 3538 3665 3336 3763 3230 3163 3137 6438 0x0010: 6261 3265 3830 3533 3763 6433 3562 3761 Service Type Attribute (6), length: 6, Value: #15 0x0000: 0000 000f Unknown Attribute (208), length: 8, Value: 0x0000: 6d74 752d 3036 NAS Port Attribute (5), length: 6, Value: 5060 0x0000: 0000 13c4 NAS IP Address Attribute (4), length: 6, Value: 10.24.194.198 0x0000: c079 c2c6 10:07:10.863964 IP (tos 0x0, ttl 64, id 28916, offset 0, flags [none], proto UDP (17), length 48) 10.24.194.196.1812 > 10.24.194.198.63712: [bad udp cksum 0x06ac -> 0x44c0!] RADIUS, length: 20 Access Accept (2), id: 0x05, Authenticator: 8f07de871a066aacfbe822e20a9b96c1 The RADIUS part of the Kamailio config is: if (is_method("REGISTER") || from_uri==myself) #if (is_method("REGISTER") ) { # authenticate requests xlog("L_INFO", "authenticate [$fd]\n"); ### RADIUS ### if (!radius_www_authorize("my.domain")) { $var(ret) = $rc; xlog("L_INFO", "response code: [$var(ret)]\n"); switch ($var(ret)) { case -7: send_reply("500", "Server Internal Error"); exit; case -1: send_reply("400", "Bad Request"); exit; default: }; if (defined($avp(digest_challenge)) && ($avp(digest_challenge) != "")) { append_to_reply("$avp(digest_challenge)"); }; send_reply("401", "Unauthorized"); exit; }; # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } Any clues? What is missing from my narrative? -- Måns Nilsson primary/secondary/besserwisser/machina MN-1334-RIPE +46 705 989668 Actually, what I'd like is a little toy spaceship!!

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kamctl online and ul show are empty!
by 5060 08 May '14

08 May '14

Hello: after i starting kamailio for while, the system shows: 3(1466) DEBUG: mi_fifo [mi_parser.c:84]: end of fifo input tree 3(1466) DEBUG: mi_fifo [fifo_fnc.c:507]: done parsing the mi tree 1(1464) DEBUG: usrloc [urecord.c:327]: Binding '102',' sip:102@192.168.1.105:5060' has expired 3(1466) DEBUG: mi_fifo [mi_parser.c:84]: end of fifo input tree 3(1466) DEBUG: mi_fifo [fifo_fnc.c:507]: done parsing the mi tree 3(1466) DEBUG: mi_fifo [mi_parser.c:84]: end of fifo input tree 3(1466) DEBUG: mi_fifo [fifo_fnc.c:507]: done parsing the mi tree ---------------------------------------------------------------------- kamctl online and ul show are empty. how can i adjust the time to resolve the problem? thanks!

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Re: [SR-Users] UPDATE --- Can "presence" module handle publish reqs with several dialog entries? + Problem SOLUTION
by Klaus Feichtinger 08 May '14

08 May '14

Pls find attached the patch for version 4.1.3. The three lines of code are equal for all affected version (incl. 3.2.4, 3.3.5, 4.0.6 and 4.1.3). regards, Klaus > Daniel-Constantin Mierla <miconda(a)gmail.com> hat am 7. Mai 2014 um 15:45 > geschrieben: > > Hello, > > can you send the patch to be reviewed? > > Cheers, > Daniel > > On 07/05/14 13:15, Klaus Feichtinger wrote: > > > > Hi Daniel, > > > > I think we´ve found the reason, why this problem occurs! > > > > The problem is caused in the "agregate_xmls" function in file > > "notify_body.c" of the "presence_dialoginfo" module: > > > > /* loop over all bodies and create the aggregated body */ > > for(i=0; i<j; i++) > > { > > /* LM_DBG("[n]=%d, [i]=%d, [j]=%d xml_array[i]=%p\n", n, i, j, > > xml_array[j] ); */ > > p_root= xmlDocGetRootElement(xml_array[i]); > > if(p_root ==NULL) { > > LM_ERR("while geting the xml_tree root element\n"); > > goto error; > > } > > if (p_root->children) { > > for (node = p_root->children; node; node = node->next) { > > if (node->type == XML_ELEMENT_NODE) { > > LM_DBG("node type: Element, name: %s\n", > > node->name); > > /* we do not copy the node, but unlink it and then > > add it ot the new node > > * this destroys the original document but we do > > not need it anyway. > > * using "copy" instead of "unlink" would also copy > > the namespace which > > * would then be declared redundant (libxml > > unfortunately can not remove > > * namespaces) > > */ > > if (!force_single_dialog || (j==1)) { > > xmlUnlinkNode(node); > > if(xmlAddChild(root_node, node)== NULL) { > > LM_ERR("while adding child\n"); > > goto error; > > } > > > > It seems to be not the best idea to "unlink" the XML node (= > > "xmlUnlinkNode(node);"), as then no "node->next" is available any more. > > Therefore, this loop will _always_ stop after one dialog entry and ignore > > any additional one! > > > > But we "solved" the problem in this way that the loop will not directly > > use "node->next", but a variable, which is set within the loop. It looks > > like this: > > xmlNodePtr next_node = NULL; > > [...] > > if (p_root->children) { > > for (node = p_root->children; node; node = next_node) { > > next_node = node->next; > > if (node->type == XML_ELEMENT_NODE) { > > [...] > > > > This solution has been tested with 2 and 3 dialog entries in a single > > PUBLISH request and it is working fine. We should discuss if this is a > > problem that should be solved generally or if it is a "private" problem in > > our use case. > > > > What do you mean? > > > > regards, > > Klaus > > > > > > >

2 1

Problems this morning using Iptel.org
by swill 08 May '14

08 May '14

Hi, I use a configuration of ipkall and iptel.org, Most of the time this works out as a great incoming solution, however, today, it seems to be ending the call as i answer, I use only one softpphone at a time, but have tried it on Zoiper, 3cx, and Phonerlite, It does the same thing, call it, it rings, answer it and it immediately goes to "Call ended". I've done powercycle reboots on my HTC EVO 4G LTE, and on my PC it acts the same on all three. It registers fine on them all, I just cant determine if it is an IPKall or an Iptel problem..Is anyone else having an issue?? Are there any suggestions on what further troubleshooting I could do?? Thanks Michael

2 1

Kamailio auto restart if stopped
by JR Richardson 07 May '14

07 May '14

Hi All, Just had a strange occurrence where kamailio has been up and running fine for several months and just died with a strange error: root@sip-router3:~# kamailio -V version: kamailio 4.1.1 (x86_64/linux) ac7d55 May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2275]: : <core> [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0x7faf565586d0), called from <core>: mem/shm_mem.c: sh_realloc(88), first free <core>: mem/shm_mem.c: sh_realloc(88) - aborting May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2271]: ALERT: <core> [main.c:775]: handle_sigs(): child process 2275 exited by a signal 6 May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2271]: ALERT: <core> [main.c:778]: handle_sigs(): core was not generated May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2271]: INFO: <core> [main.c:790]: handle_sigs(): INFO: terminating due to SIGCHLD May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2279]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2277]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2273]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2272]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2274]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2276]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2278]: INFO: <core> [main.c:841]: sig_usr(): INFO: signal 15 received May 7 10:09:51 sip-router3 /usr/local/sbin/kamailio[2271]: ERROR: ctl [ctl.c:379]: mod_destroy(): ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) Not sure what that means, memory leak? I started started it right back up with no indication of a problem. Running Kamailio for many years, I have never had an instance of it just shut down before. So I'm wondering if there are any safe scripts that check the running process for kamailio and restart if automatically if not running? I dug around a bit but not seeing any hits for auto start kamailio scripts. And another question would be if this occurrence of shutting down should be of concern or not? Thanks. JR -- JR Richardson Engineering for the Masses

2 1

Can "presence" module handle publish reqs with several dialog entries?
by Klaus Feichtinger 07 May '14

07 May '14

Hi, during simulating scenarios in relation with the notify problem I have so far, I became unsure, if Kamilio (in principal) is supporting handling of PUBLISH requests (event "dialog") that contain _more than_ one (1) dialog entry. I know that it can generate notify requests with a message body that contains more than one dialog entry. But that´s not automatically the same as handling publish requests.... So - is Kamailio able or not? Did anybody ever use it in such a scenario? regards, Klaus

2 7

Problem initial configuration of Kamalio
by Mr.Gab 07 May '14

07 May '14

Hello everybody, i'm new to kamalio and i'm trying to install on a centos server hosted on a open-stack IaaS platform. I followed the official wiki adding the "advertise statement" and define WITH_NAT. I added 3 user as test 100 101 102 that i can see typing kamctl db show subscriber but if i try to register with Bria i get Failed. Theese are sip packets that i sniffed with tcpdump. http://pastebin.com/sa74HJqM can someone help a noob to understad how to debug this kind of problem with kamalio? i tried googling but i didn't find anything. Thank you all in advice

1 0

Routing incoming calls to Asterisk trunk
by Teleport Engineer 07 May '14

07 May '14

Hello, I'd setup a test environment where Kamailio accepts external invites and routes them to an internal Asterisk (actually working as an SBC). My dialplan works fine routing the INVITE to the internal Asterisk, but the SIP message goes into the defaul sip context (the one configured in sip general), and not in the context defined for the sip trunk between Kamailio and Asterisk. The sip trunk is configured like this: [kamailio] disallow=all host=kamailio port=5060 insecure=port,invite type=friend context=from-sip-external fromdomain=mydomain qualify=yes allow=alaw allow=g722 So I expect that all the sip invites coming from kamailio should be executed in from-sip-external context. But it doesn't work like this since each invite goes in the default context. The invite that goes to Asterisk is like this <--- SIP read from kamailio-external:5060 ---> INVITE sip:extension@asterisk:5060;transport=UDP SIP/2.0 Record-Route: <sip:kamailio-external:5060;nat=yes;ftag=905e4921;lr=on> Via: SIP/2.0/UDP kamailio-external:5060;branch=z9hG4bK277a.aea3ad87.0 Via: SIP/2.0/UDP caller:28021;rport=28019;received=caller;branch=z9hG4bK-d8754z-d11f1fa1d0fb2ab4-1---d8754z- Max-Forwards: 69 Contact: <sip:524@caller:28019;transport=UDP> To: <sip:extension@kamailio-external> From: <sip:524@caller:28007>;tag=905e4921 Call-ID: YzEzMDM1NDZhZTllMTRiMDJmY2MyZTE0ZWZjMzE5OTc. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.2.21357 r21367 Allow-Events: presence, kpml Content-Type: application/sdp Content-Length: 260 What I think is that in the sip message there is no information about kamailio internal ip address but only about kamailio-external address and then Asterisk put it into the default context. If this is the reason, should I change the trunk ip address in Asterisk and should I do some manipulation on the incoming packet on kamailio? Or maybe I should use kamailio-external as fromdomain in the trunk configuration? Thanks.

2 1

Asterisk-Kamailio set up
by VOIP Tests 07 May '14

07 May '14

Hello, I have a set up with Asterisk-Kamailio as explained in http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb. This set up has been working well for us for sometime now. We are now moving Asterisk to a stand alone server ( not installing it on the same box as asterisk ) but we are running into some issues. I have just made some changes to the Asterisk bind IP address and port and also set up Kamalio as a peer on asterisk. When I make an extension to extension call the calls are failing from the FROMASTERISK route ( call is getting cancelled ). Can someone let me know what changes I need to make to correct the issue? Some pointer will help. Thank you, Arun

2 2

How to enable TLS compression
by 刘日新 06 May '14

06 May '14

Hi, all. I has configure kamailio.cfg as below: #!ifdef WITH_TLS modparam("tls", "session_id", "vic22") modparam("tls", "session_cache",1) modparam("tls", "tls_disable_compression",0) modparam("tls", "config", "/usr/cfg/tls.cfg") #!endif I was sure that this configure segment was included , in case that I set a wrong path to modparam("tls", "config", "/wrong paths/tls.cfg") the kamailio can`t boost and raised error. But, when I debug this configure with gdb, I found that, within method : init_tls_compression(), Line 384 of tls_init.c 384 } else if (cfg_get(tls, tls_cfg, disable_compression)){ 385 LOG(L_INFO, "tls: init_tls: disabling compression...\n"); 386 sk_SSL_COMP_zero(comp_methods); The param value from cfg was always 1, namely: * print ((struct cfg_group_tls *)tls_cfg)->disable_compression 1 Why that? Are there something wrong? B.R. Rixin liu

4 3

Dispatcher question
by Volkan Oransoy 06 May '14

06 May '14

Hi all, I have a multidomain setup that I share load, among two freeswtich servers with a kamailio loadbalancer using dispatcher. For specific applications like queues, I need to match a sip domain with a freeswitch server. A domain should match same freeswitch server on every request. Which algorithm should I use with ds_selecy_dst, or what is the most appropriate way to do this? Thanks. /Volkan

2 1

Port 5061 problem
by aawaise 05 May '14

05 May '14

Hello, I have configured TLS on kamailio server. Problem is that in kamailio.cfg, when I use command port =5061. Kamailio doesnot start and error given in logs is >> ERROR: tcp_init : bind(a, 0x9007f4, 16) on MyIP:5061 : Adress already in >> use >> ERROR: Error while initializing TCP port of TLS Socket MyIP:5061 I tried the command " netstat -lntp | grep 5061 " to see what service or app is using the port 5061. I found no result. It is not used by any application apparently. Any help will be highly appreciated. Thanks. Aawaise -- View this message in context: http://sip-router.1086192.n5.nabble.com/Port-5061-problem-tp127464.html Sent from the Users mailing list archive at Nabble.com.

5 6

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