- sr-users - kamailio.org

407 Proxy Authentication Required with Cisco
by proLogika 25 Jul '14

25 Jul '14

Hello, I am using Kamailio as SIP register with asterisk integration describe from hire: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . asterisk is listen on public ip: 2.3.4.5:5060 kamailio is listen on public ip: 1.2.3.4:5060 everything is working well except some software VoIP clients (like Yate) and CISCO phone like Cisco-CP7940G/8.0 and the new one from Cisco series. I’m testing now with Yate client and Cisco. They are register OK but when a call is made Kamailio is answer back with 407 Proxy Authentication Required. When I register Yate or Cisco to asterisk directly the call is passing normaly. I was trying to manipulate kamailio.cfg and more specifically the part: #!ifdef WITH_ASTERISK if (!auth_check("$fd", "sipusers", "1")) { #!else if (!auth_check("$fd", "subscriber", "1")) { #!endif auth_challenge("$fd", "0"); exit; If i commented out this part the call is passing, but I do not have auth anymore (everyone can register) Here is ngrep: U 2014/07/23 19:17:08.108458 192.168.0.40:5060 -> 1.2.3.4:5060 INVITE sip:0896995837@1.2.3.4 SIP/2.0. Max-Forwards: 20. Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK1899510692. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>. Call-ID: 931919626(a)1.2.3.4. CSeq: 13 INVITE. User-Agent: YATE/5.3.0. Contact: <sip:10891@192.168.0.40:5060>. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO. Content-Type: application/sdp. Content-Length: 481. . v=0. o=yate 1406132227 1406132227 IN IP4 192.168.0.40. s=SIP Call. c=IN IP4 192.168.0.40. t=0 0. m=audio 29696 RTP/AVP 0 8 11 98 97 102 103 104 105 106 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:11 L16/8000. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode=20. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:102 SPEEX/8000. a=rtpmap:103 SPEEX/16000. a=rtpmap:104 SPEEX/32000. a=rtpmap:105 iSAC/16000. a=rtpmap:106 iSAC/32000. a=rtpmap:101 telephone-event/8000. a=ptime:30. U 2014/07/23 19:17:08.108805 1.2.3.4:5060 -> 192.168.0.40:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK1899510692. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=6be166fd53062bbc5b6dd79656b620cd.1950. Call-ID: 931919626(a)1.2.3.4. CSeq: 13 INVITE. Proxy-Authenticate: Digest realm="1.2.3.4", nonce="U8/hMFPP4AR+N3A+ZccNiTw6rV9JTq1I". Server: kamailio (4.0.6 (x86_64/linux)). Content-Length: 0. . U 2014/07/23 19:17:08.128626 192.168.0.40:5060 -> 1.2.3.4:5060 ACK sip:0896995837@1.2.3.4 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK1899510692. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=6be166fd53062bbc5b6dd79656b620cd.1950. Call-ID: 931919626(a)1.2.3.4. CSeq: 13 ACK. Max-Forwards: 20. Contact: <sip:10891@192.168.0.40:5060>. User-Agent: YATE/5.3.0. Content-Length: 0. . U 2014/07/23 19:17:08.129076 192.168.0.40:5060 -> 1.2.3.4:5060 INVITE sip:0896995837@1.2.3.4 SIP/2.0. Max-Forwards: 20. Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>. Call-ID: 931919626(a)1.2.3.4. User-Agent: YATE/5.3.0. Contact: <sip:10891@192.168.0.40:5060>. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO. CSeq: 14 INVITE. Proxy-Authorization: Digest username="10891", realm="1.2.3.4", nonce="U8/hMFPP4AR+N3A+ZccNiTw6rV9JTq1I", uri="sip:0896995837@1.2.3.4", response="b96eebd48b3734e1d018a970fa3a2283", algorithm=MD5. Content-Type: application/sdp. Content-Length: 481. . v=0. o=yate 1406132227 1406132227 IN IP4 192.168.0.40. s=SIP Call. c=IN IP4 192.168.0.40. t=0 0. m=audio 29696 RTP/AVP 0 8 11 98 97 102 103 104 105 106 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:11 L16/8000. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode=20. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:102 SPEEX/8000. a=rtpmap:103 SPEEX/16000. a=rtpmap:104 SPEEX/32000. a=rtpmap:105 iSAC/16000. a=rtpmap:106 iSAC/32000. a=rtpmap:101 telephone-event/8000. a=ptime:30. U 2014/07/23 19:17:08.129622 1.2.3.4:5060 -> 192.168.0.40:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>. Call-ID: 931919626(a)1.2.3.4. CSeq: 14 INVITE. Server: kamailio (4.0.6 (x86_64/linux)). Content-Length: 0. . U 2014/07/23 19:17:08.130107 1.2.3.4:5060 -> 2.3.4.5:5060 INVITE sip:0896995837@1.2.3.4 SIP/2.0. Record-Route: <sip:1.2.3.4;lr=on;ftag=838449717;nat=yes>. Max-Forwards: 16. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK73f1.c43865a50d7a342a46dcebf824782de0.0. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>. Call-ID: 931919626(a)1.2.3.4. User-Agent: YATE/5.3.0. Contact: <sip:10891@192.168.0.40:5060>. Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO. CSeq: 14 INVITE. Content-Type: application/sdp. Content-Length: 499. . v=0. o=yate 1406132227 1406132227 IN IP4 1.2.3.4. s=SIP Call. c=IN IP4 1.2.3.4. t=0 0. m=audio 21888 RTP/AVP 0 8 11 98 97 102 103 104 105 106 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:11 L16/8000. a=rtpmap:98 iLBC/8000. a=fmtp:98 mode=20. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:102 SPEEX/8000. a=rtpmap:103 SPEEX/16000. a=rtpmap:104 SPEEX/32000. a=rtpmap:105 iSAC/16000. a=rtpmap:106 iSAC/32000. a=rtpmap:101 telephone-event/8000. a=ptime:30. a=nortpproxy:yes. U 2014/07/23 19:17:08.130593 2.3.4.5:5060 -> 1.2.3.4:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK73f1.c43865a50d7a342a46dcebf824782de0.0;received=1.2.3.4;rport=5060. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=as31a58bb2. Call-ID: 931919626(a)1.2.3.4. CSeq: 14 INVITE. Server: Asterisk PBX 1.8.29.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7531867c". Content-Length: 0. . U 2014/07/23 19:17:08.130770 1.2.3.4:5060 -> 2.3.4.5:5060 ACK sip:0896995837@1.2.3.4 SIP/2.0. Max-Forwards: 16. Via: SIP/2.0/UDP 1.2.3.4;branch=z9hG4bK73f1.c43865a50d7a342a46dcebf824782de0.0. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=as31a58bb2. Call-ID: 931919626(a)1.2.3.4. CSeq: 14 ACK. Content-Length: 0. . U 2014/07/23 19:17:08.131361 1.2.3.4:5060 -> 192.168.0.40:5060 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.0.40:5060;rport=5060;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=as31a58bb2. Call-ID: 931919626(a)1.2.3.4. CSeq: 14 INVITE. Server: Asterisk PBX 1.8.29.0. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE. Supported: replaces, timer. WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7531867c". Content-Length: 0. . U 2014/07/23 19:17:08.149847 192.168.0.40:5060 -> 1.2.3.4:5060 ACK sip:0896995837@1.2.3.4 SIP/2.0. Via: SIP/2.0/UDP 192.168.0.40:5060;rport;branch=z9hG4bK302777344. From: <sip:10891@1.2.3.4>;tag=838449717. To: <sip:0896995837@1.2.3.4>;tag=as31a58bb2. Call-ID: 931919626(a)1.2.3.4. CSeq: 14 ACK. Max-Forwards: 20. Contact: <sip:10891@192.168.0.40:5060>. Proxy-Authorization: Digest username="10891", realm="1.2.3.4", nonce="U8/hMFPP4AR+N3A+ZccNiTw6rV9JTq1I", uri="sip:0896995837@1.2.3.4", response="b96eebd48b3734e1d018a970fa3a2283", algorithm=MD5. User-Agent: YATE/5.3.0. Content-Length: 0. — And kamailio.cfg attached: proLogika Sent with Airmail

2 1

MESSAGE forking bug(?)
by Allen Zhang 24 Jul '14

24 Jul '14

Hi, I believe there is a bug on MESSAGE forking. Test scenario: User A has two records in the location table, with different instance ids: 111 and 222. One of user A's instance is killed, hence 222 is not reachable. User B send a MESSAGE to user A. Kamailio forked the MESSAGE. Kamailio got a '200 Message delivered' response from 111 and forwarded it to user B. The MESSAGE forked to 222 timed out. Kamailio send a '408 Time out' to user B. This violates RFC 3428 section 4. It states that the forking UAS should only send ONE final response to the sender UAC. Forking is done by t_relay(), and the registrar parameter 'append_branches' is set to default value 1. Did I do something wrong, or there is a bug? Regards, Allen

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Late parallel forking
by Guillaume Bour 24 Jul '14

24 Jul '14

Hi, I would like to do some kind of "late parallel forking" (see the example below). Is there a way to do such thing with Kamailio ? Here is an example: . A call B . B device 1 is connected so INVITE is forwarded to it . B device 2 registers succesfully . as device 1 as not yet answered and INVITE transaction is not expired, I want INVITE message to be forwarded to device 2 (so both are ringing) Regards, Guillaume Bour

4 6

Outbound registration?
by James Cloos 23 Jul '14

23 Jul '14

I don't see anything relevant in the docs, nor greping the src, but in case I missed something: Does the core, or do any of the modules support originating registration requests to other proxies a/o endpoints? I'd like to move registration responsibility to kama, and have it add a header on incoming INVITEs and the like from anything with which it has REGISTERed indicating which outbound registration is relevant to said request. If I'm right that there isn't support for that, do any of the app modules expose enough sip capability easily to write such? -JimC -- James Cloos <cloos(a)jhcloos.com> OpenPGP: 0x997A9F17ED7DAEA6

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Configuration kamailio + siremis
by Tomasz Sidło 23 Jul '14

23 Jul '14

Hello all I'm looking for the person who configures my kamailio + siremis I'll pay for the full configuration. Kamailio is to work together with a asterisk if anyone willing please contact me mail tomasz(a)sidlo.pl

1 0

Kamailio RtpProxy MHomed
by Pascal Fautré 23 Jul '14

23 Jul '14

Hi, I tried to use Kamailio / RTPProxy in mhomed setup without any luck. I had no problem to configure it with only 1 interface, without mhomed, everything worked perfectly. The RTP streams where not established correctly even if I managed to have to proper IP in the SIP INVITE (C & O). Versions: version: kamailio 4.1.4 (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_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled on 04:23:19 Jun 13 2014 with gcc 4.7.2 RTPProxy -v: Basic version: 20040107 Extension 20050322: Support for multiple RTP streams and MOH Extension 20060704: Support for extra parameter in the V command Extension 20071116: Support for RTP re-packetization Extension 20071218: Support for forking (copying) RTP stream Extension 20080403: Support for RTP statistics querying Extension 20081102: Support for setting codecs in the update/lookup command Extension 20081224: Support for session timeout notifications Here is my RTPProxy config (/etc/default/rtpproxy) : CONTROL_SOCK=udp:127.0.0.1:7722 EXTRA_OPTS=“-l PU.BL.IC.IP/PRI.VA.TE.IP -m 11000 -M 12000 -d DBUG:LOG_LOCAL3 Here are snippets of my kamailio.cfg: port=5060 mhomed=1 # RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return; xlog("NATMANAGE M=$rm OU=$ou RURI=$ru RD=$rd F=$fu T=$tu NH=$nh(d) IP=$si ID=$ci\n"); if(dst_ip == PUBLIC_IP) { if(is_ipv4($nh(d)) && is_in_subnet($nh(d), PRIVATE_NET)) { xlog("NATMANAGE coei\n"); rtpproxy_manage("coei", PRIVATE_IP); } else { xlog("NATMANAGE coee\n"); rtpproxy_manage("coee", PUBLIC_IP); } } else { if(is_ipv4($nh(d)) && is_in_subnet($nh(d), PRIVATE_NET)) { xlog("NATMANAGE coii\n"); rtpproxy_manage("coii", PRIVATE_IP); } else { xlog("NATMANAGE coie\n"); rtpproxy_manage("coie", PUBLIC_IP); } } 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(); } } #!endif return; } Calls were correctly going to the desired rtpproxy_manage options. Now I’m not quite sure I’m using the correct ones. I had to specify the PUBLIC_IP or PRIVATE_IP in the rtpproxy_manage calls in order to have the correct IP address in the C and O headers of the SIP INVITE. Without that, the public IP would be sent as C and O params to phones on the private subnet. In fact not a single call direction would give correct RTP streams. Any idea where I missed the turn? Cheers

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core:drop_requests meaning
by Alex Balashov 23 Jul '14

23 Jul '14

Hi, I was wondering what precisely these stats mean: core:drop_replies = 0 core:drop_requests = 3 Assuming one does not explicitly drop() requests anywhere in script, would a continuous monotonic increase in drop_requests be considered a problem? Or are there some normal circumstances and ordinary behaviours (i.e. auth challenges) that would register a request drop? Thanks, -- Alex -- Alex Balashov - Principal Evariste Systems LLC Tel: +1-678-954-0670 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ Please be kind to the English language: http://www.entrepreneur.com/article/232906

2 3

server for git repository is down
by Daniel-Constantin Mierla 22 Jul '14

22 Jul '14

Hello, a short note to say that the server for git repository on sip-router.org is down. The hosting provider hasn't reported any maintenance, I am digging further into it to see what happened. Meanwhile, the mirror on github can be used if anyone wants to fetch the source tree: - https://github.com/kamailio/kamailio Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 1

How install Kamailio with Asterisk
by ANTHONY HESNAUX 22 Jul '14

22 Jul '14

hello all, I would install kamailio with ASTERISK in realtime on CENTOS 6.5 x64. I followed the tutorial aspito: http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but its not work. I have a problem with kamailio.cfg Its not possible to start kamailio When I loadmodule "uac.so" I want to make this configuration: SIP phone1 -> Register Kamailio -> ASTERISK Register SIP phone 2 -> Register Kamailio -> ASTERISK Register kamailio: 192.168.0.100 Port: 5060 ASTERISK: 192.168.0.100 Port 5080

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Unknown caller gets online user's identity
by g.aloitus＠gmail.com 21 Jul '14

21 Jul '14

Hello, I'm using Kamailio version 4.1.4+precise (amd64). I have followed "Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration using Asterisk Database" (http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb). One main difference in my setup compared to that one is that I continued use of Kamailio's database. The problem is as follows: I decided to put Kamailio and through it Asterisk reachable from internet. I have tried to configure Asterisk so that only calls of registered users would be possible, and they could only call to other registered users or conference rooms and echo test number. Then I took the following steps: I ensured that there was no online users with kamctl online. Then I launched MicroSIP (www.microsip.org), but I did not defined account, I simply set the protocol to tls and media encryption to mandatory, because I'm using these. I called to extension with xxx(a)my.public.ip.address (where xxx is extension) getting "unauthorized". And that was what I wanted. But if there is online users, calls go through, and incoming call is coming from Asterisk (in syslog I can find out that src_user=asterisk). Kamailio and Asterisk are listening the same IP address, but different port. I have refused connections to the Asterisk's port with iptables. I have defined my public IP address as domain in sip.conf. There is also other domain defined which corresponds to users' domain I am using in Kamailio's database. In kamailio.cfg there is if statement which prevents Kamailio not to be open relay: if (from_uri!=myself && uri!=myself) ... If I change this for example: if (from_uri!=myself || uri!=myself) I get what I want this time: no calls from outside, but I somewhat think that this is not a final solution. I have not found from log files such information which would have helped me. I have not yet investigated this problem so much that I could tell the logic behind the selection of online user's identity which is used. However, if I make a call to conference room I notice that Asterisk is thinking that one of online users has joined the conference. If I can recall correctly, I started with Kamailio version 3.2, and integrated it with Asterisk 11 (currently 11.10.2). Is there something which has changed in Kamailio, but what I have not changed in my setup which could explain this. Best, Teijo

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kamcmd htable access problem, probably with int->string conversion
by Moritz Graf 21 Jul '14

21 Jul '14

Hi There, i implemented a mapping via htable. For debugging purpose I use kamcmd to see if the values are correctly filled. Doing so I discovered that there is a problem getting the values of the keys, where the name is a integer value. So it's the following behaviour: ##################################################################### root@kamailio-stage:~# kamcmd htable.dump maptodest { entry: 55 size: 1 slot: { item: { name: 1 value: 0000000000000000001 } } } { entry: 504 size: 1 slot: { item: { name: blub value: 0000000000000000001 } } } { entry: 686 size: 1 slot: { item: { name: 001 value: 0000000000000000001 } } } root@kamailio-stage:~# kamcmd htable.get maptodest blub { item: { name: blub value: 0000000000000000001 } } root@kamailio-stage:~# kamcmd htable.get maptodest 001 error: 500 - Not enough parameters (htable name and key name) ##################################################################### Is there a reasonable explanation why it works with blub, but not with 001? Why is he complaining about the number of parameters? Does anyone know a workarround? Thanks for any advise!! greeetz Moritz -- Moritz Graf, M.Sc. Betrieb NGN-Plattform G-FIT GmbH & Co. KG Greflingerstr. 26, 93055 Regensburg

2 3

pkg.stats
by Vitaliy Aleksandrov 21 Jul '14

21 Jul '14

Hi list, I have a problem with pkg.stats for tcp.main process on kamailio "4.1.2 73ea61-dirty". kamailio has 64M of private memory for each process (-M 64). pkg.stats for tcp.main process shows that is uses 75M and still has 62M free: { entry: 36 pid: 10032 rank: -4 used: 73109720 free: 65891192 real_used: 79146048 } root@webRTC:~# ps -p 10032 -o pid,tid,state,pcpu,pmem,rss,size,sz,vsz,args PID TID S %CPU RSS SIZE SZ VSZ COMMAND 10032 10032 S 0 26420 67508 76821 307284 /.../kamailio -f /.../kamailio.cfg -P /.../kamailio.pid -m 128 -M 64 -u kamailio -g kamailio This server handles 1000+ tcp/tls connections and works just fine for the last two months. That's why i think this is not a memory leak and just a pkg.stats problem. Does anybody else saw this or maybe somebody is already working to fix it ?

2 3

Msilo not delivering message (status 408)
by Peter Villeneuve 21 Jul '14

21 Jul '14

Hi, This is most likely a stupid mistake on my route config, but msilo correctly stores messages for offline users, but when they come back online, the message never arrives. Looking through the debug logs indeed there is an error, with status 408, which suggests kamaiio can't find a route to the user that just logged. Before I spend hours looking through all the routing blocks, perhaps someone more exprienced in kamailio can see the problem right away and offer a solution? Thanks DEBUG: msilo [msilo.c:1148]: m_dump(): msg [1-12] for: sip:1010@my.domain.com DEBUG: <core> [io_wait.h:617]: io_watch_del(): DBG: io_watch_del (0x8e3040, 9, -1, 0x10) fd_no=2 called DEBUG: <core> [tcp_read.c:1437]: release_tcpconn(): releasing con 0x7f61ced1b1b0, state 1, fd=9, id=1 DEBUG: <core> [tcp_read.c:1438]: release_tcpconn(): extra_data 0x7f61ced1c4f8 DEBUG: <core> [tcp_main.c:3385]: handle_tcp_child(): handle_tcp_child: reader response= 7f61ced1b1b0, 1 from 0 DEBUG: <core> [io_wait.h:390]: io_watch_add(): DBG: io_watch_add(0x89ec40, 26, 2, 0x7f61ced1b1b0), fd_no=21 DEBUG: <core> [tcp_main.c:3515]: handle_tcp_child(): handle_tcp_child: CONN_RELEASE 0x7f61ced1b1b0 refcnt= 1 DEBUG: tm [t_reply.c:1304]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 DEBUG: tm [t_reply.c:2086]: local_reply(): DEBUG: local_reply: branch=0, save=0, winner=0 DEBUG: tm [t_reply.c:2123]: local_reply(): DEBUG: local transaction completed DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x7f61ced1d868, callback type 1024, id 0 entered DEBUG: msilo [msilo.c:1317]: m_tm_callback(): *completed with status 408* [mid: 140058058335472/13] DEBUG: msilo [msilo.c:1325]: m_tm_callback(): message <13> was not sent successfully DEBUG: msilo [ms_msg_list.c:236]: msg_list_set_flag(): mid:13 fl:8 DEBUG: tm [t_reply.c:1304]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=408 DEBUG: tm [t_reply.c:2086]: local_reply(): DEBUG: local_reply: branch=0, save=0, winner=0 DEBUG: tm [t_reply.c:2123]: local_reply(): DEBUG: local transaction completed DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x7f61ced5f098, callback type 1024, id 0 entered DEBUG: msilo [msilo.c:1317]: m_tm_callback(): completed with status 408 [mid: 140058058329184/12] DEBUG: msilo [msilo.c:1325]: m_tm_callback(): message <12> was not sent successfully DEBUG: msilo [ms_msg_list.c:236]: msg_list_set_flag(): mid:12 fl:8

4 19

please help with Msilo config
by Miguel Rios 19 Jul '14

19 Jul '14

Hi list, I'm a newbie when it comes to kamailio, although I have a fair amount of SIP experience. I'm trying to setup a very basic kamailio install (4.1 on Wheezy) with Msilo support. I just used the default kamailio.cfg file (changing obviously the relevant parameters for my setup) and have not touched the routing blocks. I'm very confused about how routing works, and I don't have a background in C. I've read the wiki up and down and have looked at inumerous tutorials spread out on the internet (most of which just add to my confusion because they seem out of date), but I still don't get it. I managed to setup a working kamailio server where local users can call each other fine. Now when I try to add the msilo module, and specially the routing example from MSILO Module is when things get tricky. I'm sure my error is a fairly basic one and has to do with the routing logic and syntax. Could some kind soul please share a copy of the whole ####### Routing Logic ######## on downwards for a basic no frills kamailio with msilo setup? Thanks, Miguel MSILO Module 3. Parameters 3.1. db_url (string) Database URL. Default value is “mysql://kamailio:kamailiorw@localhost/kamailio”. Example 1.1. Set the “db_url” parameter View on kamailio.org Preview by Yahoo

3 3

Fwd: Msilo not delivering message (status 408)
by Peter Villeneuve 18 Jul '14

18 Jul '14

Thanks Daniel. UDP is the same story as TCP and TLS, unfortunately. Looking through the logs, I see strange errors in parsing the contact header when msilo prepares to send the notification back to 1000. This is for the notification message that should have been sent back to 1000 after he sent a message to 1010 (offline). I also see that kamailio reports that 1000's location was not found, even though he's obviously registered. The more I play around with this, the more confused I become. Do you think this contact parsing error could be the root of the problems? Cheers, Peter Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request: Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <MESSAGE> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: <sip:1000@my.domain.com> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK90dc.68116a97000000000000000000000000.0>; state=16 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [receive.c:152]: receive_msg(): After parse_msg... Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [receive.c:193]: receive_msg(): preparing to run routing scripts... Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_addr_spec.c:893]: parse_addr_spec(): end of header reached, state=10 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [28]; uri=[sip:1000@my.domain.com] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [< sip:1000@my.domain.com>#015#012] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <10> <MESSAGE> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=25 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=533cb9e91f4b999cf76861cbb9ed54ed-bb2e Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [parser/parse_addr_spec.c:893]: parse_addr_spec(): end of header reached, state=29 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks result: 1 Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: siputils [sipops.c:161]: w_is_first_hop(): no 2nd via found - first hop Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: *ERROR: <core> [parser/contact/contact.c:175]: skip_name(): skip_name(): Error in contact, scheme separator not found* Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: ERROR: <core> [parser/contact/contact.c:202]: parse_contacts(): parse_contacts(): Error while skipping name part Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: ERROR: <core> [parser/contact/parse_contact.c:59]: contact_parser(): contact_parser(): Error while parsing contacts Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: ERROR: <core> [parser/contact/parse_contact.c:88]: parse_contact(): parse_contact(): Error while parsing Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: ERROR: nathelper [nhelpr_funcs.c:300]: get_contact_uri(): failed to parse Contact body Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: siputils [checks.c:103]: has_totag(): no totag Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: NOTICE: <script>: !------------------------------------>WE HAVE A NEW MESSAGE Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: registrar [lookup.c:181]: *lookup(): contact for [1000] found by address* Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 14==9 && [79.my.server.IP1] == [127.0.0.1] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 14==14 && [79.my.server.IP1] == [79.my.server.IP1] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: NOTICE: <script>: ------------------MESSAGE PROCESSING: *location not found*, storing offline message ID: (sip:1000@85.xx.xx.xx:59287;transport=TLS) Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:605]: m_store(): ------------ start ------------ Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:648]: m_store(): using user id [sip:1000@85.xx.xx.xx:59287;transport=TLS] Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:732]: m_store(): 'From' header: <sip:registrar@my.domain.com> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:849]: m_store(): message stored. T:<sip:1000@my.domain.com> F:< sip:registrar@my.domain.com> Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:858]: m_store(): sending info message. Jul 18 15:50:32 pbx /usr/sbin/kamailio[12600]: DEBUG: msilo [msilo.c:899]: m_store(): contact header found On Thu, Jul 17, 2014 at 11:01 PM, Daniel-Constantin Mierla < miconda(a)gmail.com> wrote: > You should not call m_store() if src_ip==myself -- in this way you avoid > to store notification messages that cannot be delivered. > > Why notifications are not delivered has to be troubleshooted there. Have > you tried also with UDP? > > Cheers, > Daniel > >

1 0

[OT] LVUG, the London VoIP User Group
by Richard Brady 18 Jul '14

18 Jul '14

Hi folks LVUG is a monthly get-together in London to discuss open-source VoIP technology, including Kamailio. If you're based in or near London please join us this coming Tuesday for the third LVUG which will be held in Old Street. The format is a couple short talks followed by beers at the venue or in a nearby pub. Full details here: www.meetup.com/London-VoIP-User-Group-LVUG/ Hope to see you there! Richard -- Richard Brady E: rnbrady(a)gmail.com

1 0

Kamailio presence (NAT, XCAP)
by Radu Vatav 18 Jul '14

18 Jul '14

Hello, I'm just trying to use SIP as a replacement for Skype, nothing more than that. All is working except of presence. Using Jitsi 2.4.4997 (Windows x64) from home behind NAT and kamailio 4.1.4+wheezy on a rented server somewhere. My biggest problem is that I don't know exactly how it is supposed to work. Ok, this is what I understood so far: the clients connect via SIP to the server and can call each other, but for some reason the server can't notify them via SIP of presence changes when they are behind NAT. So the clients also connect to the built in xcap-server, register themselves as presence "watchers" and the xcap-server notifies them via udp/tcp of presence changes of the other users which are watched. All is very nice but it's not working and I don't know where to look for the problem. Which are exactly the steps in communication and what data exactly is there sent which way (when working correctly, which it doesn't) - no clue. I somehow put the configuration together from the 2-3 examples I found on the internet, I don't know if it is ok to attach something to this email, so here it is: https://vatav.cc/kam/kam_cfg.7z Here is the network activity on the server: https://vatav.cc/kam/kam_ngrep.7z So me ("radu") and "alice" are home behind the NAT, one is online and wants to see how the other one goes online and offline. I don't know what exactly to look for, but it doesn't look to me like there is anything sent to one user when the other one goes on-/off-line (there is a lot of "proxy authentication required" which drives me nuts, again I don't know if this is correct or not; Jitsi seems to be connected to the XCAP server, if I type in a wrong address for "XCAP server" Jitsi complains it can't connect). Some other things: - "kamctl online" shows almost real-time who is logged-in, so this works - siremis shows a bunch of users in "watchers list", but none in "active watchers list" - actually when a user goes online it contains the contacts of that user, but only briefly, when I refresh the view, the list is empty again. I don't know if this is how it is supposed to work, anyway, Jitsi doesn't show any change of the status of the user who changed status at all. Any help would be appreciated. It's sad something this simple as I need (basic Skype replacement) is so difficult to get working. Maybe this is why there is relatively low interest in SIP and everybody stays with MS/Google/... because "there isn't something else". Best regards, Radu Vatav

2 1

db_postgres Error
by Rene Montilva 18 Jul '14

18 Jul '14

Hi list i'm config SQLOps Module with postgresql db module or is postgresql driver? this is my kamailio config: #!ifndef DBURL #!define DBURL "postgres://user:ouser@x.x.x.x/kamailio" #!endif sql_query("ca", "select * from xxxxx","ra"); if($dbr(ra=>rows)> 0){ xlog("L_INFO", "saldo:$dbr(ra=>[0,0]) y seg:$dbr(ra=>[0,1]) "); $var(credit) = $dbr(ra=>[0,0]); # 30$ of credit $var(cost_per_sec) = $dbr(ra=>[0,1]); # 1$ per sec $var(i_pulse) = "1"; # 1$ to establish the call $var(f_pulse) = "1"; # 1$ per second sql_result_free ("ra"); if($var(cost_per_sec) > $var(credit)) { sl_send_reply("402","payment required"); exit; } } i got this warning and would like why this happen: WARNING: db_postgres [km_dbase.c:235]: db_postgres_submit_query(): postgres result check failed with code 7 (PGRES_FATAL_ERROR) WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 0, error [SSL SYSCALL error: Connection timed out#012] WARNING: db_postgres [km_dbase.c:239]: db_postgres_submit_query(): postgres query command failed, connection status 1, error [SSL SYSCALL error: EOF detected#012]

2 1

how to m_store to database not just a "body" with MSILO?
by denis kremov 18 Jul '14

18 Jul '14

Hello All, Subj, need you help, How to store this type of message to MSILO database? Content-Type: message/external-body;access-type=URL;URL="https://xxx.com//download/539c28884_3e071c9db68044f687bf.jpg" I modified and recompiled msilo.c to resolve this error: ERROR: msilo [msilo.c:626]: m_store(): body of the message is empty! but after that - m_store just store this information to database: mysql> select * from silo; +----+-------------------------+-------------------------+-----------+-----------+------------+------------+----------+-----------------------+------+------------+--------+--------+ | id | src_addr | dst_addr | username | domain | inc_time | exp_time | snd_time | ctype | body | extra_hdrs | callid | status | +----+-------------------------+-------------------------+-----------+-----------+------------+------------+----------+-----------------------+------+------------+--------+--------+ | 30 | sip:777351300@xxx.in | sip:777581288@xxx.in | 777581288 | xxx.in | 1402479947 | 1403479946 | 0 | message/external-body | | | | 0 | +----+-------------------------+-------------------------+-----------+-----------+------------+------------+----------+-----------------------+------+------------+--------+--------+ 1 row in set (0.00 sec) How can I store to database "access-type=URL" and and URL="xxxx" ? Thanks All! -- denis kremov

3 2

Groups in Kamailio vs Freeswitch
by Peter Villeneuve 18 Jul '14

18 Jul '14

Hi guys, I'm designing a new service for a client and I was wondering what your opinion is of the 2 options I'm considering to separate users on the server. Basically I want to create closed groups where each user can only call and receive calls from members of the same group (single domain install). Initially I was thinking of just forwarding all the invites to FS and using the dialplan to enforce this segregation, but now I realize there is a groups module in kamailio too. Which 2 courses of action do people recommend? I'm more familiar with FS than Kamailio, so it would be easier for me to just forward everything to FS, although that will be more expensive and probably unnecessary if I could do it all in Kamailio with Siremis. Would kamailio's group module do the trick? Are there any tutorials out there I can follow? Cheers, Peter

2 1

trying to conifgure telepresence client software
by Fred Hunter 18 Jul '14

18 Jul '14

Hi, I recently registered with IP tel for the purposes of carrying out SiP based teleconferencing. I am attempting to register my software client, Polycom m100 desktop but am having some difficulty. Can you tell me what I should be entering into the 'registrar server' field, 'domain' and 'registrar server'? [cid:image001.png@01CFA03E.F01888F0] ---------------------------------------------------------------------------- www.puntersouthallgroup.com Punter Southall Group is a trading name of Punter Southall Group Limited. Registered office: 11 Strand, London, WC2N 5HR. Registered in England and Wales No. 04096788. This message and any attachments are confidential. If you have received this message in error please notify the sender and delete this email immediately. Unauthorised dissemination, disclosure, copying or use is prohibited. This email has been scanned for viruses and other malicious items. We reserve the right to monitor the content of all emails sent and received by our systems. Please consider the environment before printing this email. ----------------------------------------------------------------------------

2 1

Re: [SR-Users] how to remove non-digits using dp_translate
by Daniel-Constantin Mierla 18 Jul '14

18 Jul '14

Hmm, I see there is none inside dialplan for the moment. The option would be to do it in config file with a function from textops. Also, check if regex module is a good alternative for you: - http://kamailio.org/docs/modules/stable/modules/regex.html Cheers, Daniel On 16/07/14 16:50, Henry Fernandes wrote: > Can anyone show me how to use the regexp global flag using dp_translate? > -H > > From: Henry Fernandes <henry(a)usinternet.com <mailto:henry@usinternet.com>> > Reply-To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>> > Date: Friday, 11 July, 2014 4:18 PM > To: <miconda(a)gmail.com <mailto:miconda@gmail.com>>, "Kamailio (SER) - > Users Mailing List" <sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org>> > Subject: Re: [SR-Users] how to remove non-digits using dp_translate > > I don't know how to apply the global flag in Kamailio. Can you give > me an example? > -H > > From: Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > Reply-To: <miconda(a)gmail.com <mailto:miconda@gmail.com>>, "Kamailio > (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org > <mailto:sr-users@lists.sip-router.org>> > Date: Thursday, 3 July, 2014 1:47 AM > To: "Kamailio (SER) - Users Mailing List" > <sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org>> > Subject: Re: [SR-Users] how to remove non-digits using dp_translate > > Hello, > > you have to replace [^0-9] with empty, using the global flag. > > Cheers, > Daniel > > On 02/07/14 23:11, Henry Fernandes wrote: >> I'd like to use the dialplan module to remove all non-digits from a >> string. For example, "1-306-555-1234" becomes "13065551234" and >> "(306) 555-1234" becomes "3065551234". >> >> Unfortunately, I can't figure out the regular expressions for >> subst_exp and repl_exp. I can easily replace one instance of a >> non-digit, however I can't figure out how to replace all non-digits. >> >> -H >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla -http://www.asipto.comhttp://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 > <mailto:sr-users@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 <mailto:sr-users@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

Some of the calls drops after 15 minutes + some seconds
by Andras FOGARASI 18 Jul '14

18 Jul '14

Hi, I have a simple kamailio install (2 servers, using location service and a failover node with dispatcher, STUN, clients behind different NATs), without rtpproxy, only peer-to-peer RTP and TURN server if the connection is really messy (it's not relevant here). Signaling is over TLS. Both of the clients are behind NAT. Basically everything works, but some of the calls are dropped after 15 minutes and some seconds, for me it seems the RTP connection is dropped but not sure. I cannot find find out wether it's a kamailio problem or client/NAT problem. Has anyone idea what is going on? Thanks, Andras

3 8

Really need help on kamailio-snmpstats module
by Allen Zhang 17 Jul '14

17 Jul '14

Hi This is a re-post. I'm desperate. What I'm trying to do: Use rpmbuild to build kamailio rpms from source code. We made a lot of changes in the kamailio code. I can't use the rpms from the build service. The plan is to push these changes to kamailio but that's not done, yet. The problem is when I try to install kamailio-snmpstats module from the rpm built, it reports error: Error: Package: kamailio-snmpstats-4.0.0-0.el6.x86_64 (company_name) Requires: net-snmp-agent-libs I've installed net-snmp on the build machine and the machine that needs to install kamailio-snmpstats. I have tried to use the kamailio.spec file from kamailio build server but it didn't solve the problem. Can anyone help, please? Regards, Allen

2 6

Condition Statements - How many levels
by Joel White 16 Jul '14

16 Jul '14

I was wondering how many levels deep in conditional _if_ _else_ statements can go. I need to create a statement that has 4 possible outcomes and need to know if this is possible. I have two variables I want one thing to happen if only the first variable is present Another if only the second variable is present Yet another if they are both present And a fail over if None of the variables are present I do see that is can be done with 3 conditions using _if_ _if else_ _else_ how can I accomplish 4 conditional statements? Thank you in advance Joel

2 2

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