sr-users December 2012

sr-users@lists.kamailio.org

71 participants
113 discussions

by Jon Bonilla

Hi all I have two different kamailio servers running in a single machine. The first one is a sl proxy and the second one acts as stateful proxy/registrar. From time to time the stateless one goes out of pkg mem. It has 8MB (also tried with 16 in another setup). The sl proxy listens the public interface for external connections and the loopback interface for conecting to the stateful proxy. As I have seen since I installed a mem debug kamailio 3.3.2 version there 6 days ago, the internal socket children have grown on mem usage while the external ones remain more or less stable. As example one uf the external udp receivers uses less than 1MB while one of the internal ones uses almost 6MB. I'm attaching the mem dump of one of those internal udp receiver processes via "ngcp-sercmd lb cfg.set_now_int core mem_dump_pkg 25437" command. I see several tens of thousands of lines like these ones: N address=0x7fd1207383c0 frag=0x7fd120738390 size=56 used=1 alloc'd from <core>: parser/parse_param.c: parse_params(539) start check=f0f0f0f0, end check= c0c0c0c0, abcdefed I have no idea about how to continue debugging this. I'm pasting some more info regarding the process list, mem status and kamailio version/flags. Thanks sp1:~# ngcp-sercmd lb ps 25416 attendant 25429 udp receiver child=0 sock=77.77.77.77:5060 25430 udp receiver child=1 sock=77.77.77.77:5060 25431 udp receiver child=2 sock=77.77.77.77:5060 25432 udp receiver child=3 sock=77.77.77.77:5060 25433 udp receiver child=4 sock=77.77.77.77:5060 25434 udp receiver child=5 sock=77.77.77.77:5060 25435 udp receiver child=6 sock=77.77.77.77:5060 25436 udp receiver child=7 sock=77.77.77.77:5060 25437 udp receiver child=0 sock=127.0.0.1:5060 25438 udp receiver child=1 sock=127.0.0.1:5060 25439 udp receiver child=2 sock=127.0.0.1:5060 25440 udp receiver child=3 sock=127.0.0.1:5060 25441 udp receiver child=4 sock=127.0.0.1:5060 25442 udp receiver child=5 sock=127.0.0.1:5060 25443 udp receiver child=6 sock=127.0.0.1:5060 25444 udp receiver child=7 sock=127.0.0.1:5060 25445 slow timer 25446 timer 25447 ctl handler 25448 MI FIFO 25449 tcp receiver (generic) child=0 25450 tcp receiver (generic) child=1 25451 tcp receiver (generic) child=2 25452 tcp receiver (generic) child=3 25453 tcp receiver (generic) child=4 25454 tcp receiver (generic) child=5 25455 tcp receiver (generic) child=6 25456 tcp receiver (generic) child=7 25457 tcp receiver (tcp:127.0.0.1:5060) child=8 25458 tcp receiver (tcp:127.0.0.1:5060) child=9 25459 tcp main process sp1:~# ngcp-sercmd lb pkg.stats { entry: 0 pid: 25416 rank: 0 used: 901136 free: 15417544 real_used: 1359672 } { entry: 1 pid: 25429 rank: 1 used: 909360 free: 15408648 real_used: 1368568 } { entry: 2 pid: 25430 rank: 2 used: 909400 free: 15408608 real_used: 1368608 } { entry: 3 pid: 25431 rank: 3 used: 909352 free: 15408656 real_used: 1368560 } { entry: 4 pid: 25432 rank: 4 used: 909344 free: 15408664 real_used: 1368552 } { entry: 5 pid: 25433 rank: 5 used: 909392 free: 15408712 real_used: 1368504 } { entry: 6 pid: 25434 rank: 6 used: 909344 free: 15408664 real_used: 1368552 } { entry: 7 pid: 25435 rank: 7 used: 909344 free: 15408664 real_used: 1368552 } { entry: 8 pid: 25436 rank: 8 used: 909352 free: 15408656 real_used: 1368560 } { entry: 9 pid: 25437 rank: 9 used: 5884872 free: 2480784 real_used: 14296432 } { entry: 10 pid: 25438 rank: 10 used: 5935272 free: 2359728 real_used: 14417488 } { entry: 11 pid: 25439 rank: 11 used: 5971616 free: 2305912 real_used: 14471304 } { entry: 12 pid: 25440 rank: 12 used: 5821672 free: 2522672 real_used: 14254544 } { entry: 13 pid: 25441 rank: 13 used: 5945584 free: 2310056 real_used: 14467160 } { entry: 14 pid: 25442 rank: 14 used: 5868760 free: 2495264 real_used: 14281952 } { entry: 15 pid: 25443 rank: 15 used: 5903672 free: 2457952 real_used: 14319264 } { entry: 16 pid: 25444 rank: 16 used: 5899208 free: 2453872 real_used: 14323344 } { entry: 17 pid: 25445 rank: -1 used: 983944 free: 15409512 real_used: 2267696 } { entry: 18 pid: 25446 rank: -1 used: 909168 free: 15409512 real_used: 1367704 } { entry: 19 pid: 25447 rank: -2 used: 983664 free: 15407912 real_used: 2267032 } { entry: 20 pid: 0 rank: 0 used: 918424 free: 15400064 real_used: 1377152 } { entry: 21 pid: 25449 rank: 17 used: 989192 free: 15329200 real_used: 1448016 } { entry: 22 pid: 25450 rank: 18 used: 989192 free: 15329200 real_used: 1448016 } { entry: 23 pid: 25451 rank: 19 used: 989192 free: 15329200 real_used: 1448016 } { entry: 24 pid: 25452 rank: 20 used: 989192 free: 15329200 real_used: 1448016 } { entry: 25 pid: 25453 rank: 21 used: 989192 free: 15329200 real_used: 1448016 } { entry: 26 pid: 25454 rank: 22 used: 989192 free: 15329200 real_used: 1448016 } { entry: 27 pid: 25455 rank: 23 used: 989192 free: 15329200 real_used: 1448016 } { entry: 28 pid: 25456 rank: 24 used: 989192 free: 15329200 real_used: 1448016 } { entry: 29 pid: 25457 rank: 25 used: 989192 free: 15329200 real_used: 1448016 } { entry: 30 pid: 25458 rank: 26 used: 989192 free: 15329200 real_used: 1448016 } { entry: 31 pid: 25459 rank: -4 used: 983240 free: 15334096 real_used: 2266992 } sp1:~# kamailio -V version: kamailio 3.3.2 (x86_64/linux) flags: STATS: Off, USE_IPV6, 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 13:08:06 Oct 31 2012 with gcc 4.4.5

11 years, 7 months

Debugging and syslog

by Daniel W. Graham

Hello, Reaching out for some assistance with debugging, I am not seeing any xlog messages in my log file. If there is any other info I didn't include that would be helpful please let me know. Any ideas are appreciated. Using Kamailio 3.3.2 & Cent OS 6 Rsyslog.conf -------------------- *.info;mail.none;authpriv.none;cron.none;local0.none /var/log/messages # Kamailio local0.* -/var/log/kamailio Kamailio CFG ------------------------------ #!define WITH_DEBUG #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif log_facility=LOG_LOCAL0 loadmodule "xlog.so" Example xlog- xlog("L_INFO", "No more gateways \n"); Thanks, Dan-

11 years, 8 months

Send Bye to all calls in one profile

by Camila Troncoso

Hi, I have a kamailio proxy server that´s controlling the limit of calls that clients can make with profile grouping. In other side I also limit clients call duration base on users credit ( DATABASE) and *timeout *feature . Before the calls get answer I calculate how much time the client has left for this call and set the timeout variable. When the time is reached the proxy sends BYEs two both sides of this call. I want when the call timeout, as before explain, all other calls in the same profile end too, because the client has expired his credit. Is there a way to achieve this? Regards, Camila *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

11 years, 8 months

File kamailio_fifo is removed...

by Bruno Bresciani

Hi, Kamailio is running fine, but when I try run again by command line without killing the previous, the kamailio_fifo is removed and functions fifo in kamctl doen't work anymore. Why when I try running kamailio when already exist one, it remove de file kamailio_fifo? I can't find the part of source code that file kamailio_fifo is removed... Someone knows? Best Regards

11 years, 8 months

Wrong ACK packet sent from Cisco gw

by Nawfel Oujdi

Hello! I m facing the same strange behaviour with my AS5300 voice gateway. When the gw is connected directly to PBX everythings works well but when i put a sip proxy forwarding calls between gw and PBX all the calls hangs up after 5 sec (+or -). Looking into the trace sip i realize that gw send a wrong ACK in reply of INVITE , then sip proxy discard it and PBX hangs the call cause he never receive the ACK. ACK sip:79.125.120.12:5060;lr=on;did=ce.3716ea02 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 Route: <sip:911111500@PBX:5060> To: <sip:911111500@sip_proxy>;tag=as7f388e3f Date: Mon, 17 Jan 2011 09:26:36 GMT Call-ID: B6F61A2E-215211E0-802BD462-C4432B89@cisco_gw To work fine , the content of Route header should be in ACK header and viceversa. I tried to compare between the sip trace of a wrong call and a good one (using other cisco gw AS5350 who works well with sip proxy in the same escenario) and i realize that the only difference is the INVITE of wrong case doesn' t send branch number in the via header. INVITE sip:911111500@sip_proxy:5060 SIP/2.0 Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw" From: <sip:911873699@cisco_gw>;tag=65FB8-B18 To: <sip:911111500@sip_proxy> i m using c5300-is-mz.123-26.bin ios version. Anybody understand what is happening in there?? is there any solution?? i ll send more information if it s requested. Thanks in advance. Nawfel Oujdi here is the result of ngrep: U 2011/01/13 15:14:43.791514 cisco_gw:51703 -> sip_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 6. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.791893 sip_server:5060 -> cisco_gw:5060 SIP/2.0 100 Giving a try. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: OpenSIPS (1.6.3-notls (i386/linux)). Content-Length: 0. . U 2011/01/13 15:14:43.791957 sip_server:5060 -> asterisk_server:5060 INVITE sip:911111500@sip_server:5060 SIP/2.0. Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Supported: timer,100rel. Min-SE: 1800. Cisco-Guid: 1295951687-508957152-2608788105-28919687. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. CSeq: 101 INVITE. Max-Forwards: 5. Remote-Party-ID: <sip:911873699@cisco_gw >;party=calling;screen=yes;privacy=off. Timestamp: 1294928083. Contact: <sip:911873699@cisco_gw:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Length: 270. . v=0. o=CiscoSystemsSIP-GW-UserAgent 8894 2421 IN IP4 cisco_gw. s=SIP Call. c=IN IP4 cisco_gw. t=0 0. m=audio 16924 RTP/AVP 18 101. c=IN IP4 cisco_gw. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. U 2011/01/13 15:14:43.792775 asterisk_server:5060 -> sip_server:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Length: 0. . U 2011/01/13 15:14:43.793770 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.794688 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:43.856520 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:43.993417 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. U 2011/01/13 15:14:43.993613 sip_server:5060 -> cisco_gw:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 316. . v=0. o=root 1750021131 1750021131 IN IP4 79.125.41.121. s=Asterisk PBX 1.6.2.13. c=IN IP4 79.125.41.121. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=oldmediaip:asterisk_server. a=oldmediaip:asterisk_server. U 2011/01/13 15:14:44.038774 cisco_gw:57947 -> sip_server:5060 ACK sip:sip_server:5060;lr=on;did=015.864b8107 SIP/2.0. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Date: Thu, 13 Jan 2011 14:14:43 GMT. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. Route: <sip:911111500@asterisk_server:5060>. Max-Forwards: 6. Content-Length: 0. CSeq: 101 ACK. . U 2011/01/13 15:14:44.193431 asterisk_server:5060 -> sip_server:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP sip_server;branch=z9hG4bK3e35.3ed9b366.0;received=sip_server. Via: SIP/2.0/UDP cisco_gw:5060;x-route-tag="cid:Orange@cisco_gw". Record-Route: <sip:sip_server;lr=on;did=015.864b8107>. From: <sip:911873699@cisco_gw>;tag=4F226C8-2DC. To: <sip:911111500@sip_server>;tag=as19e8a82f. Call-ID: 4D776A24-1E5611E0-9B81F289-1B94787@cisco_gw. CSeq: 101 INVITE. Server: Asterisk PBX 1.6.2.13. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO. Supported: replaces, timer. Require: timer. Session-Expires: 1800;refresher=uas. Contact: <sip:911111500@asterisk_server>. Content-Type: application/sdp. Content-Length: 260. . v=0. o=root 1750021131 1750021131 IN IP4 asterisk_server. s=Asterisk PBX 1.6.2.13. c=IN IP4 asterisk_server. t=0 0. m=audio 10798 RTP/AVP 18 101. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. -- ------------------------------ Aviso legal: Este mensaje electrónico está dirigido únicamente a la(s) dirección(es) indicadas anteriormente; el carácter confidencial, personal e intransferible del mismo está protegido legalmente. Cualquier revelación, uso o reenvío no autorizado, completo o en parte, está prohibido. Si ha recibido este mensaje por equivocación, notifíquelo inmediatamente a la persona que lo ha enviado y borre el mensaje original junto con sus ficheros anexos sin leerlo ni grabarlo, total o parcialmente.

11 years, 8 months

Help with Asterisk RT integration

by Jon Morby

Hi I'm trying to integrate a (K) front end cluster with an Asterisk back end cluster and Asterisk RT (legacy system) I've followed the recipe at http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb (with one minor exception …) if (from_uri!=myself && uri!=myself) became if (uri!=myself) as with the original line in place we were able to spoof traffic from 3rd party sites and route out onto the PSTN (which I thought was bad) … anyway … The problem I'm seeing currently is that when a call is passed down a SIP trunk to an end user on the (K) platform we're losing the DNID Asterisk delivers the call to SIP/account/DNID (K) however just tries to deliver the call to DNID@domain which comes back as not found. I've had limited success hand coding aliases into the alias_db however we're still missing the DNID info and having to scrape it from the SIP To field (with a) limited success and b) fears of a support nightmare if we try and move existing customers onto the new platform). If we were to do this for real I'd either have to modify usrloc or try some perl_exec magic I'm guessing … (although it might be possible with a view on the existing sql database … but I'd rather not have to do that if there is a simpler way) I confess I'm struggling to get my head around the config and docs and hoped someone could point me in the right direction? For legacy reasons Asterisk needs to be in the critical path on this particular build … what I'm looking for is a simple recipe and some helpful pointers on how to implement it that will allow enable me to swing (K) into the path between our end user SIP devices and the existing asterisk back ends without losing the ability to deliver hundreds of numbers down a single SIP trunk to a subscriber, and that doesn't require them to make any changes on their end as they will still see the equivalent of SIP:${DNID}@example.com arriving on their PBX This should be simple, but I'm obviously missing something :) Help and pointers gratefully received -- Jon Morby FidoNet - the internet made simple! 10 - 16 Tiller Road, London, E14 8PX tel: 0845 004 3050 / fax: 0845 004 3051

11 years, 8 months

Kamailio 1.5 and Asterisk 1.8 issue (Protocol Error, Unspecified)

by Rowie

Hi, We are having an issue where a phone (snom in particular) cannot make a call through Asterisk. It just hangup and does not allow the call to go through. I am including a a sip trace on this thread to show what is happening within the call. Please see below: Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:41:476 (490 bytes): SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-wowp1kmdy4rl;rport=5060 From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f Call-ID: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 -------------------------------------------------------------------------------- Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:41:481 (387 bytes): ACK sip:vm9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wowp1kmdy4rl;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=6wkdms1r20 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as0b87218f i: 3c26755bf15c-9iq08xqqblo6 CSeq: 4 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 -------------------------------------------------------------------------------- Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:130 (868 bytes): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.30.0.64:5060;received=10.30.0.64;branch=z9hG4bK-5evtiw6dm0po;rport=5060 Record-Route: <sip:10.1.10.80;lr=on> From: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 To: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 INVITE Server: Asterisk PBX 1.8.7.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: <sip:9513261429@10.1.10.83:5060> Content-Type: application/sdp Content-Length: 256 v=0 o=root 1355451627 1355451627 IN IP4 10.1.10.83 s=Asterisk PBX 1.8.7.1 c=IN IP4 10.1.10.83 t=0 0 m=audio 16094 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -------------------------------------------------------------------------------- Sent to udp:10.1.10.80:5060 at 26/10/2011 10:22:42:132 (385 bytes): ACK sip:9513261429@10.1.10.83:5060 SIP/2.0 v: SIP/2.0/UDP 10.30.0.64:5060;branch=z9hG4bK-wszafb7cbzpw;rport Route: <sip:10.1.10.80;lr=on> f: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 t: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 i: 3c2676547a8d-2t5yi6jok1sv CSeq: 2 ACK Max-Forwards: 70 m: <sip:91421@10.30.0.64:5060>;reg-id=1 l: 0 -------------------------------------------------------------------------------- Received from udp:10.1.10.80:5060 at 26/10/2011 10:22:42:232 (503 bytes): BYE sip:91421@10.30.0.64:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.10.80;branch=z9hG4bKe723.bf70c1f4.0 Via: SIP/2.0/UDP 10.1.10.83:5060;branch=z9hG4bK69f53cf1 Max-Forwards: 69 From: <sip:9513261429@ser.gowireless.net;user=phone>;tag=as3f8c0f96 To: "Virgil Menendez" <sip:91421@ser.gowireless.net>;tag=qi3i8ze6z8 Call-ID: 3c2676547a8d-2t5yi6jok1sv CSeq: 102 BYE User-Agent: Asterisk PBX 1.8.7.1 X-Asterisk-HangupCause: Protocol error, unspecified X-Asterisk-HangupCauseCode: 111 Content-Length: 0 -- View this message in context: http://old.nabble.com/Kamailio-1.5-and-Asterisk-1.8-issue-%28Protocol-Error… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

11 years, 9 months

Kamailio and Ubuntu Upstart script

by Moacir Ferreira

As I posted here before, Ubuntu has moved to Upstart as the main engine to start daemons. As a result, Kamailio will fail to start on boot time because it starts before MySQL that has already been moved to the new Upstart daemon starting fashion. I am struggling to get some help from the Ubuntu community explaining how we can match the previous init script behavior when using Upstart but I got no success till now. Anyway, I have created an Upstart script for Kamailio that does most everything that the old script use to do. However, in this script I am making sure MySQL starts before Kamailio. If you have another SQL engine you must change the script to match your SQL engine by changing the “start on started” statement on the script. To use it you have to: A – Create a kamailio.conf file at “/etc/init/” with the contents of the kamailio.conf script in attach, that is the new Upstart stile init script. Please notice that I just adapted the existing script, no news… B – Create a symbolic link at /etc/init.d/ pointing to this kamailio.conf at /etc/init/. C – If you want, you can still have a /etc/default/kamailio file like it used to be and the script will behave like it used to behave when using the default file. Hopefully someone more “influent” at the Ubuntu community may get some help on how to adapt their “start on started” Upstart stanza to behave like we need for multiple possible SQL engines. Cheers! Moacir

11 years, 9 months

pseudovariable values written from app_lua sometimes appear incorrect

by elactrum＠jamailca.com

Hi, I am using app_lua to frequently store some data in htable and avp entries, using sr.pv.sets. When I read the values using sr.pv.get immediately after writing them, they seem correct. However, when I read them later from Lua, I have noticed that sometimes the value is incorrect or nil. Am I doing something wrong? The following code uses rtimer to duplicate the problem on startup, printing the errors to syslog. Tested on version: kamailio 3.3.1 (i386/linux) 2d0f6f ############### # kamailio.cfg ############### loadmodule "pv.so" loadmodule "xlog.so" loadmodule "htable.so" loadmodule "rtimer.so" loadmodule "app_lua" modparam("htable", "htable", "status=>size=14;autoexpire=0;") modparam("rtimer", "timer", "name=ta;interval=10;mode=1") modparam("rtimer", "exec", "timer=ta;route=LUA_TEST") # the following path may need to change depending on filesystem setup modparam("app_lua", "load", "/etc/kamailio/kamailio.lua") route { route(LUA_TEST); } route[LUA_TEST] { $var(lua_test_key) = $(pp{s.toupper}) + "::"; lua_runstring("htable_test([[status=>$var(lua_test_key)]])"); lua_runstring("avp_test([[status=>$var(lua_test_key)]])"); } ---------------- -- kamailio.lua ---------------- -- Test of sr.pv operations in Lua function htable_test (namekey) local pvar_prefix = "$sht("..namekey pvar_test(pvar_prefix) end function avp_test ( name ) local pvar_prefix = "$avp("..name pvar_test(pvar_prefix) end function pvar_test ( pvar_prefix ) for value = 10000, 10050 do local pvar = pvar_prefix..value..")" --sr.log("info", "Lua attempting to set '"..pvar.."' to value '"..value.."'") sr.pv.sets(pvar, value) end for value = 10000, 10050 do -- now we check what was stored local pvar = pvar_prefix..value..")" local checkvar = sr.pv.get(pvar) if ( tostring(checkvar) ~= tostring(value) ) then sr.err(string.format("CHECK: Lua checking pvar '%s' and got value '%s' which is NOT '%s'", tostring(pvar), tostring(checkvar), tostring(value) )) end end end -- end of kamailio.lua Thanks, -Andrew

11 years, 9 months

Kamailio-Java Integration

by ron.kamailio＠mcleodnet.com

I am looking for a way to integrate Java business logic with Kamailio to influence routing decisions, add/remove/modify headers, record accounting-related information, etc. A while back there was a project named WeSIP, but it doesn't appear to active anymore. Is anyone doing something like this now, or any ideas or suggestions? Thanks, Ron

11 years, 9 months

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sr-users December 2012