- sr-users - kamailio.org

Kamailio Load balancing Across Multiple Asterisk Servers -Chanspy
by Jonathan Hunter 14 Jul '15

14 Jul '15

Hi guys, I have a kamailio load balancing requests across 2 Asterisk Servers, and all is working well. Only problem now is when utilizing feature codes to look to listen into calls using the chanspy application I am having issues. I am passing calls to *2222 into Asterisk, where an IVR answers the call, then sends an INVITE back to kamailio to check to see if the extension I want to listen into is in an active call (hash table), and the call will then be forwarded to the Asterisk in question and chanspy applied so it can be listened to. The problem I have is when I send an INVITE from Asterisk Server, and then kamailio sends an INVITE back to the same Asterisk Server I get the following failure in asterisk console; Not accepting call completion offers from call-forward recipient I appreciate its potentially more an Asterisk question but I wondered if anyone had experienced the same issue? I have setup a completely load balanced environment so dont want to use Primary/Secondary Server routing. Thanks Jon

2 4

Nat Detect does not work with GS Phones
by Alberto Sagredo 14 Jul '15

14 Jul '15

I have found an issue detecting NAT of GS 1.4.23 Phones when using Stun on this phones Usual nat_uac_test with numbers 19 and 3 does not seems to detect is behind nat so NATMANAGE is not called Here its some trace. ANy clue how to handle this phones if they activate STUN for example U 80.26.x.x:52768 -> 192.168.0.170:8002 INVITE sip:2@x.x.x.x:8002 SIP/2.0. Via: SIP/2.0/UDP 80.26.x.x:52768;branch=z9hG4bK358742535;rport. From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=1478444456. To: <sip:2@x.x.x.x:8002>. Call-ID: 244257786-52768-56(a)IA.CG.BIE.BCH. CSeq: 550 INVITE. Contact: "Anonymous" <sip:212@80.26.x.x:52768>. X-Grandstream-PBX: true. Max-Forwards: 70. User-Agent: Grandstream GXP2140 1.0.4.23. Privacy: id. P-Preferred-Identity: <sip:212@x.x.x.x:8002>. Supported: replaces, path, timer. Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE. Content-Type: application/sdp. Accept: application/sdp, application/dtmf-relay. Content-Length: 335. . v=0. o=212 8000 8000 IN IP4 80.26.x.x. s=SIP Call. c=IN IP4 80.26.x.x. t=0 0. m=audio 55422 RTP/AVP 0 8 18 9 2 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:9 G722/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19)) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; }

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Possible problem with multiple threads handling same SIP message
by Dirk Teurlings - SIGNET B.V. 14 Jul '15

14 Jul '15

Hello, Running kamailio 4.2.5 for a few weeks now, coming from 4.2.3. Since the upgrade we increased the logging substantially in our routing to get more insight into what is going on. We noticed that in a few cases the log showed a SIP message was being threated by multiple forks, each considering if it was a non-existing call. e.g. Jul 9 13:19:32 son-sbc1-dc1 /usr/sbin/kamailio[17170]: INFO: <script>: 28ca767d5524f7ac095efcd02d6995f5@1.2.3.4:5060 - CB, Checking service restrictions Jul 9 13:19:32 son-sbc1-dc1 /usr/sbin/kamailio[17170]: INFO: <script>: 28ca767d5524f7ac095efcd02d6995f5@1.2.3.4:5060 - CB, Checking international restrictions Jul 9 13:19:32 son-sbc1-dc1 /usr/sbin/kamailio[17171]: INFO: <script>: 28ca767d5524f7ac095efcd02d6995f5@1.2.3.4:5060 - CB, Checking service restrictions Jul 9 13:19:32 son-sbc1-dc1 /usr/sbin/kamailio[17171]: INFO: <script>: 28ca767d5524f7ac095efcd02d6995f5@1.2.3.4:5060 - CB, Checking international restrictions As you can see this shows different threads handling the exact same callid (printed by $ci). And both run the same checks in our routing plan. The main problem is, I'm not able to reproduce this on the testplatform, but only see it happen on the main systems. Also, it doesn't happen that often, in this case there are about maybe 10 similar calls like this over the run of a few weeks. This case show the call being handled by 2 threads, but we've even seen with one call it being handled by 4 threads, all with the same $ci and the same messages, just different PIDs. I've checked and made sure the routing isn't recursively calling itself. But even if that was true, I would imagine the same PID for all the log entries. Running production in debug mode isn't an option, as it will interfere too much with the day-to-day business. So I'm hoping to get a few pointers here maybe, I'm happy to look at the source code more in depth, I just need a little help in the right direction to not get lost in other code. Any help is much appreciated. Cheers, Dirk

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Re: [SR-Users] code coverage in kamailio using gcov
by Daniel-Constantin Mierla 14 Jul '15

14 Jul '15

Hello, On 09/07/15 12:54, Surendra Pullaiah wrote: > > Hello > > > > I am trying to implement code coverage tool in > kamailio using gcov. I have added CFLAGS and LIBS in Makefile.defs and > it got compiled successfully. > > My problem is when I am running kamailio it is saying > lib64/kamailio/libkcore.so.1: undefined symbol: __gcov_merge_add. > > > > Can anyone help me on this. > Maybe those defines and libs need to be added to internal libraries -- see the Makefiles inside the folders of lib/ directory in kamailio source tree. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 0

Kamailio as pass through proxy
by kai.ohnacker＠cbc.de 14 Jul '15

14 Jul '15

Hello, I try to setup a SIP infrastructure with Kamailio server in private network and a pass through Kamailio proxy in the DMZ for NAT handling. (-> TEST.jpg) The pass through proxy uses a new routing algorithm and this works fine for external calls to internal calls. (->Kamailio.cfg.txt) The logic behind is if a packet on external interface arrives, the proxy change the URI and forward the packet to the internal SIP server. Otherwise, the proxy shall forward the packet to external interface if a packet arrives at the internal interface. Because the SIP server has a fix IP, it's easy to route to this server, but the way back is more complicated and I think my algorithm is not working. But the first problem is the routing behavior from the private server if I try to make a call from intern to extern. The server sends the request to the external IP address from the client and the IP address from the pass through Kamailio proxy is not considered. 8(2720) ERROR: <core> [udp_server.c:576]: udp_send(): sendto(sock,0x7f4724dd9238,1328,0,192.168.1.20:5060,16): Network is unreachable(101) 8(2720) ERROR: tm [../../forward.h:208]: msg_send(): udp_send failed 8(2720) WARNING: tm [t_fwd.c:1610]: t_send_branch(): ERROR: t_send_branch: sending request on branch 0 failed 8(2720) exec: *** cfgtrace:request_route=[RELAY] c=[/usr/local/etc/kamailio/kamailio.cfg] l=553 a=24 n=sl_reply_error 8(2720) ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: Unfortunately error on sending to next hop occurred (477/SL) 8(2720) exec: *** cfgtrace:request_route=[RELAY] c=[/usr/local/etc/kamailio/kamailio.cfg] l=555 a=2 n=exit 7(2718) exec: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio.cfg] l=474 a=5 n=route To change the IP address which is used from the private server as the destination address, I added the following lines to the Kamailio.cfg. modparam("registrar", "use_path",1) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received",1) modparam("path", "use_received", 1) But the routing behavior has not changed. (perhaps this configure was not the right) Is there a way to configure the private server that the last IP address from the record-route header is used? Cheers, Kai

2 1

Removing headers in a failure route on 3xx redirect
by Daniel Tryba 13 Jul '15

13 Jul '15

I'm adding and removing headers in the standard route[RELAY], but if there is a redirection involved headers aren't removed. route[RELAY] { #stdstuff route(ADDCHECKSUM); # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { sl_reply_error(); } exit; } route[ADDCHECKSUM] { if($avp(dst_accountcode) || $avp(src_accountcode)) { #some magic remove_hf("X-rand"); remove_hf("X-csum"); append_hf("X-rand: $var(rand)\r\n"); append_hf("X-csum: $var(checksum)\r\n"); } return; } This works well unless the destination is a redirecting sip server. failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } if (t_check_status("3[0-9][0-9]")) { #... route(RELAY); exit; } -user calls destination. -dispatcher sets destination is send to redirectserver and routes to RELAY, addchecksums add some headers called from relay. -redirectserver replies 302. -failure route triggers and ends with a route to RELAY. Any existing old values should be removed and the newer ones added. But end the result is that I have two sets op X-rand/csum headers for the INVITE to the redirection contact. How do I get rid of every X-csum/rand header except the last added ones? Version tested: 4.2.5.

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Pipelimit
by Fred Posner 13 Jul '15

13 Jul '15

I know in ratelimit, you can define a method within queues... for example, modparam("ratelimit", "queue", "3:INVITE") to match on INVITE method. With pipelimit, is the method called from the config, such as: if(is_method("INVITE")) { # Checking limit on INVITE if (!pl_check("$au", "traildrop", "$var(limita)")) { pl_drop(); exit; } } if(is_method("REGISTER")) { # Checking limit on REGISTER if (!pl_check("$au", "traildrop", "$var(limitb)")) { pl_drop(); exit; } } -- Fred Posner The Palner Group, Inc. http://www.palner.com (web) +1-503-914-0999 (direct)

2 1

msilo delete rows
by Evgeny Zhavoronkov 13 Jul '15

13 Jul '15

Hi All, I'm fighting with msilo module now. I recive an offline message, but msilo gets 408 status and think that a message wasn't sent succesfully. According to the log, the error somewhere in t_relay, but I couldn't find where exactly. Therefore, msilo doesn't delete rows from silo table in database and everytime the user reregisters he gets offline messages again and again. -- lifayk e: lifayk(a)gmail.com pgp: 0x84CAC40B

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Does Kamailio support Cx interface Diameter over SCTP, somebody tested it?
by Jan Devai 13 Jul '15

13 Jul '15

Hi Team, Does anybody know if Kamailio support Cx interface Diameter over SCTP, somebody tested it? Thank you Jan

2 1

working config
by lifayk 13 Jul '15

13 Jul '15

Hello All! Can somebody share a working/example config with biderectional nathelper, tcp transport and push notifications (tsilo) for Kamailio >=4.3? Thanks in advance! -- lifayk e: lifayk(a)gmail.com pgp: 0x84CAC40B

2 1

sip: uri to tel: uri
by manuel espinoza 13 Jul '15

13 Jul '15

Hi, The arch looks like this SBC1 <-------- SIP-Router (Kamailio) <------------- Asterisk Asterisk sends an invite to kamailio using "sip:" as request uri scheme, but SBC1 expects that invite to arrive using tel:, is there a way to change this in kamailio and forward that invite? Something like: $ru = "tel: +" + $rU; Thanks!!!

2 1

sctp on Centos 7
by micunek＠gmail.com 13 Jul '15

13 Jul '15

Hi, I would like to enable and use sctp in kamailio but I do not know exactly how to do it. It seems that documentation is a little bit inconsistent. 1. I have Centos 7 and kamailio installed from repositories. 2. I set loadmodule "sctp" in kamailio.cfg 3. I got this error: Jul 9 08:08:54 kamailio: : sctp [sctp_mod.c:99]: mod_register(): sctp core support not enabled Jul 9 08:08:54 kamailio: ERROR: <core> [sr_module.c:600]: load_module(): /usr/lib64/kamailio/modules/sctp.so: mod_register failed Jul 9 08:08:54 cw-aio kamailio: : <core> [cfg.y:3432]: yyerror_at(): parse error in config file /etc/kamailio_pcscf/kamailio.cfg, line 230, column 12-17: failed to load module 4. I checked mod_register() function and found out that USE_SCTP should be set. 5. kamailio -V did not show USE_SCTP 6. Next I tried to compile kamailio sources from 4.3 branch make include_modules="db_mysql tls sctp" SCTP=1 cfg make all 7. Still the same problem "mod_register(): sctp core support not enabled" 8. Then I added USE_SCTP to C_DEFS and compile again. This attempt was successfull. I was able to see USE_SCTP in kamailio -V output. Questions: 1. What is the right procedure to enable sctp in kamailio? 2. Why the -DUSE_SCTP was not used when SCTP=1 was used. According to documentation: "The core Makefile variable SCTP must be set to 1 (which is by default set to 1 in Makefile.defs) and sources compiled with -DUSE_SCTP (automatically set when SCTP=1). In other words, if core Makefiles are not changed and SCTP variable is not overwritten from command line, then the SCTP support in core is enabled." Regards, Peter

2 1

nat_uac_test question
by Kristian Kielhofner 13 Jul '15

13 Jul '15

Hello, I'm testing some basic flags with nat_uac_test. I have the following (rudimentary) script: route[NATDETECT] { #!ifdef WITH_NAT if (has_body("application/sdp")) { if (nat_uac_test("18")) { xlog("SCRIPT: SDP NAT Detected\n"); setflag(6); } } else { if (nat_uac_test("2")) { xlog("SCRIPT: Other NAT Detected\n"); setflag(6); } } if (isflagset(6)) { xlog("SCRIPT: NAT detected\n"); force_rport(); if (is_method("REGISTER")) { fix_nated_register(); } else { if(is_first_hop()) set_contact_alias(); } setflag(FLT_NATS); } #!endif return; } When presented with this REGISTER message (ngrep output, so first line is IP source address and port): U 192.168.113.17:6060 -> 192.168.113.11:5060 REGISTER sip:192.168.113.11;transport=udp SIP/2.0. Via: SIP/2.0/UDP 192.168.113.17:6060;branch=z9hG4bKOzvuzaIa;rport. From: <sip:user_1@192.168.113.11>;tag=3F785909-559E7E9F0009321B-B5BFFB40. To: <sip:user_1@192.168.113.11>. CSeq: 10 REGISTER. Call-ID: 1C193C6B-559E7E9F00093255-B5BFFB40. Max-Forwards: 70. User-Agent: FreeSWITCH. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Contact: <sip:gw+sip0@10.40.47.190:5080;transport=udp;gw=sip0>;expires=3600. Content-Length: 0. . U 192.168.113.11:5060 -> 192.168.113.17:6060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.113.17:6060;branch=z9hG4bKOzvuzaIa;rport=6060;received=192.168.113.17. From: <sip:user_1@192.168.113.11>;tag=3F785909-559E7E9F0009321B-B5BFFB40. To: <sip:user_1@192.168.113.11>;tag=b27e1a1d33761e85846fc98f5f3a7e58.1045. CSeq: 10 REGISTER. Call-ID: 1C193C6B-559E7E9F00093255-B5BFFB40. Contact: <sip:gw+sip0@10.40.47.190:5080;transport=udp;gw=sip0>;expires=3600;received="sip:192.168.113.17:6060". Server: Kamailio. Content-Length: 0. . Log output: Jul 9 10:00:41 kamailio /usr/sbin/kamailio[3886]: ERROR: <script>: SCRIPT: Other NAT Detected Jul 9 10:00:41 kamailio /usr/sbin/kamailio[3886]: ERROR: <script>: SCRIPT: NAT detected Log output, the presence of the added "received" parameter in the returned Contact, etc show that NAT was detected. However, the IP source and port match the one presented in the Via header. What am I missing here? Thanks! -- Kristian Kielhofner

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pua_publish MWI message via xmlrpc
by Asgaroth 10 Jul '15

10 Jul '15

Hi All, I am having difficulty getting pua_publish to issue a NOTIFY when I send an XMLRPC command to generate a PUBLISH into the presence server. I have been reading the pua_mi module docs, but I cannot see where I am going wrong with the following xmlrpc command, please would someone point me in the correct direction. I have an XML file which is the basis of the XMLRPC request via curl, I have attached it to this email for you to view, please would you take a look and let me know where I am going wrong, I suspect it is my message body as the pua module complains about the message body being NULL: Jul 10 17:27:40 reg01 /usr/sbin/kamailio[7736]: ERROR: pua [add_events.c:125]: pres_process_body(): while parsing xml memory Jul 10 17:27:40 reg01 /usr/sbin/kamailio[7736]: ERROR: pua [send_publish.c:650]: send_publish(): while processing body Jul 10 17:27:40 reg01 /usr/sbin/kamailio[7736]: ERROR: pua [send_publish.c:652]: send_publish(): NULL body Jul 10 17:27:40 reg01 /usr/sbin/kamailio[7736]: ERROR: pua_mi [mi_func.c:290]: mi_pua_publish(): sending publish failed Kamailio verion is "version: kamailio 4.3.0 (x86_64/linux) c6aa95" I can issue other XMLRPC commands, urloc lookups method lists and those work, I just cant see where I am going wrong with this one. Any pointers would be greatly appreciated. Thanks

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dmq/dmq_userloc crash on startup (kamailio 4.3.0)
by Asgaroth 10 Jul '15

10 Jul '15

Hi All, Another thing I noticed while playing with kamailio and dmq/dmq_usrloc is that 7 out of 10 times the kamailio process crashes when I initially start it up. Any tips/requests for further debugging would be greatly appreciated. Kamailio version: version: kamailio 4.3.0 (x86_64/linux) c6aa95 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: c6aa95 compiled on 16:14:27 Jun 23 2015 with gcc 4.4.7 I start kamailio as follows, from commandline: /usr/sbin/kamailio -P /var/run/kamailio_registrar01/kamailio.pid -f /shared/kamailio/etc/kamailio_registrar.cfg -l "udp:(10.6.0.173, 10.7.0.173)" -m 1024 -M64 -A RESOURCE_REGISTRAR01 -l tcp:10.6.0.173:80 -u root -g root -ddd DMQ related settings are as follows: # grep -i dmq module_definitions.cfg loadmodule "dmq.so" loadmodule "dmq_usrloc.so" # ----- dmq params ----- modparam("dmq", "server_address", DMQ_ADDRESS) modparam("dmq", "notification_address", "sip:sip-registrars.internal_domain.com:5060") modparam("dmq", "multi_notify", 1) modparam("dmq", "num_workers", 4) # ----- dmq_usrloc params ----- modparam("dmq_usrloc", "enable", 1) DMQ_ADDRESS is defined as "sip:10.6.0.173:5060" in this servers case. sip-registrars.internal_domain.com resolves to the following address list 10.6.0.173, 10.6.0.174, 10.6.0.175 Here is the debug log of the crash: Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [worker.c:82]: worker_loop(): dmq_worker [0 27445] getting lock Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [worker.c:84]: worker_loop(): dmq_worker [0 27445] lock acquired Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=66fabe782d71b7a9842d4a6a5eb93c4d-1de2 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [notification_peer.c:454]: dmq_notification_callback(): dmq triggered from dmq_notification_callback Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [dmqnode.c:153]: build_dmq_node(): build_dmq_node sip:10.6.0.173:5060;status=active with private memory Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [dmqnode.c:153]: build_dmq_node(): build_dmq_node sip:10.6.0.175:5060;status=active with private memory Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [dmqnode.c:153]: build_dmq_node(): build_dmq_node sip:10.6.0.174:5060;status=active with private memory Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [notification_peer.c:468]: dmq_notification_callback(): received 0 new or changed nodes Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [notification_peer.c:528]: build_notification_body(): body_len = 1024 - clen = 0 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [notification_peer.c:528]: build_notification_body(): body_len = 1024 - clen = 35 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [notification_peer.c:528]: build_notification_body(): body_len = 1024 - clen = 70 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq_usrloc [usrloc_sync.c:388]: usrloc_dmq_request_sync(): requesting sync from dmq peers Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq_usrloc [usrloc_sync.c:404]: usrloc_dmq_request_sync(): sending serialized data {"action":3} Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq_usrloc [usrloc_sync.c:226]: usrloc_dmq_send(): sending dmq broadcast... Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [socket_info.c:563]: grep_sock_info(): checking if host==us: 10==10 && [10.6.0.173] == [10.6.0.173] Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [socket_info.c:566]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: tm [uac.c:249]: t_uac_prepare(): DEBUG:tm:t_uac: next_hop=<sip:usrloc@10.6.0.174:5060> Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: tm [uac.c:150]: dlg2hash(): DEBUG: dlg2hash: 24685 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [socket_info.c:563]: grep_sock_info(): checking if host==us: 10==10 && [10.6.0.173] == [10.6.0.173] Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [socket_info.c:566]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: tm [uac.c:249]: t_uac_prepare(): DEBUG:tm:t_uac: next_hop=<sip:usrloc@10.6.0.175:5060> Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: tm [uac.c:150]: dlg2hash(): DEBUG: dlg2hash: 24686 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [dmq_funcs.c:163]: bcast_dmq_message(): skipping node sip:10.6.0.173:5060 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: sl [sl.c:280]: send_reply(): reply in stateless mode (sl) Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [msg_translator.c:158]: check_via_address(): (10.6.0.174, 10.6.0.174, 0) Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq [worker.c:134]: worker_loop(): sent reply Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=66fabe782d71b7a9842d4a6a5eb93c4d-1ac2 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq_usrloc [usrloc_sync.c:250]: usrloc_dmq_handle_msg(): dmq message received from sip:usrloc@10.6.0.174:5060 Jul 1 21:04:48 bfievkrl01 /usr/sbin/kamailio[27445]: DEBUG: dmq_usrloc [usrloc_sync.c:360]: usrloc_dmq_handle_msg(): Received DMQ_SYNC. Sending all contacts... And here is the backtrace from the core file: # gdb /usr/sbin/kamailio /core.27445 GNU gdb (GDB) Red Hat Enterprise Linux (7.2-75.el6) Copyright (C) 2010 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html> This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type "show copying" and "show warranty" for details. This GDB was configured as "x86_64-redhat-linux-gnu". For bug reporting instructions, please see: <http://www.gnu.org/software/gdb/bugs/>... Reading symbols from /usr/sbin/kamailio...(no debugging symbols found)...done. [New Thread 27445] Reading symbols from /lib64/libdl.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libdl.so.2 Reading symbols from /lib64/libresolv.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libresolv.so.2 Reading symbols from /lib64/libc.so.6...(no debugging symbols found)...done. Loaded symbols for /lib64/libc.so.6 Reading symbols from /lib64/ld-linux-x86-64.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/ld-linux-x86-64.so.2 Reading symbols from /usr/lib64/kamailio/modules/db_unixodbc.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/db_unixodbc.so Reading symbols from /usr/lib64/libodbc.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libodbc.so.2 Reading symbols from /usr/lib64/kamailio/libsrdb1.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/libsrdb1.so.1 Reading symbols from /usr/lib64/kamailio/libkcore.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/libkcore.so.1 Reading symbols from /usr/lib64/libltdl.so.7...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libltdl.so.7 Reading symbols from /lib64/libpthread.so.0...(no debugging symbols found)...done. [Thread debugging using libthread_db enabled] Loaded symbols for /lib64/libpthread.so.0 Reading symbols from /usr/lib64/kamailio/modules/mi_fifo.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/mi_fifo.so Reading symbols from /usr/lib64/kamailio/libkmi.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/libkmi.so.1 Reading symbols from /usr/lib64/kamailio/modules/tm.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/tm.so Reading symbols from /usr/lib64/kamailio/modules/sl.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/sl.so Reading symbols from /usr/lib64/kamailio/modules/pv.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/pv.so Reading symbols from /usr/lib64/kamailio/libsrutils.so.1...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/libsrutils.so.1 Reading symbols from /lib64/libm.so.6...(no debugging symbols found)...done. Loaded symbols for /lib64/libm.so.6 Reading symbols from /usr/lib64/kamailio/modules/kex.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/kex.so Reading symbols from /usr/lib64/kamailio/modules/usrloc.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/usrloc.so Reading symbols from /usr/lib64/kamailio/modules/registrar.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/registrar.so Reading symbols from /usr/lib64/kamailio/modules/xlog.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/xlog.so Reading symbols from /usr/lib64/kamailio/modules/auth.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/auth.so Reading symbols from /usr/lib64/kamailio/modules/auth_db.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/auth_db.so Reading symbols from /usr/lib64/kamailio/modules/rr.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/rr.so Reading symbols from /usr/lib64/kamailio/modules/path.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/path.so Reading symbols from /usr/lib64/kamailio/modules/nathelper.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/nathelper.so Reading symbols from /usr/lib64/kamailio/modules/siputils.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/siputils.so Reading symbols from /usr/lib64/kamailio/modules/textops.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/textops.so Reading symbols from /usr/lib64/kamailio/modules/xmlrpc.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/xmlrpc.so Reading symbols from /usr/lib64/libxml2.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libxml2.so.2 Reading symbols from /lib64/libz.so.1...(no debugging symbols found)...done. Loaded symbols for /lib64/libz.so.1 Reading symbols from /usr/lib64/kamailio/modules/ctl.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/ctl.so Reading symbols from /usr/lib64/kamailio/modules/cfg_rpc.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/cfg_rpc.so Reading symbols from /usr/lib64/kamailio/modules/dmq.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/dmq.so Reading symbols from /usr/lib64/kamailio/modules/dmq_usrloc.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/kamailio/modules/dmq_usrloc.so Reading symbols from /lib64/libnss_files.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libnss_files.so.2 Reading symbols from /usr/lib64/gconv/ISO8859-1.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/gconv/ISO8859-1.so Reading symbols from /usr/lib64/psqlodbc.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/psqlodbc.so Reading symbols from /usr/lib64/libssl.so.10...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libssl.so.10 Reading symbols from /usr/lib64/libpq.so.5...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libpq.so.5 Reading symbols from /usr/lib64/libodbcinst.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libodbcinst.so.2 Reading symbols from /lib64/libgssapi_krb5.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libgssapi_krb5.so.2 Reading symbols from /lib64/libkrb5.so.3...(no debugging symbols found)...done. Loaded symbols for /lib64/libkrb5.so.3 Reading symbols from /lib64/libcom_err.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libcom_err.so.2 Reading symbols from /lib64/libk5crypto.so.3...(no debugging symbols found)...done. Loaded symbols for /lib64/libk5crypto.so.3 Reading symbols from /usr/lib64/libcrypto.so.10...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libcrypto.so.10 Reading symbols from /lib64/libcrypt.so.1...(no debugging symbols found)...done. Loaded symbols for /lib64/libcrypt.so.1 Reading symbols from /lib64/libldap_r-2.4.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libldap_r-2.4.so.2 Reading symbols from /lib64/libkrb5support.so.0...(no debugging symbols found)...done. Loaded symbols for /lib64/libkrb5support.so.0 Reading symbols from /lib64/libkeyutils.so.1...(no debugging symbols found)...done. Loaded symbols for /lib64/libkeyutils.so.1 Reading symbols from /lib64/libfreebl3.so...(no debugging symbols found)...done. Loaded symbols for /lib64/libfreebl3.so Reading symbols from /lib64/liblber-2.4.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/liblber-2.4.so.2 Reading symbols from /usr/lib64/libsasl2.so.2...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libsasl2.so.2 Reading symbols from /usr/lib64/libssl3.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libssl3.so Reading symbols from /usr/lib64/libsmime3.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libsmime3.so Reading symbols from /usr/lib64/libnss3.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libnss3.so Reading symbols from /usr/lib64/libnssutil3.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib64/libnssutil3.so Reading symbols from /lib64/libplds4.so...(no debugging symbols found)...done. Loaded symbols for /lib64/libplds4.so Reading symbols from /lib64/libplc4.so...(no debugging symbols found)...done. Loaded symbols for /lib64/libplc4.so Reading symbols from /lib64/libnspr4.so...(no debugging symbols found)...done. Loaded symbols for /lib64/libnspr4.so Reading symbols from /lib64/libselinux.so.1...(no debugging symbols found)...done. Loaded symbols for /lib64/libselinux.so.1 Reading symbols from /lib64/librt.so.1...(no debugging symbols found)...done. Loaded symbols for /lib64/librt.so.1 Reading symbols from /lib64/libnss_dns.so.2...(no debugging symbols found)...done. Loaded symbols for /lib64/libnss_dns.so.2 Core was generated by `/usr/sbin/kamailio -P /var/run/kamailio_registrar01/kamailio.pid -f /shared/kam'. Program terminated with signal 11, Segmentation fault. #0 0x00007fdad84d76a5 in memcpy () from /lib64/libc.so.6 Missing separate debuginfos, use: debuginfo-install kamailio-4.3.0-0.el6.bf.x86_64 (gdb) bt #0 0x00007fdad84d76a5 in memcpy () from /lib64/libc.so.6 #1 0x00007fdacfeff03d in usrloc_get_all_ucontact () from /usr/lib64/kamailio/modules/dmq_usrloc.so #2 0x00007fdacff02dfd in usrloc_dmq_handle_msg () from /usr/lib64/kamailio/modules/dmq_usrloc.so #3 0x00007fdad012d4a7 in worker_loop () from /usr/lib64/kamailio/modules/dmq.so #4 0x00007fdad011232c in ?? () from /usr/lib64/kamailio/modules/dmq.so #5 0x0000000000592675 in ?? () #6 0x00000000005923b0 in ?? () #7 0x000000000059297f in init_child () #8 0x00000000004a7680 in main_loop () #9 0x00000000004acd36 in main () (gdb) ^C(gdb) Quit

2 4

dialog: use keepalive & timeout features together
by Andrey Utkin 09 Jul '15

09 Jul '15

Would be nice if it would be possible to - make timer of "default_timeout" reset to 0 when keep-alive reply is received from remote part of call leg, so that the timeout expires (triggering call termination) only if some call party gets offline (we don't want just to limit the duration of normal call, even if call parties don't send any new SIP requests); OR - make unresponsiveness of call parties to keep-alive messages (controlled with ka_timer, ka_interval parameters) lead to some action, like call termination. Please advise if I miss something, and how to implement this better. Thanks in advance. -- Andrey Utkin

2 3

Re: [SR-Users] Kamailio deployment
by Daniel Tryba 09 Jul '15

09 Jul '15

On Thursday 09 July 2015 17:46:22 Srujan Kumar K wrote: > I am connecting my kamailio server to MSC for my solution which has both > inbound and outbound. ... > Kamailio server IP : 172.22.14.12 > MSC IP : 172.28.0.68 Why does you kamailio config bind to the msc ipadress?

1 0

[Kamailio-Users] HA+DRBD setup
by Siddhardha Garige 09 Jul '15

09 Jul '15

Hello all, Just wondering if anyone know any tutorial on setting up HA+DRBD solution for kamailio. Especially creating partitions, DRBD devices and mount points. Thanks in advance, -Sid "May the light be with you." ______________________________________________ Siddhardha Garige www.luminepixels.com

11 24

Kamailio deployment
by Srujan Kumar K 09 Jul '15

09 Jul '15

Hai, I am connecting my kamailio server to MSC for my solution which has both inbound and outbound. The SIP trunk is not UP and we don't see any packets transmitted between the 2 nodes. We are also trying to load SIPI/SIPT module Kamailio server IP : 172.22.14.12 MSC IP : 172.28.0.68 I have also attached configuration file with this mail. Requesting your help on this. Regards ~Srujan

1 0

Moving from haproxy to db_cluster
by Jan Hazenberg 09 Jul '15

09 Jul '15

Hi, I have a question about the db_cluster module. We are currently using a galera mysql cluster from severalnines, in this setup haproxy is installed on the kamailio nodes handling the database failover. The question is if a database node fails and the TCP connection breaks the sql query will probably fail when using haproxy since it is not aware of the query. Will the db_cluster module send the query to the next available mysql node in the cluster without a failing query? If thats the case i should not notice the failing dbnode in my kamailio script. In other words, will there be a advantage in using the db_cluster module instead of haproxy? Thanks, Jan Hazenberg

2 3

code coverage in kamailio using gcov
by Surendra Pullaiah 09 Jul '15

09 Jul '15

Hello I am trying to implement code coverage tool in kamailio using gcov. I have added CFLAGS and LIBS in Makefile.defs and it got compiled successfully. My problem is when I am running kamailio it is saying lib64/kamailio/libkcore.so.1: undefined symbol: __gcov_merge_add. Can anyone help me on this. Regards Surendra

1 0

Function sdp_remove_codecs_by_id seems to be not working
by José Seabra 09 Jul '15

09 Jul '15

Hello, I'm using the function sdp_remove_codecs_by_id from sdpops module in order to remove some codecs in INVITE request before send it to freeswitch, but the function doesn't remove the codec, and it doesn't give any error message. I'm using this function in request route. Kamailio version is 4.2.2. INVITE that kamailio receives from phone: INVITE sip:401@teste.d <sip%3A401(a)teste.itcenter.com.pt>emo.pt;user=phone SIP/2.0 Record-Route: <sip:10.0.20.102:5062 ;r2=on;lr=on;ftag=oztyflbzbx;nat=yes;lb=yes> Record-Route: <sip:100.64.250.4;r2=on;lr=on;ftag=oztyflbzbx;nat=yes;lb=yes> Via: SIP/2.0/UDP 10.0.20.102:5062 ;branch=z9hG4bKecf3.3ff3f7e77d2abc0fd3f74c61eeb68a0b.0 Via: SIP/2.0/UDP 192.168.10.147:5060 ;received=100.64.250.254;branch=z9hG4bK-f0jm82qox75w;rport=5060 From: "301" <sip:301@teste.demo.pt <sip%3A301(a)teste.itcenter.com.pt> >;tag=oztyflbzbx To: <sip:401@teste.demo.pt <sip%3A401(a)teste.itcenter.com.pt>;user=phone> Call-ID: 3c3a58a25d63-ghfc5xdg1sn0 CSeq: 1 INVITE Max-Forwards: 69 Contact: <sip:301@192.168.10.147:5060 ;alias=100.64.250.254~5060~1;line=c1r2c8u6>;reg-id=1 X-Serialnumber: 000413262FA0 P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom370/8.4.35 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Call-Info: <sip:teste.demo.pt <http://teste.itcenter.com.pt> >;appearance-index=1 Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 391 v=0 o=root 24935823 24935823 IN IP4 192.168.10.147 s=call c=IN IP4 192.168.10.147 t=0 0 m=audio 19410 RTP/AVP 0 8 9 99 3 18 4 101 a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv INVITE that kamailio send to freeswitch after execute sdp_remove_codecs_by_id("18"): INVITE sip:401@teste.demo.pt;user=phone SIP/2.0. Record-Route: <sip:10.0.20.100;lr=on;ftag=zvjgcz9zs9;proxy=yes;did=441.0eb2>. Record-Route: <sip:10.0.20.102:5062 ;r2=on;lr=on;ftag=zvjgcz9zs9;nat=yes;lb=yes>. Record-Route: <sip:100.64.250.4;r2=on;lr=on;ftag=zvjgcz9zs9;nat=yes;lb=yes>. Via: SIP/2.0/UDP 10.0.20.100;branch=z9hG4bK8711.bb31396197409170b2c1bd05b24e7f36.0. Via: SIP/2.0/UDP 10.0.20.102:5062 ;branch=z9hG4bK8711.07ffcc13fb96f90f6b4dbe4b2dfd0fa5.0. Via: SIP/2.0/UDP 192.168.10.147:5060 ;received=100.64.250.254;branch=z9hG4bK-aq7e0puz8p6o;rport=5060. From: "301" <sip:301@teste.demo.pt>;tag=zvjgcz9zs9. To: <sip:401@teste.demo.pt;user=phone>. Call-ID: 3c3a7c84e065-pr2hm0uk9yfz. CSeq: 2 INVITE. Max-Forwards: 68. Contact: <sip:301@192.168.10.147:5060 ;alias=100.64.250.254~5060~1;line=ttnfv9c7>;reg-id=1. X-Serialnumber: 000413262FA0. P-Key-Flags: resolution="31x13", keys="4". User-Agent: snom370/8.4.35. Accept: application/sdp. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE. Allow-Events: talk, hold, refer, call-info. Supported: timer, 100rel, replaces, from-change. Call-Info: <sip:teste.itcenter.com.pt>;appearance-index=1. Session-Expires: 3600;refresher=uas. Min-SE: 90. Content-Type: application/sdp. Content-Length: 403. . v=0. o=root 228603317 228603317 IN IP4 100.64.250.4. s=call. c=IN IP4 100.64.250.4. t=0 0. m=audio 49404 RTP/AVP 0 8 9 99 3 18 4 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:9 G722/8000. a=rtpmap:99 G726-32/8000. a=rtpmap:3 GSM/8000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:4 G723/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. a=rtcp:49405. SDP body has no changes related with codecs. Anyone call help please. Thank you BR José Seabra -- Cumprimentos José Seabra

3 6

Textops and Multipart Body - adding ISUP segfaults Kamailio
by andre second 09 Jul '15

09 Jul '15

Hi, I am using Kamailio version: kamailio 4.3.0 (x86_64/linux) c6aa95 on CentOS 6 I am trying to encapsulate ISUP in the INVITE: ...if(has_body("application/sdp")){ set_body_multipart(); if(msg_apply_changes()) { $var(acm) = "7e Od 04 55 75 69 20 4d 61 6b 65 43 61 6c 6c"; append_body_part("$var(acm)","application/isup; version=itu-t92+","signal; handling=optional"); xlog("L_INFO", "ISUP Changes Applied Succesfully"); }} ...And getting the following error: Jul 7 14:59:12 ukrcld1000 kamailio: DEBUG: <core> [msg_translator.c:1691]: get_boundary(): boundary is <--unique-boundary-1>...Jul 7 14:59:12 ukrcld1000 kamailio: DEBUG: <core> [msg_translator.c:1808]: check_boundaries(): last bondary without -- at the end Jul 7 14:59:12 ukrcld1000 kamailio: DEBUG: <core> [msg_translator.c:1612]: replace_body(): old size body[365] actual[534]Jul 7 14:59:12 ukrcld1000 kamailio: CRITICAL: <core> [data_lump.c:463]: free_lump(): non free-able lump: 0x7f7a33d5ae40 flags=1 and then Kamailio crashes. Can you please recommend? Thanks,Andrei

2 3

usleep with variable
by Antanas Masevicius 09 Jul '15

09 Jul '15

Hello, been struggling to make a random sleep with cfgutils functions. ex: $avp(i:444) = 123; usleep("$avp(i:444)"); gives an error: ERROR: <core> [route.c:1189]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio-sp1new.cfg:723 Kamailio version: 4.3 If usleep accepts only numbers, is there any other way to sleep by passing variable? best regards, Antanas -- # <#> Antanas Masevičius Technikos Direktorius UAB Nacionalinis telekomunikacijų tinklas Panerių g. 51, LT-03160 Vilnius *Tel.* 8 5 2056000 *Tiesiog. tel.* 8 700 00031 *Mob. tel.* 8 685 77730 *Faks.* 8 700 00034 *El. p.:* antanas.masevicius(a)ntt.lt <#> # www.ntt.lt <http://www.ntt.lt> | www.spykas.com <http://www.spykas.com>

1 1

MetaSwitch ignores Contact: header
by Markus 08 Jul '15

08 Jul '15

Dear list, I'm running kamailio-4.2.3-3.1.x86_64 in a very simple setup: one Asterisk server behind kamailio. But I'm using dispatcher, the other servers are just down currently. :) Different vendors route DIDs to kamailio's IP and kamailio handles the communication towards Asterisk. Now I have a new vendor who uses MetaSwitch, and there is a problem. All calls get dropped with a BYE from the Asterisk side after 32 seconds because Asterisk never receives the ACK, because kamailio doesn't forward it, because MetaSwitch sends it as ACK sip:number@KAMAILIO-IP instead of ACK sip:number@ASTERISK-IP That means MetaSwitch ignores what we tell it in the Contact: header. Is this MetaSwitch'es "fault" or kamailio's? All other vendors work fine without problems. Therefore I have compared an "old" vendor and the new one. Here is the result that shows the difference: Old vendor - works: OLDVENDOR-IP.5060 > KAMAILIO-IP.5060: INVITE sip:number@KAMAILIO-IP KAMAILIO-IP.5060 > OLDVENDOR-IP.5060: SIP/2.0 100 trying KAMAILIO-IP.5060 > ASTERISK-IP.5060: INVITE sip:number@KAMAILIO-IP ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 100 Trying ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 200 OK KAMAILIO-IP.5060 > OLDVENDOR-IP.5060: SIP/2.0 200 OK OLDVENDOR-IP.5060 > KAMAILIO-IP.5060: ACK sip:number@ASTERISK-IP:5060 KAMAILIO-IP.5060 > ASTERISK-IP.5060: ACK sip:number@ASTERISK-IP:5060 New vendor with MetaSwitch - doesn't work: NEWVENDOR-IP.5060 > KAMAILIO-IP.5060: INVITE sip:number@KAMAILIO-IP:5060 KAMAILIO-IP.5060 > NEWVENDOR-IP.5060: SIP/2.0 100 trying KAMAILIO-IP.5060 > ASTERISK-IP.5060: INVITE sip:number@KAMAILIO-IP:5060 ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 100 Trying ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 200 OK KAMAILIO-IP.5060 > NEWVENDOR-IP.5060: SIP/2.0 200 OK NEWVENDOR-IP.5060 > KAMAILIO-IP.5060: ACK sip:number@KAMAILIO-IP:5060 ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 200 OK KAMAILIO-IP.5060 > NEWVENDOR-IP.5060: SIP/2.0 200 OK NEWVENDOR-IP.5060 > KAMAILIO-IP.5060: ACK sip:number@KAMAILIO-IP:5060 ASTERISK-IP.5060 > KAMAILIO-IP.5060: SIP/2.0 200 OK KAMAILIO-IP.5060 > NEWVENDOR-IP.5060: SIP/2.0 200 OK NEWVENDOR-IP.5060 > KAMAILIO-IP.5060: ACK sip:number@KAMAILIO-IP:5060 ... and so on ... Can I do something about this or is a change on MetaSwitch'es side in order? Thank you! Markus

3 3

BYE request with Sipp refused
by Loic Chabert 08 Jul '15

08 Jul '15

Hello everyone, I'm a newbie in kamailio world and make some tests to dispatch call over several asterisk instances. This part works, my call has been distributed over two instances (if i make a call with jitsi for example). Now, i want to test performance of my cluster and use Sipp to chalenge it. All method is working fine, except the BYE method, where kamailio send me back a 404 Not here. This is my SIPP scenario: ... <pause/>  <send retrans="500"> <![CDATA[ BYE sip:[service]@[remote_ip] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp <sip:sipp@loicchabert.fr>;tag=[pid]SIPpTag00[call_number] To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param] Call-ID: [call_id] CSeq: 2 BYE Contact: sip:sipp@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]]> </send> ... I use the classic file of kamailio, with some adjustement for dispatching. Any ideas concerning this problem ? This BYE is not in transaction so he is drop by kamailio ? Thank in advance, Regards.

3 6

Sending a message form within kamailio
by Shouvanik Chakrabarti 08 Jul '15

08 Jul '15

Hello, I am writing a module for kamailio and as part of its functionality I need to generate and send SIP messages. How would I go about sending an SIP message from within kamailio? Any help would be deeply appreciated. Thanks, Shouvanik

3 5

Kamailio as radius server
by Ali Taher 08 Jul '15

08 Jul '15

Hello, I'm wondering if kamailio can act as radius server who's responsible of authentication,authorization and accounting, where kamailio receive accounting record types (start-stop-interim) and write them in a database. Thanks Ali

3 3

CNXCC cnxcc_set_max_time Version 4.2
by Jonathan Hunter 08 Jul '15

08 Jul '15

Hi Guys, I have just moved from kamailio 4.1, to 4.2 and have installed via git version 4.2 I am testing with the function cnxcc_set_max_time, as I want to set the maximum call duration. Having moved to 4.2, i cant get the module to terminate the call. I have set dlg_flag, and cc_flag, following examples on the wiki but I cant get it to work. Does anyone have any suggestions for me please? My same code on 4.1 works ok. Thanks Jon

1 1

Potential ip_addr_match_net bug
by Kyle Kurz 07 Jul '15

07 Jul '15

I'm using allow_address function of the permissions module and I'm seeing strange behavior with network masks not divisible by 8. I have 172.16.0.0/12 in my address table and Kamailio is marking packets from 172.56.16.232 as being part of that local subnet. If I change the subnet to 172.16.0.0/16, I no longer have an issue. I know there was a bug several years ago related to this ( http://git.kamailio.org/gitlist/index.php/sip-router/commit/0ead7ab31bf0df6…), but that code is very different, so I'm wondering if a regression has crept in? I'm using Kam 4.1.6. Thanks, Kyle -- *Kyle Kurz* Digium, Inc. · Senior Software Engineer 9115 Brown Deer Rd · San Diego, CA 92121 · US direct +1 858.537.1166 Check us out at: http://digium.com · http://asterisk.org <http://www.asterisk.org/>

3 3

dmq_replicate deserializing?
by Asgaroth 07 Jul '15

07 Jul '15

Hi All, I am playing with the new dmq_replicate module and am banging my head against an issue I have come accross. I can see the kamailio registrars sending the replication messages to the other node(s) in the dmq bus, and the payload looks to be of type json. However, on the recieving nodes, when I try to issue a kamctl ul show --brief, the output of the AOR's are garbled. I had a look at the dmq/dmq_ursloc moduled, and the dmq module docs mention that we may need to load our own deserializers, is this the case when using the dmq_usrloc module as well? I am only using dmq to replicate registration messages. Here is a sample of the dmq workers on a recieving node applying the update, and you can see that it thinks the contact to add is 'p÷#031#002', is was expecting the actual AOR of the subriber to show up here. Am I missing something simple here or is there something more sinister at play. Any pointers would be greatly appreciated. Kamailio Version: version: kamailio 4.3.0 (x86_64/linux) c6aa95 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: c6aa95 compiled on 16:14:27 Jun 23 2015 with gcc 4.4.7 Debug log below: Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:606]: parse_msg(): SIP Request: Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:608]: parse_msg(): method: <KDMQ> Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:610]: parse_msg(): uri: <sip:usrloc@10.6.0.174:5060> Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:612]: parse_msg(): version: <SIP/2.0> Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/parse_via.c:1254]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK9b3c.fe993716000000000000000000000000.0>; state=16 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/parse_via.c:2642]: parse_via(): end of header reached, state=5 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:496]: parse_headers(): parse_headers: Via found, flags=2 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:498]: parse_headers(): parse_headers: this is the first via Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [receive.c:134]: receive_msg(): After parse_msg... Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [receive.c:177]: receive_msg(): preparing to run routing scripts... Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: tm [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=450 global id=449 T start=0xffffffffffffffff Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=10 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:173]: get_hdr_field(): DEBUG: get_hdr_field: <To> [30]; uri=[sip:usrloc@10.6.0.174:5060] Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:175]: get_hdr_field(): DEBUG: to body [<sip:usrloc@10.6.0.174:5060>#015#012] Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <10> <KDMQ> Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=471 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=50105, isACK=0 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: tm [t_lookup.c:1080]: t_check_msg(): DEBUG: t_check_msg: msg id=450 global id=450 T end=(nil) Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: dmq [message.c:53]: dmq_handle_message(): dmq_handle_message [KDMQ sip:usrloc@10.6.0.174:5060] [ ] Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: dmq [message.c:65]: dmq_handle_message(): dmq_handle_message peer found: usrloc Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq [worker.c:84]: worker_loop(): dmq_worker [0 23014] lock acquired Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: <core> [parser/parse_addr_spec.c:172]: parse_to_param(): DEBUG: add_param: tag=390c95c339281829d3cea6f43c8512cb-7e62 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: <core> [parser/parse_addr_spec.c:894]: parse_addr_spec(): end of header reached, state=29 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq_usrloc [usrloc_sync.c:250]: usrloc_dmq_handle_msg(): dmq message received from sip:usrloc@10.6.0.173:5060 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq_usrloc [usrloc_sync.c:353]: usrloc_dmq_handle_msg(): Received DMQ_UPDATE. Update contact info... Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq_usrloc [usrloc_sync.c:60]: add_contact(): 'p÷#031#002' found in usrloc Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq_usrloc [usrloc_sync.c:62]: add_contact(): get_ucontact = 0 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq_usrloc [usrloc_sync.c:72]: add_contact(): Found contact Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: usrloc [ucontact.c:1688]: update_ucontact(): exists callback for type= UL_CONTACT_UPDATE Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: usrloc [ul_callback.h:84]: run_ul_callbacks(): contact=0x7ffe86eabad8, callback type 2/15, id 0 entered Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq_usrloc [usrloc_sync.c:494]: dmq_ul_cb_contact(): Callback from usrloc with type=2 Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq_usrloc [usrloc_sync.c:517]: dmq_ul_cb_contact(): Contact recieved from DMQ... skip Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq_usrloc [usrloc_sync.c:85]: add_contact(): Release record Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq_usrloc [usrloc_sync.c:87]: add_contact(): Unlock udomain Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: sl [sl.c:280]: send_reply(): reply in stateless mode (sl) Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: <core> [msg_translator.c:158]: check_via_address(): (10.6.0.173, 10.6.0.173, 0) Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [usr_avp.c:631]: destroy_avp_list(): destroying list (nil) Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [xavp.c:446]: xavp_destroy_list(): destroying xavp list (nil) Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23007]: DEBUG: <core> [receive.c:278]: receive_msg(): cleaning up Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq [worker.c:134]: worker_loop(): sent reply Jul 1 20:40:52 bfievkrl02 /usr/sbin/kamailio[23014]: DEBUG: dmq [worker.c:82]: worker_loop(): dmq_worker [0 23014] getting lock

2 6

SRTP to RTP with Kamailio
by Alberto Sagredo 07 Jul '15

07 Jul '15

Hi I have a Kamailio in front of an Asterisk Cluster and moved to rtpengine to give support for SRTP from rtpproxy What i want is to have SRTP on Kamailio-rtpengine leg but RTP inside network for Asterisk. I have found that config https://loadmultiplier.com/sites/default/files/kamailio/pcscf/kamailio.cfg But could not be able to make it work? Anyone did this scenareo and could give me some tips? I have enabled sdops and be able to remove SRTP and crypto lines but not be able to make it work as i mention Thanks in advanced Alberto

1 1

IMS server with a core network
by test testt 07 Jul '15

07 Jul '15

hello, I posted a message in the kamailio forum but it wasn't in the good place, so I rebroadcast at this e-mail as it was say. Hello, I have installed Kamailio 4.2 and I would like to use it as IMS server but I don't have any ideas how to do this. I have a LTE core network which includes HSS ans PCRF. I saw on this page ( http://www.kamailio.org/w/2013/05/ims-kamailio/ ) that is apparently easy, just need to take the P-CSCF, S-CSCF and I-CSCF Kamailio and Diameter config files in the examples folder and run kamailio but I don't know how to use them and I didn't find any tutorials about it. Could you help me please ? Marc link: https://github.com/kamailio/kamailio/issues/227#issuecomment-118073500 I tried the 2 tutorials that it has given but it's not really working I am using ubuntu 15.04 Thanks Marc

1 0

Dispatcher, Failure Route and fr_timeout
by Asgaroth 07 Jul '15

07 Jul '15

Hi All, I would like to reduce the time it takes for kamailio to timeout an initial message to a dispatcher destination so that it will select the next destination in the set, before, hopefully, a restransmission. In this case I am testing registration messages. I came accross the t_set_fr() function in the TM module documentation today and have been experimenting with it for some hours. I have an issue where a 408 timout is sent back to the UAC before the last destination in the dispatcher set is selected, I suspect I am hitting the timeout somewhere, but I cannot see where, I presume the fr_timeout is set every time I issue the t_set_fr() call and the associated t_relay. I'm just wondering if I am doing this correctly, if one of the guru's out there wouldnt mind casting an eyeball over this config and let me know if I am on the right track, or if I am going about this in the wrong way. Any suggestions are most welcome. What appears to be happening is, I have 3 destinations in the dispatcher set, it looks like the 1st 2 are tried, which correctly timeout and are set inactive-probing, but on the last destination, which is working, I appear to be getting a 408 timeout being sent back from kamailio. Here is a snip of my configuration: #!define NO_RESPONSE_TIMER 250 request_route { t_on_failure("DISPATCHER_FAILURE"); t_set_fr(0, NO_RESPONSE_TIMER); if(!ds_select_dst("1", "0")) { xlog("L_ERR", "No dispatcher destinations available for $ru\n"); sl_send_reply("500", "No destination available"); exit; } route("RELAY") } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } failure_route[DISPATCHER_FAILURE] { if (t_is_canceled()) { exit; } # select next destination only for local timeout if (t_branch_timeout() && !t_branch_replied()) { xlog("L_INFO", "route[DISPATCHER_FAILURE] : $rm : timeout and no reply ($si:$sp->$Ri:$Rp->$du)\n"); xlog("L_INFO", "route[DISPATCHER_FAILURE] : $rm : setting $du to probing state"); ds_mark_dst("ip"); if(ds_next_dst()) { xlog("L_INFO", "route[DISPATCHER_FAILURE] : $rm : next destination selected ($du)\n"); t_set_fr(0, NO_RESPONSE_TIMER); t_relay(); exit; } else { xlog("L_INFO", "route[DISPATCHER_FAILURE] : $rm : No destinations available for $rd"); send_reply("500", "No destination available"); exit; } } }

2 2

How to configure Cx - diameter running on SCTP
by Jan Devai 07 Jul '15

07 Jul '15

Hi guys, I would like to configure Cx interface, i.e. Diameter on SCTP between I/S-CSCF and external HSS. Can somebody Guide me how to achieve it or if somebody who already did that, can he share his configurations files with me? Thank you Jan

1 0

clarification on tcp keepalive
by Kelvin Chua 07 Jul '15

07 Jul '15

using kamailio 4.3 on ubuntu server 14.04 i noticed tcp socket will be killed by kamailio after around 11 minutes with or without tcp_keepalive=no. tcpdump shows that by default (tcp_keepalive=yes) there is indeed keepalive packets going back and forth. if set to no, there would be nothing in between. if i have clients who have registration timeout of 1700, they would send re-registration around half of that, or around 14 minutes after registration. then the socket would be long dead and they would be unreachable for 3 minutes. based on procfs, ubuntu have settings for 9 probes with 75 seconds interval so that means around 11 minutes of sending keepalives. so i tried changing these values, on procfs. and tried playing around with these kamailio cfg settings as well: tcp_keepcnt tcp_keepidle tcp_keepintvl whatever i do, the interval between keepalives are always 2 minutes and after 4-5 probes, kamailio will send a FIN to the client. i checked the source code tcp_main.c and it looks alright. could this be an ubuntu issue? anybody seen this before? I created a workaround by setting expire 900 for all registrations but this is not ideal. Kelvin Chua

2 1

error in kamctlrc line 13 not sure of "CHROOTED DIRECTORY"
by Joseph Zimmer 07 Jul '15

07 Jul '15

Sorry, this is my problem: have included the kamctlrc, kamailio.cfg, and modus.lst files if that helps, run create command, not sure of path for chrooted directory in kamctlrc file so I have not been able to create my initial database. I know that is holding up creation of MySQL database. can you assist, sent contribution to this email address, could not find any other. thanks in advance, joe zimmer +1 703-957-0186 USA root@telephonepros:~# /usr/local/sbin/kamdbctl create /usr/local/etc/kamailio//kamctlrc: line 13: =/path/to/chrooted/directory: No such file or directory MySQL password for root: INFO: test server charset /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 105: mysql: command not found /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 106: mysql: command not found Usage: grep [OPTION]... PATTERN [FILE]... Try 'grep --help' for more information. /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 112: [: =: unary operator expected INFO: creating database kamailio ... /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 71: mysql: command not found ERROR: Creating database kamailio failed! root@telephonepros:~#

2 1

DIAMETER based on TCP requires a FIXED I/S-CSCFSOURCE PORT in order to make Cx work towards a External HSS
by Jan Devai 07 Jul '15

07 Jul '15

Hi experts, from the current trace we can see that the I/S-CSCF always changes SOURCE port when contacting our eternal HSS. In order to make the Cx work HSS expects the Source port to be fixed on the TCP layer and in our I/S-CSCF configuration we did not find any setting for SOURCE Port. Can somebody help us please ;-) ? We appreciate your support, Br, Jan

3 2

Looking for kamailio service supplier.
by ZhuYue Zhang 07 Jul '15

07 Jul '15

Hi Sir, This is a Chinese company from ABB group, We're looking for a VOIP system solution, As our understanding, kamailio is the best choice, Does any Chinese company take this business? Thank you and Best Regards. [cid:image001.png@01D02128.0536F2C0] Zhuyue.zhang ABB Genway Xiamen Electrical Equipment CO.,LTD 7F,NO.23 Wanghai Road,Software Park 2,Lvling Road Xiamen SEZ,Fujian,China 361008 Phone: (86)592-2954819 Mobile: (86)15260200863

1 0

mongodb and jansson
by Kelvin Chua 06 Jul '15

06 Jul '15

when using mongodb with jansson, intermittent failures will be seen because jansson would change the order of keys inside a json object. mongodb would expect the first key to be a command, for example "insert". this is disastrous! below is a fix for jansson which will preserve the order on which the json object was created. --- modules/jansson/jansson_utils.c.old 2015-07-06 17:41:25.852770681 +0800 +++ modules/jansson/jansson_utils.c 2015-07-06 17:42:27.384769579 +0800 @@ -33,7 +33,7 @@ val->flags = 0; if(json_is_object(v) || json_is_array(v)) { - const char* value = json_dumps(v, JSON_COMPACT); + const char* value = json_dumps(v, JSON_COMPACT|JSON_PRESERVE_ORDER); *freeme = (char*)value; val->rs.s = (char*)value; val->rs.len = strlen(value); Kelvin Chua

2 1

Re: [SR-Users] dispather
by Slava Bendersky 06 Jul '15

06 Jul '15

Hello Daniel, Thank you for reply, based on you example. Can I do some thing like that. modparam("dispatcher", "dstid_avp", "$avp(dsdst_id)") switch ($fd) { case "domain1.tld": $avp(dsdst_id)=1; break; case "domain2.tld": $avp(dsdst_id)=2; break; } ds_select_dst($avp(dsdst_id),4); Slava. From: "Daniel Tryba" <d.tryba(a)pocos.nl> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, July 3, 2015 6:18:03 AM Subject: Re: [SR-Users] dispather On Thursday 02 July 2015 09:12:41 Slava Bendersky wrote: > Is dispatcher module can do call routing based on domain ? > This only inbound traffic. > So if call come from @domain1.tld -----> asterisk box 1 @domain2.tld -----> > asterisk box 2. Sure, just make the dispatcher target dynamic with an avp: if($fd=="domain1.tld") { $avp(dispatcher_id)=1; } ... ds_select_dst($avp(dispatcher_id),4); _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Re: [SR-Users] Kamailio multiple registration
by Fadi Hawari 06 Jul '15

06 Jul '15

Dear Sir , i need to register 10k users in sequance could you please let me know how to do it . Regards

5 4

modules.lst
by Joseph Zimmer 06 Jul '15

06 Jul '15

Hey guys, new to kamailio, keep getting stuck on step 7 - creating first database.Can you assist? see below: root@telephonepros:~# /usr/local/sbin/kamdbctl create /usr/local/etc/kamailio//kamctlrc: line 13: =/path/to/chrooted/directory: No such file or directory MySQL password for root: INFO: test server charset /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 105: mysql: command not found /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 106: mysql: command not found Usage: grep [OPTION]... PATTERN [FILE]... Try 'grep --help' for more information. /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 112: [: =: unary operator expected INFO: creating database kamailio ... /usr/local/lib/kamailio//kamctl/kamdbctl.mysql: line 71: mysql: command not found ERROR: Creating database kamailio failed! root@telephonepros:~#

2 1

Siremis 4.2 and Kamailio 4.3
by Alberto Sagredo 03 Jul '15

03 Jul '15

Hi Im installing Siremis 4.2 and getting this error when login. INstallation Wizard was fine and all dependencies are installed following Asipto Guide for Siremis. Any idea what could be wrong? Popup i get is.. {"target":"ERROR","content":" \n[2015-07-03 10:04:05 (GMT)] An exception occurred while executing this script: \nError message: #8192, Non-static method BizSystem::getConfiguration() should not be called statically, assuming $this from incompatible context<\/font> \nScript name and line number of error: \/var\/www\/siremis-4.2.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php:268<\/font> \n *function:<\/b> errorHandler ( 8192, \"Non-static method BizSystem::getConfiguration() should not be ca...\", \"\/var\/www\/siremis-4.2.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php\", 268, Array(0) ) @ \/var\/www\/siremis-4.2.0\/openbiz\/bin\/sysheader.inc 117\nfunction:<\/b> userErrorHandler ( 8192, \"Non-static method BizSystem::getConfiguration() should not be ca...\", \"\/var\/www\/siremis-4.2.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php\", 268, Array(0) ) @ \/var\/www\/siremis-4.2.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php 268\nfunction:<\/b> setQuoteIdentifiers ( ) @ \/var\/www\/siremis-4.2.0\/openbiz\/bin\/data\/BizDataObj_Abstract.php 177\nfunction:<\/b> __construct ( Array(1) ) @ \/var\/www\/siremis-4.2.0\/openbiz\/bin\/data\/BizDataObj.php 37\nfunction:<\/b> __construct ( Array(1) ) @ \/var\/www\/siremis-4.2.0\/openbiz\/bin\/ObjectFactory.php 166\nfunction:<\/b> constructObject ( \"system.do.UserDO\" ) @ \/var\/www\/siremis-4.2.0\/openbiz\/bin\/ObjectFactory.php 56\nfunction:<\/b> getObject ( \"system.do.UserDO\", 0 ) @ \/var\/www\/siremis-4.2.0\/openbiz\/bin\/BizSystem.php 252\nfunction:<\/b> getObject ( \"system.do.UserDO\" ) @ \/var\/www\/siremis-4.2.0\/siremis\/modules\/service\/authService.php 92\nfunction:<\/b> authDBUser ( \"admin\", \"admin\" ) @ \/var\/www\/siremis-4.2.0\/siremis\/modules\/service\/authService.php 66\nfunction:<\/b> authenticateUser ( \"admin\", \"admin\" ) @ \/var\/www\/siremis-4.2.0\/siremis\/modules\/user\/form\/LoginForm.php 92\nfunction:<\/b> Login ( ) @ \/var\/www\/siremis-4.2.0\/openbiz\/bin\/BizController.php 310\nfunction:<\/b> invoke ( ) @ \/var\/www\/siremis-4.2.0\/openbiz\/bin\/BizController.php 110\nfunction:<\/b> dispatchRequest ( ) @ \/var\/www\/siremis-4.2.0\/openbiz\/bin\/BizController.php 32\nfunction:<\/b> include_once ( \"\/var\/www\/siremis-4.2.0\/openbiz\/bin\/BizController.php\" ) @ \/var\/www\/siremis-4.2.0\/siremis\/bin\/controller.php 6<\/div>\n------------------------------Please ask system administrator for help...<\/div>\n"}]* *BR*

2 5

subst + insert_hf issue
by Igor Potjevlesch 03 Jul '15

03 Jul '15

Hello, I'm doing a subst on the From header for some particular cases, like this one: subst('/^(From|f):(.*)<sip:(.*)@/\1: "Anonymous" <sip:\3@/g'); In the same route (REQUEST_ROUTE called "INVITE"), I do the following: insert_hf("X-CarrierName: $avp(s:carrier_id)\r\n"); For many User-Agent, the X-CarrierName header is inserted after the @ into the user part of the From. So the From: looks like this: From:"Anonymous" <sip:0123456789@X-CarrierName: ABCDE\r\n The domain name part of the From is on the next line: <domain>:5060>;tag=3fdce0-a658bfa-13c4-50029-55941ccd-5eab9369-55941ccd\r\n I can't reproduce. The only potential explanation is that this impact the User-Agent that have the From at the top of the headers. If someone know what could going wrong here. Regards, Igor

2 3

Diameter sorce/destination port
by micunek＠gmail.com 03 Jul '15

03 Jul '15

Hi All, I am trying to use Cx (Diameter) interface via TCP towards our HSS according to Carsten Bock example. However kamailio client obtain TCP RST from HSS because a kamailio source port is different than port where kamailio is listen. I do not know if this is Diameter/Cx requirement or not. I did not find it in specifications. Is there any way how to ensure that kamailio use same port number for source and destination (3869)?. <?xml version="1.0" encoding="UTF-8"?> <DiameterPeer FQDN="...3gppnetwork.org" Realm="...3gppnetwork.org" Vendor_Id="10415" Product_Name="CDiameterPeer" AcceptUnknownPeers="1" DropUnknownOnDisconnect="1" Tc="30" Workers="4" QueueLength="8" TransactionTimeout="5" SessionsHashSize="128" DefaultAuthSessionTimeout="3600" MaxAuthSessionTimeout="3600" > <Peer FQDN="hss....3gppnetwork.org" Realm=".3gppnetwork.org" port="3868"/> <Acceptor port="3869" /> <Auth id="16777216" vendor="10415"/>  <Realm name="....3gppnetwork.org"> <Route FQDN="hss...3gppnetwork.org" metric="10"/> </Realm> <DefaultRoute FQDN="hss.....3gppnetwork.org" metric="10"/> </DiameterPeer> Thanks Peter

2 1

Segment fault
by Nick Raper 03 Jul '15

03 Jul '15

Hi, Can anyone help with a segment fault we had today please? Our Kamailio has been running for months with no problem but today we received this error 3 times and Kamailio stopped. Jul 1 12:48:43 sipa kernel: kamailio[2853]: segfault at 7f516819f6fc ip 00007f516a51c4a6 sp 00007fffd0fbe420 error 7 in pv.so[7f516a510000+36000] The full log files are attached for the three seg faults. Regards, [cid:image001.jpg@01CFEF9D.A1F537A0] Nick Raper Infrastructure Engineer 4th Dimension Technology | South Africa Office: +27 21 506 6260 | 24 hour helpline: +27 21 506 6260 | Fax: +27 86 610 5792 Email: nick(a)4dt.co.za<mailto:nick@4dt.co.za> | Web: www.4dt.co.za<http://www.4dt.co.za/> P Please consider the environment before printing this email

3 9

dispather
by Slava Bendersky 03 Jul '15

03 Jul '15

Hello Everyone, Is dispatcher module can do call routing based on domain ? This only inbound traffic. So if call come from @domain1.tld -----> asterisk box 1 @domain2.tld -----> asterisk box 2. Or which way is better to achieve it ? Thank you.

2 1

start-up crash if wdir is set and WITH_AUTH is defined (Kamailio 4.2)
by jack wang 02 Jul '15

02 Jul '15

Hello, If the 'wdir' (or 'workdir') is not set in kamctl.cfg or WITH_AUTH feature is not defined, kamailio can normally be started. Actually , the purpose is thatI want to add the 'wdir' to set the path coredump file... And ther error logs like following: (modified for hiding private info) ./kamailio[3565]: ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module auth_db (/xxx/kamailio/lib/kamailio/modules/auth_db.so) ./kamailio[3565]: ERROR: <core> [pt.c:341]: fork_process(): init_child failed for process 2, pid 3565, "udp receiver child=0 sock=10.100.90.62:5060 (10.100.90.62:5060)" ./kamailio[3565]: CRITICAL: <core> [main.c:1621]: main_loop(): Cannot fork ./kamailio[3568]: ERROR: db_sqlite [dbase.c:67]: db_sqlite_new_connection(): failed to open sqlite database 'tmp/SQLITE_DB' ./kamailio[3564]: ERROR: db_sqlite [dbase.c:67]: db_sqlite_new_connection(): failed to open sqlite database 'tmp/SQLITE_DB' ./kamailio[3568]: ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool ./kamailio[3568]: ERROR: auth_db [authdb_mod.c:172]: child_init(): unable to connect to the database ./kamailio[3568]: ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module auth_db (/xxx/kamailio/lib/kamailio/modules/auth_db.so) ./kamailio[3568]: ERROR: <core> [mi.c:128]: init_mi_child(): failed to init proc rpc for sip handling ./kamailio[3564]: ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool ./kamailio[3564]: ERROR: auth_db [authdb_mod.c:172]: child_init(): unable to connect to the database ./kamailio[3568]: CRITICAL: mi_fifo [mi_fifo.c:233]: fifo_process(): Failed to init the mi process ./kamailio[3564]: ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module auth_db (/xxx/kamailio/lib/kamailio/modules/auth_db.so) ./kamailio[3564]: ERROR: <core> [pt.c:341]: fork_process(): init_child failed for process 1, pid 3564, "udp receiver child=0 sock=[2010:0:0:0:0:0:0:1]:5060" ./kamailio[3567]: CRITICAL: <core> [main.c:1693]: main_loop(): cannot fork timer process ./kamailio[3569]: ERROR: db_sqlite [dbase.c:67]: db_sqlite_new_connection(): failed to open sqlite database 'tmp/SQLITE_DB' ./kamailio[3569]: ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool ./kamailio[3564]: CRITICAL: <core> [main.c:1621]: main_loop(): Cannot fork ./kamailio[3569]: ERROR: auth_db [authdb_mod.c:172]: child_init(): unable to connect to the database ./kamailio[3569]: ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module auth_db (/xxx/kamailio/lib/kamailio/modules/auth_db.so) ./kamailio[3569]: ERROR: <core> [pt.c:341]: fork_process(): init_child failed for process 6, pid 3569, "ctl handler" ./kamailio[3569]: ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module ctl (/xxx/kamailio/lib/kamailio/modules/ctl.so) ./kamailio[3569]: ERROR: <core> [main.c:1707]: main_loop(): error in init_child ./kamailio[3563]: ALERT: <core> [main.c:781]: handle_sigs(): child process 3565 exited normally, status=255 ./kamailio[3563]: ALERT: <core> [main.c:781]: handle_sigs(): child process 3566 exited normally, status=255 ./kamailio[3572]: ERROR: db_sqlite [dbase.c:67]: db_sqlite_new_connection(): failed to open sqlite database 'tmp/SQLITE_DB' ./kamailio[3572]: ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool ./kamailio[3572]: ERROR: auth_db [authdb_mod.c:172]: child_init(): unable to connect to the database ./kamailio[3572]: ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module auth_db (/xxx/kamailio/lib/kamailio/modules/auth_db.so) ./kamailio[3572]: ERROR: <core> [pt.c:341]: fork_process(): init_child failed for process 9, pid 3572, "RTIMER SEC EXEC" ./kamailio[3572]: ERROR: rtimer [rtimer_mod.c:164]: child_init(): failed to start timer routine as process ./kamailio[3572]: ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module rtimer (/xxx/kamailio/lib/kamailio/modules/rtimer.so) ./kamailio[3572]: ERROR: <core> [main.c:1707]: main_loop(): error in init_child ./kamailio[3573]: ERROR: db_sqlite [dbase.c:67]: db_sqlite_new_connection(): failed to open sqlite database 'tmp/SQLITE_DB' ./kamailio[3573]: ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool ./kamailio[3573]: ERROR: auth_db [authdb_mod.c:172]: child_init(): unable to connect to the database ./kamailio[3573]: ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module auth_db (/xxx/kamailio/lib/kamailio/modules/auth_db.so) ./kamailio[3571]: ERROR: db_sqlite [dbase.c:67]: db_sqlite_new_connection(): failed to open sqlite database 'tmp/SQLITE_DB' ./kamailio[3573]: ERROR: <core> [pt.c:479]: fork_tcp_process(): init_child failed for process 10, pid 3573, "tcp receiver (generic) child=0" ./kamailio[3571]: ERROR: <core> [db.c:322]: db_do_init2(): could not add connection to the pool ./kamailio[3573]: ERROR: <core> [tcp_main.c:4869]: tcp_init_children(): fork failed: No such file or directory ./kamailio[3571]: ERROR: auth_db [authdb_mod.c:172]: child_init(): unable to connect to the database ./kamailio[3571]: ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module auth_db (/xxx/kamailio/lib/kamailio/modules/auth_db.so) ./kamailio[3571]: ERROR: <core> [pt.c:341]: fork_process(): init_child failed for process 8, pid 3571, "Dialog Clean Timer" ./kamailio[3571]: ERROR: dialog [dialog.c:742]: child_init(): failed to start clean timer routine as process ./kamailio[3571]: ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module dialog (/xxx/kamailio/lib/kamailio/modules/dialog.so) ./kamailio[3571]: ERROR: <core> [main.c:1707]: main_loop(): error in init_child Thanks in advance. :)

2 1

Problem Storing USRLOC Additional Attributes
by Luca Mularoni 02 Jul '15

02 Jul '15

Hi All, I'm using kamailio 4.2.5 (compiled from source and aligned up to commit [00a36ae07c587992d6486277ce2e9ae6c009685f]). In order to enrich users data collected during registering phase, I would like to store them in USRLOC Additional Attributes through the use of 'xavp_contact' structure, but I'm having problems storing them. Below relevant chunks of config: ... # ----- usrloc params ----- modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", 0) modparam("usrloc", "xavp_contact", "ulvals") ... route[REGISTRAR] { ... # Just an example $xavp(ulvals=>A) = "A"; $xavp(ulvals[0]=>B) = "B"; $xavp(ulvals[0]=>C) = "C"; $xavp(ulvals[0]=>D) = "D"; ... if (!save("location")) sl_reply_error(); exit; } route[LOCATION] { ... if(!lookup("location") { # Handle Failure ... } # Print USRLOC Additional Attributes # Just an Example $var(i) = 0; $var(N) = $branch(count); while($var(i)<=$var(N)) { xlog("LOC - [$var(i)] - A [$xavp(ulvals[$var(i)]=>A)]\n"); xlog("LOC - [$var(i)] - B [$xavp(ulvals[$var(i)]=>B)]\n"); xlog("LOC - [$var(i)] - C [$xavp(ulvals[$var(i)]=>C)]\n"); xlog("LOC - [$var(i)] - D [$xavp(ulvals[$var(i)]=>D)]\n"); $var(i) = $var(i) + 1; } ... } And this is the relevant chunk of syslog when a local subscriber is found in 'location' table: ... LOC - [0] - A [A] LOC - [0] - B [<null>] LOC - [0] - C [<null>] LOC - [0] - D [D] ... Summing up, it seems that when xavp structure is stored in memory, just first and last elementsare inserted. In real config I'm using more than 4 attributes and behaviour is the same: just first and last are stored. I've tried also with modparam("usrloc", "db_mode", 0) but the problem remains. Where I'm wrong in using this structure? Or it could be a bug? Thank You in advance for your help Luca Mularoni

1 0

(no subject)
by Nelson Migliaro 01 Jul '15

01 Jul '15

Hello everybody, My SIP vendor request me to replace FROM before sending the traffic. In order to achieve this I use uac_replace_from. UAC module is setup in restore_mode = auto. In my insfrastructure I have an Asterisk and then a Kamailio that connects to vendor via internet. Softphone -> Asterisk -> Kamailio -> Internet -> SIP vendor If caller ID is setup in Asterisk using CALLERID(num)=34888888888 and then INVITE is forwarded to Kamailio, the call is established and finished correctly but the URI in TO field in BYE request from Kamailio to Asterisk contains garbage. In the scenario the callee hangs up the call. Example of TO Field with garbage: 34888888888 <sip:50026896@no{soy,ns^^> What I do see is that the number "50026896" that is part of the URI is the same I use in: uac_replace_from("50026896", "sip:50026896@sip.vendor.es"); Something else that I have found is that vsf field is the same in the INVITE and in the BYE. ---------------------------------------------------------------------------------------------------------------- 2015/06/23 17:48:38.552442 192.168.0.2:5060 -> 192.168.0.1:5060 BYE sip:34888888888@192.168.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2;branch=z9hG4bKfb49.73c6517609cdb6f7ec00b1f40a05dbe9.0 Via: SIP/2.0/UDP 8.8.8.8.8;rport=5060;branch=z9hG4bKfcdf.3767c59c2d0e3d8ab695669845ce4cea.0 branch=z9hG4bK04boo6104o5hcso0c2a1sd0000g00.1 Call-ID: 4bf8effb45b0ae8e049366297924cbba@192.168.0.1:5060 From: <sip:28999999999@192.168.0.2;tag=k0eci3x3-CC-30 To: 34888888888 <sip:50026896@no{soy,ns^^>;tag=as041b7d84 CSeq: 1 BYE Reason: Q.850;cause=16;text="normal call clearing" Max-Forwards: 67 Content-Length: 0 ------------------------------------------------------------------------------------------------------------------ DEBUG: uac [replace.c:525]: restore_uri(): getting 'vsf' Route param DEBUG: uac [replace.c:533]: restore_uri(): route param is 'AAAAAAEECQkCAgsNAXBeL0NGQUsfVl02Ni44Mw--' (len=40) DEBUG: uac [replace.c:607]: restore_uri(): decoded uris are: new=[sip:50026896@no{soy,ns#005#007] old=[sip:34888888888@8.8.8.8] DEBUG: uac [replace.c:525]: restore_uri(): getting 'vst' Route param DEBUG: uac [replace.c:533]: restore_uri(): route param is 'AAAAAAQPAw8MDgsAAHZBKRVdAhoVHQ4XH1BdYWJhbnRlLmVz' (len=48) DEBUG: uac [replace.c:607]: restore_uri(): decoded uris are: new=[ sip:28999999999@192.168.0.1] old=[sip:999999999@sip.vendor.es] -----------------------------------------------------------------------------------------------------------------------------------

2 1

Kamailio no recieved 480/486 messages
by Yuriy Gorlichenko 01 Jul '15

01 Jul '15

At my confiig I have route like this onreply_route[REPLY_FROM_WS] { if(status=~"[12][0-9][0-9]") { xlog("L_INFO", "Manage_Reply from webrtc client {$si:$sp} for method {$rm}: $rs"); rtpengine_manage("force trust-address replace-origin replace-session-connection DTLS=passive ICE=remove RTP/AVP"); route(NATMANAGE); } } This route successfully handling replies 180 183 and 200 and then (after natmanage route) resend to mediaserver. When callee client reject call it send 480 ot 486 message that not resended by kamailio nowhere. First of all I thought that 480 and 486 replies are failure replies but at log I see its at onreply route. So Ok . I added this to onreply_route if (status=~"48[06]"){ xlog("L_INFO", "Manage_Reply from webrtc client {$si:$sp} for method {$rm}: $rs"); rtpengine_manage("force trust-address replace-origin replace-session-connection DTLS=passive ICE=remove RTP/AVP"); route(NATMANAGE); } But message just going through natmanage route and goes nowhere My NATMANAGE route is xlog("L_INFO","NATMANAGE reply {$rm}"); if (is_reply()) { if(isbflagset(FLB_NATB)) { add_contact_alias(); } xlog("L_INFO","reply {$rm}"); } I see at my log only first NATMANAGE xlog... no more... thnks for answer

2 3

dialog flags vs dialog variables vs dialog profiles
by Muhammad Shahzad 01 Jul '15

01 Jul '15

Hi, Lasting night i was doing some tests and identified a strange behavior from dialog module. I am using Kamailio v4.3.0 rev. c6aa95. I basically wanted to identify if an incoming sequential ACK (from caller for an answered call) or a BYE (from callee who wants to hangup the call) belong to an existing dialog. First I tried to use dialog profiles as they were already enabled in my kamailio.cfg for every call. Calling the function "is_in_profile" after loose_route in WITHINDLG route, does not works for ACK, but works for BYE. Then I added dialog flags to initial INVITE (using function "dlg_setflag") and check them for ACK and BYE after loose_route (using "dlg_isflagset" method), i have the same behavior. They worked/matched only for BYE, not for ACK. Lastly I set dialog variables (e.g. dlg_var(my_call) = "1") in initial INVITE and checked for them for ACK and BYE requests after loose_route. Only they worked perfectly, i.e. i can match them for both incoming ACK and BYE requests. Is this correct and expected behavior? Why dialog profile and flags do not match for a dialog in unconfirmed state (i.e. ACK from caller after successful call answer)? They only work for confirmed dialogs (e.g. BYE from callee).On the other hand, dialog variables seems to work everywhere (i.e. all sequential requests). Thank you.

2 1

TLS certificate verification failed
by Austin Einter 01 Jul '15

01 Jul '15

Hi All I am having a sip client, I am using GNU tls for transport layer security. I am using Kamailio (TLS enabled) to test all the functionalities. I did registration over TLS. It was fine. I tried to make call. I was able to send INVITE. Kamaiio trying to connect to called party (which is again my sip endpoint). That time certificate verification failed. *Error log* * 8(14180) ERROR: tls [tls_server.c:1193]: tls_read_f(): TLS write:error:14090086:SSL routines:ssl3_get_server_certificate:certificate verify failed 8(14180) ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading 8(14180) DEBUG: <core> [tcp_read.c:1481]: release_tcpconn(): releasing con 0x7f962f6b6db8, state -2, fd=5, id=3* How do I debug and fix this issue. Can somebody help me out. Thanks Austin

2 1

Media timeout using RTPProxy
by Sharath Kumar 01 Jul '15

01 Jul '15

Hello, Is there an API that can notify Kamailio of media inactivity in the RTPProxy? The documentation has something called "timeout_socket" . I don't understand how to use it though. Any help will be greatly appreciated. Thanks Sharath

2 1

How can I figure out the problem why call is not establish?
by zhangkai.gis 01 Jul '15

01 Jul '15

Hello. When I installed kamailio I add users then I can make a call.But When I install asterisk I found the asterisk take over the kamailio and I can register users.But When I make a call.It shows me can not establish the call.How can find the problem?Any logs?Or tools? Thanks, zhangkai

2 1

Fwd: websocket devices keepalive not working
by Yuriy Gorlichenko 01 Jul '15

01 Jul '15

As I read at README to Nathelper it may not work with usrloc table, Is it write? now I use NAT traversal module and Websocket module to keepalive mechanisms, that uses UDP or WS. with UDP it is wery simple method - OPTIONS replies. With WebSocket module keepalive mechanism uses thomething else. Not a SIP methods, I see at TCPDUMP packets that marks with modparam("websocket", "ping_application_data", "WebSockets rock") I need to handle replies of keepaliwe mechanism. How I may handle replies form this mechanism? 2015-06-30 18:37 GMT+03:00 Yuriy Gorlichenko <ovoshlook(a)gmail.com>: > thnks for response. I thing - nathelper is what I need. If I resolve issue > above (why it not work with setbflag(7) ) I thing this questions will be > closed. > > P.S may be it help - my kamailio version is > > version: kamailio 4.4.0-dev1 (x86_64/linux) 1dbd53 > flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, > DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, > F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, > USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > id: 1dbd53 > compiled on 06:24:22 Jun 23 2015 with gcc 4.8.4 > > 2015-06-30 18:18 GMT+03:00 Camille Oudot <camille.oudot(a)orange.com>: > >> Le Tue, 30 Jun 2015 15:49:26 +0300, >> Yuriy Gorlichenko <ovoshlook(a)gmail.com> a écrit : >> >> > 4) I tried to do it with tcpops >> > >> > Only for ws and tcp (WS is use TCP as i know) >> > tcp_keepalive_enable("15", "5", "15"); >> > >> > same result... >> >> Hi Yuriy, >> >> the tcp_keepalive_enable() function will send keepalive packets at TCP >> level, not at application level (SIP OPTIONS). It can be used to check >> if the remote peer connection is still valid, and close the TCP >> connection on the server side if it is no more valid. If you enable the >> TCP keepalives, you should see these packets using tcpdump or wireshark. >> >> Along with "handle_lost_tcp" option on usrloc, this can be used to >> maintain the usrloc table as consistent as possible. >> >> -- >> Camille >> > >

1 0

WWW-Authenticate header Query
by Anil Kumar 01 Jul '15

01 Jul '15

Hi, I am using Kamailio(4.2.5) as my SIP proxy also enabled authentication. Added two users xyz_0 and xyz_1 to the kamailio database. While Registration I am getting 401 Unauthorized as shown below SIP/2.0 401 Unauthorized Via: SIP/2.0/TCP 192.168.1.107:48601;branch=z9hG4bK858339335 From: <sip:xyz_0@192.168.1.107>;tag=251958751 To: <sip:xyz_0@192.168.1.107 <sip%3Axyz(a)192.168.1.107> >;tag=ca5a1db24813f1937965921e320b98e4.6232 Call-ID: 1808202998(a)192.168.1.107 CSeq: 2 REGISTER WWW-Authenticate: Digest realm="192.168.1.107", nonce="VZJ3hlWSdlr0HS/jGQljEKmY20ZrO0Qz" Server: kamailio (4.2.5 (x86_64/linux)) Content-Length: 0 The WWW-Authenticate header is not complete here, May I know the reason for this ? I am expecting algorithm, domain and other stuff in the WWW-Authenticate header. I am not able to proceed further , Please do needful. Thanks

2 1

REST API
by Marek Cervenka 01 Jul '15

01 Jul '15

hello, is there REST API for kamailio for basic functions like subscribers, dispatcher, blacklist/whitelist, ACL, dial plan table, ... ? thank you -- --------------------------------------- Marek Cervenka =======================================

2 1

websocket devices keepalive not working
by Yuriy Gorlichenko 30 Jun '15

30 Jun '15

I try to send keepalive requests (options) to clients at usrloc I tried 4 mechanisms 1) nathelper modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "natping_interval", 15) modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@mydomain") but it didn't send keepalives to noone client 2)nat_traversal I tried to trigger keepalives with nat_keepalive() when REGISTER coming to my service It works fine with UDP packets modparam("nat_traversal", "keepalive_interval", 15) modparam("nat_traversal", "keepalive_method", "OPTIONS") modparam("nat_traversal", "keepalive_from", "sip:pinger@mydomain") modparam("nat_traversal", "keepalive_state_file", "/var/run/keepalive_state") 3)So I also tried to do it with WebSocket keepalive mechanism, but it still not work for (off course I checked only WS devices) modparam("websocket", "keepalive_mechanism", 15) modparam("websocket", "keepalive_timeout", 15) modparam("websocket", "keepalive_interval", 15) 4) I tried to do it with tcpops Only for ws and tcp (WS is use TCP as i know) tcp_keepalive_enable("15", "5", "15"); same result... So I need mechanism to check all devices for keepalive. I think nathelper works with all protos (As I read at docs) Can somebody explain to me what I doing wrong?

3 3

REST API
by Marek Červenka 30 Jun '15

30 Jun '15

hello, is there REST API for kamailio for basic functions like subscribers, dispatcher, blacklist/whitelist, ACL, dial plan table, ... ? thank you -- --------------------------------------- Marek Cervenka =======================================

1 0

interconnect 2 Kamailio server
by kai.ohnacker＠cbc.de 30 Jun '15

30 Jun '15

Hello, after some trying with SEMS and Blox as SBC, I changed my idea of SIP infrastructure from Kamailio (private) -SBC in DMZ (Private & Public) - www (Public) to Kamailio (private) - Kamailio with rtpproxy in DMZ (Private & Public) - www (Public). Now, my questions are: How can I connect 2 Kamailio SIP server together? Which modules I need? Is there somewhere a good howto? Cheers, Kai Ohnacker

2 5

BYE from CANCLEd path gets forwarded to caller
by Daniel Tryba 30 Jun '15

30 Jun '15

I was testing a scenario whith parallel forking (multiple registers) where 2 endpoints answer near simultaniously. When phone that was just a bit slower hangs up, the BYE gets forwarded to the caller and thus the connection is terminated. This is as far as I know for stateful transactions (t_relay()) I always assumed in this scenario the state machine in kamailio would prevent this kind of behavior by itself. Is there a way to prevent these messages being forwarded?

2 2

Which Open Source IVR can be used with Kamailio and how?
by zhangkai.gis 30 Jun '15

30 Jun '15

Hello, Which Open Source IVR can be used with Kamailio and how? Thanks, zhangkai

2 8

invalid char in json text
by Kelvin Chua 30 Jun '15

30 Jun '15

Playing around with ndb_mongodb getting this error: ERROR: ndb_mongodb [mongodb_client.c:262]: mongodbc_exec_cmd(): Failed to run command: lexical error: invalid char in json text query is very simple: db.cdrs.insert({time:new Date(),src_user:"kelvin",term_ani:"14112",src_domain:"domain.com ",src_ip:"0",dst_ouser:"18188",dst_user:"18112",dst_domain:"domain.com"}) which is inside a pv mongodb_cmd("mgs1", "datab", "cdrs", "$var(MNG_MISSED)", "mgr1"); tried this: insert({time:new Date(),src_user:"kelvin",term_ani:"14112",src_domain:" domain.com",src_ip:"0",dst_ouser:"18188",dst_user:"18112",dst_domain:" domain.com"}) and other combinations, same error any ideas? i guess we need some more examples on the documentation Kelvin Chua

3 7

Siremis Register Page problem.
by Sven Juhkam 29 Jun '15

29 Jun '15

Hi! In SIREMIS I am getting the info after trying to register, that "Public registration is not enabled!" but i have set siremis/modules/ser/config/common.Main.php to $cfg_siremis_public_registrations = true; What is the problem? Thanks and Greetings, Sven

1 0

answer back from wrong IP / Interface
by Mihaly Zachar 27 Jun '15

27 Jun '15

Hi Gents, We have a redirect server based on Kamailio (4.2.3). The server has multiple IPs, but Kamailio is bound to only one address: http://view.xscreenshot.com/9dfdf449a387bdc6f9a719eae3620416 *Sometimes* it answers back from a wrong IP address, here is the proof: http://view.xscreenshot.com/ac661d6b303eda6653b37be7cf65aa65 The interesting thing is that if I do: root@tralab-route-01:~# ip ro get 10.121.24.51 10.121.24.51 via 10.121.43.30 dev vlan486 src 10.121.43.17 cache ipid 0x05ad root@tralab-route-01:~# So the 10.121.43.17 IP is selected, but I *do* expect that Kamailio send back the answer from the same socket where it got the request. What could be the issue ? Why does it do that *sometimes* ? In the majority of the time it works well.. I am not sure why this happens... Thanks, Mitya

1 1

don't fragment flag in rtpengine request
by Juha Heinanen 27 Jun '15

27 Jun '15

i was wondering why sometimes requests to rtpengine don't get through and icmp unreachable destination error message is received. it turned out that in these error cases ip packet containing the rtpengine request has 0x02 (don't fragment) flag set even when the packet was bigger than 4 KB. in this case kamailio was running in an amazon container which may be doing something weird. is it somehow possible from the application (= kamailio) to control setting of that flag? -- juha

1 0

deb building system
by Victor Seva 26 Jun '15

26 Jun '15

Hello, I would like to inform that the deb nightly builds are going to be disabled. I'm having problems with the actual configuration and until I found the time to fix them the service will be out of order. No new debs will be produced until further notice. Cheers, Victor

2 1

limit calls to incoming only?
by Peter Villeneuve 26 Jun '15

26 Jun '15

Hi, Is it possible for Kamailio to restrict certain users (like in a group) to only receive calls but never dial out or send SIP Simple messages outbound, while allowing all other "regular" users to make outgoing calls and text messages as well as receive them? If so, which module do you guys recommend I look into? Cheers, Peter

2 3

fix_nated_contact NAT rtpproxy problems
by Vik Killa 25 Jun '15

25 Jun '15

Hello, We are trying to get Kamailio working as an SBC using rtpproxy(); The config seems to work except with some carriers, kamailio does not recognize the ACK because it does not match the contact header of the OK I've been trying to figure out how to get kamailio to send the OK back using the public IP address but I cannot figure out how. I've tried using various deployments of the nathelper module such as fix_nated_contact() set_contact_alias() No matter what though, Kamailio will not change the contact header it receives from FS before sending it back to the carriers. This has become frustrating as I read tutorials that showed this is possible, yet I am unable. Thank you. /V

1 0

"00" prefix record problem on ACC
by Yasin CANER 25 Jun '15

25 Jun '15

3 2

linking math.h while compiling a module
by Shouvanik Chakrabarti 25 Jun '15

25 Jun '15

Hello, I have developed a Kamailio module to handle group messaging and am using certain functions from the libm.so library. Compilation happens fine, but Kamailio subsequently fails to start because the math.h symbols are unlinked/undefined. Could someone please suggest a workaround for this. Thanks, Shouvanik.

2 4

How to disable retransmits via TCP connection? etc.
by Andrey Utkin 25 Jun '15

25 Jun '15

Hi! We meet such an issue with some kinds of client internet connections (e.g. reproduces always on local wifi, but doesn't repr. on 3g inet). We don't know what are exact network characteristics, but what happens from perspective of traffic on SIP server is the following: 1. the SIP client registers successfully to Kamailio (which is recent git master HEAD) with TCP transport for SIP; 2. another client calls this user; 3. Kamailio relays INVITE, it gets transmitted fully and ACK-ed on TCP level; 4. then, within 0.1-0.2 second, Kamailio fires first retransmit of that INVITE, which doesn't get fully transmitted (it is shown trunkated in sniffer output, no TCP ACK replies for it); 5. Kamailio tries to retransmit INVITE again, with the same result as in #5. 6. The caller gets SIP/2.0 408 Request Timeout. A piece of traffic from "ngrep -t -e -d any -W byline port 5060": https://gist.githubusercontent.com/krieger-od/219f9975e5efb980ff5b/raw/096a… The called side app is based on mobile Linphone app. Switching to UDP is not an option (in some networks SIP messages get delivered trunkated which breaks calls), we will check how it works with TLS transport a bit later when there's technical possibility. I have two questions: 1. How would I completely disable retransmissions to TCP connections? 2. Any ideas what can be the reason for this issue? Retransmissions by themselves? -- Andrey Utkin

4 11

Kamailio Server Maintenance - Wed-Thu Night
by Daniel-Constantin Mierla 25 Jun '15

25 Jun '15

Hello, during the night between Wed (June 24) and Thu (June 25), planned to start not early than 00:00 GMT+1, there will be some schedule maintenance work to the infrastructure that is hosting some of the kamailio.org servers. The main affected services will be: - main website (www.kamailio.org) - wiki portals - mailing lists (lists.sip-router.org) - git mirror (git.kamailio.org) It is expected to have short downtimes of few minutes. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

1 1

Kamailio + RTPEngine + Asterisk
by Javier Aristizábal 24 Jun '15

24 Jun '15

Hello folks, I am trying to configure this on the same server: Kamailio v4.2 port 5060 RTPEngine v2.3 port 22222 Asterisk v11.18.0 port 5080 I am follow the Asipto tutorials, for the Kamailio and the real time configuration. Now, I can talk between extension and PSTN without issues. But the trouble is with the Status registration with Asterisk. And I need that Status OK because I am using queues in asterisk for call centre. I got this from with: sip show peers Name/username Host Dyn Forcerport Comedia ACL Port Status Description Realtime 101 (Unspecified) D Yes Yes 0 UNREACHABLE Cached RT the status is UNREACHABLE and the asterisk table sipregs won't update after each registration. To be sure, I tried first all config only with asterisk to check that is not a problem with MySQL or something. On my cfg file I have this for the asterisk registration: route[REGFWD] { if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } Any clue? Thanks! -- Javier Aristizábal

2 16

Kamailio proxy at Amazon EC2 (behind NAT)
by Alexandru Covalschi 24 Jun '15

24 Jun '15

Hello. I'm trying to set up this (v 4.2 stable): peer <--> ec2 <--kamailio+rtpengine--> asterisk scheme I use advertised adress for SIP and WS connections. The problem is that on SIP I get one way audio - I can receive audio from asterisk, but I can't transmit audio there - my SIP UA tries to send data to Kamailio-s local EC2 IP. In case of WebRTC I get lot's of erros: Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: WARNING: <core> [msg_translator.c:2778]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [msg_translator.c:1996]: build_req_buf_from_sip_req(): could not create Via header Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: <core> [forward.c:584]: forward_request(): building failed Jun 23 01:58:57 kamailio /usr/sbin/kamailio[18325]: ERROR: sl [sl_funcs.c:387]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) The call reaches Asterisk, but not vice-versa. No media is being transferred. Rtpengine flags I use: For SIP: rtpengine_manage("trust-adress replace-origin replace-session-connection RTP/AVP"); For WS: rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force RTP/AVP"); Do you have any ideas how ti fix that? I also make REGFWD's to Asterisk -- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

2 15

Kannot start kamailio 4.3
by Yuriy Gorlichenko 23 Jun '15

23 Jun '15

Try to start kamailio on Ubuntu 14.04.02 Get this errors ERROR: ctl [init_socks.c:115]: init_unix_sock(): ERROR: init_unix_sock: bind: No such file or directory [2] Jun 23 08:54:29 kamailio-test kamailio[3706]: ERROR: ctl [ctl.c:273]: mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed Jun 23 08:54:29 kamailio-test kamailio[3706]: ERROR: <core> [sr_module.c:945]: init_mod(): Error while initializing module ctl (/usr/local/lib64/kamailio/modules/ctl.so) my params for ctl # ----- ctl params ----- modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") version: kamailio 4.4.0-dev1 (x86_64/linux) 1dbd53 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 1dbd53 compiled on 06:24:22 Jun 23 2015 with gcc 4.8.4

4 9

kamailio fails after start
by Yuriy Gorlichenko 22 Jun '15

22 Jun '15

Hello. I Installed kamailio on ubuntu 14.04 that runs as virtual systemOpenVZ. after starting kamailio I see that it runs ok with kamailio start or kamctl start at ps -ax I see working processes. But after one minute of working kamailio fails with kamailio: ERROR: <core> [daemonize.c:315]: daemonize(): Main process exited before writing to pipe I see nothing more at debug file. No errors. (I run with debug=4) my cfg parameters at start mhomed=1 memdbg=5 memlog=5 fork=yes children=6 version: kamailio 4.2.4 (x86_64/linux) ac3682 flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: ac3682 compiled on 04:27:04 Apr 10 2015 with gcc 4.8.2

2 2

Kamailio transformation
by Ali Taher 22 Jun '15

22 Jun '15

Hello, Suppose I have $var(a) = 1234#5678 and $var(b)=123455667 I need to get the number before '#' character if exists , meaning that from $var(a) I must get 1234# and from $var(b) I must get empty string. I tried $var(pref)= $(var(a){s.select,0,#}); but I'm getting the whole string if # doesn't exists. There is a way to check if # exists by subtracting the length of the string before and after removing # character and check if length changes , but I want a cost-less method. Any help would be appreciated. Thanks Ali

3 2

Capture a Match from Regex comparison
by Sebastian Damm 22 Jun '15

22 Jun '15

Hi, we have some cases, where we get multiple PCMA lines with different numbers from a peer. One of our backend systems doesn't handle that correctly. So what I want to do is detect whether there is a PCMA line with a non-standard number in the SDP. This is how an SDP looks like: v=0 o=hiQ9200 2990220150519102450 1229717573 IN IP4 1.2.3.4 s=Phone Call c=IN IP4 1.2.3.4 t=0 0 m=audio 40174 RTP/AVP 8 0 18 96 97 13 100 98 99 a=rtpmap:8 PCMA/8000 a=fmtp:8 vad=no a=rtpmap:0 PCMU/8000 a=fmtp:0 vad=no a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:96 G726-32/8000 a=rtpmap:97 AAL2-G726-32/8000 a=rtpmap:100 telephone-event/8000 a=fmtp:100 0-15 a=rtpmap:98 PCMA/8000 a=gpmd:98 vbd=yes a=rtpmap:99 PCMU/8000 a=gpmd:99 vbd=yes a=sqn: 0 a=cdsc: 1 image udptl t38 a=sendrecv a=pmft: T38 a=ptime:20 I tried something like that (expecting that the number doesn't change): if (sdp_get_line_startswith("$avp(badcodec)", "a=rtpmap:98 PCMA")) { But this makes Kamailio drop the INVITE when arriving at the line in the config. I guess, "startswith" stops parsing the line at the colon. So I thought I'll make it more generically: sdp_get("$avp(sdp)"); if ($avp(sdp) =~ "a=rtpmap:([1-9][0-9][0-9]?) PCMA") { sdp_remove_codecs_by_id("$missingvariable"); } What I need is to capture the number in the a=rtpmap line so I can delete the codec by ID. But I couldn't find a way to capture the matches from my regex comparison. Am I missing something? (Other approaches for my problem would be welcome, too.) Thanks in advance. Best Regards, Sebastian

2 3

Kamailio with Blox SBC (opensips based) - Wrong CSeq
by kai.ohnacker＠cbc.de 22 Jun '15

22 Jun '15

Hello, my setup is a kamailio server as registrar and Blox as SBC. Blox is a freeware GUI with Opensips as framework. Kamailio is located in the private network and Blox operates with 2 NICs (private and public) Now the following problem: If I make a call from public (phonerlite) to private (mayah), the connection don’t reach the “framed” state and is “broken” after 30 sec. After evaluating the log files, I found that the CSeq from Kamailio is wrong. The invite comes from CSeq: 9 INVITE The reply is with CSeq 1 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.3:6060;branch=z9hG4bK38ae.cf317344.0 Max-Forwards: 70 To: <sip:1001@192.168.1.3:8000>;tag=qTKGZvr0ItW From: "PhonerLite" <sip:1000@192.168.1.3:8000>;tag=1005520878 Call-ID: SBCbFgsQn5HcHpEUxJUVF8gQWVZREt6QwJvWVJkXCdAUzdkXl41CUJ4cEdWZV1NQ0tFYQ CSeq: 1 Contact: <sip:1001@192.168.0.33:5060> Record-Route: <sip:192.168.0.1;lr=on> User-Agent: MAYAH 4.9.12.0-2.1.0.45 Content-Length: 0 And here the information from the opensips log files: Jun 22 14:30:32 localhost blox-0-9-6-beta[1975]: ERROR:core:parse_cseq: no method found Jun 22 14:30:32 localhost blox-0-9-6-beta[1975]: ERROR:core:parse_cseq: bad cseq Jun 22 14:30:32 localhost blox-0-9-6-beta[1975]: ERROR:core:get_hdr_field: bad cseq Perhaps someone of you knows this failure or can tell me how to solve this problem? Cheers, Kai

3 4

Redirect with path
by Kelvin Chua 22 Jun '15

22 Jun '15

on my registrar server: after calling lookup, and getting AOR with path, how do I send the path info back as a redirect? Kelvin Chua

2 2

event_route: what modules/functions are likely to work or not work
by Armen Babikyan 22 Jun '15

22 Jun '15

Hello, With respect to *_route blocks in kamailio configs, I gather that event_route is the newest type of route in the config vocabulary. However, many modules have functions that specifically state that they work within certain routes, like REQUEST_ROUTE and FAILURE_ROUTE, with the implication that they won't work in other routes. For example, there are some functions that "work" in EVENT_ROUTE as well, and that this could be a matter of either documentation not being updated yet, or funny/bad behavior I may encounter at some future point in time... Generally speaking, are there any general statements that can be made about what "common" modules/functions are definitely only available within request_route and onreply_routes? event_route blocks, in particular, elucidate many other use cases for several modules' functions that I'm sure the original authors didn't think of, and it would be great to have a high-level commentary/discussion like "module XYZ or modules that touch A, B, and C don't have a snowball's chance of working in an event_route". Comments are welcome. Thanks! Armen

2 1

Installation problem, kamailio 4.3.0 on ubuntu 14.04.02
by Edin Fejzić 22 Jun '15

22 Jun '15

I just installed kamailio 4.3.0 with apt-get. But it won't start. mysql and /etc/kamailio/kamctlrc done. Minimum change to kamailio.cfg Any idea? syslog output: /usr/sbin/kamailio[1291]: INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob /usr/sbin/kamailio[1291]: INFO: rr [rr_mod.c:174]: mod_init(): outbound module not available /usr/sbin/kamailio[1291]: INFO: usrloc [hslot.c:51]: ul_init_locks(): locks array size 1024 /usr/sbin/kamailio[1291]: ERROR: ctl [init_socks.c:115]: init_unix_sock(): ERROR: init_unix_sock: bind: No such file or directory [2] /usr/sbin/kamailio[1291]: ERROR: ctl [ctl.c:273]: mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed /usr/sbin/kamailio[1291]: ERROR: <core> [sr_module.c:945]: init_mod(): Error while initializing module ctl (/usr/lib/x86_64-linux-gnu/kamailio/modules/ctl.so) kernel: [ 13.978764] init: kamailio main process (1291) terminated with status 255

2 1

Kamailio invite counter
by Ali Taher 19 Jun '15

19 Jun '15

Hello, I want to define a counter that increase in every invite. I tried the below : if (is_method("INVITE")) { $var(u)=$var(u)+1; xlog("L_INFO","Counter : $var(u) " ); } But the counter is reset to 1 randomly (even before restarting kamailio service) How can have a proper counter that count invite messages. Thanks Ali

4 7

Port 5061 issue
by Ankhit Vivekananda 19 Jun '15

19 Jun '15

Hello, I have configured kamailio 4.0.4 long back. Now, I wanted to enable the usage of port 5061 with TLS. By doing so, the client(video camera) can place outgoing calls fine. But while receiving an incoming call, the message does not get sent out to the other client. Hence, the call fails. I can place and receive calls on port 5060 successfully without any issues. Can anyone please tell me why port 5061 is giving me a problem. Also, as a test, I set up kamailio on another machine(version 4.1) and 5061 works fine in all directions. Its only on the other machine that I have problems with. Appreciate any help! Thank you, Ankhit

2 1

SIP-I trunking
by Jérôme Simionato 19 Jun '15

19 Jun '15

Hello, I'm planning to interconnect kamailio with a SIP-I provider (an i have no more information at this time from my telco). I read the documentation about the SIP-I module, who need to be completed to match our needs. basically, we need to read the location parameter for incoming call, and be able to set redirecting number for outgoing call Does anybody use this module for SIP-I ? Have you experiences of SIP-I trunking with telco ? And another question: when you make an ISUP (SS7) interconnection with a telco, you need to pass specific test to validate the correct implementation of the SS7 protocol (and it's national variant, eg. SPIROU in france). Is it the same thing on an SIP-I interconnection ? Any information about it will be appreciate, Cheers, Jérôme.

2 3

SIP Ping a device using Kamailio
by Nelson Migliaro 19 Jun '15

19 Jun '15

Hello, Is there a way to ping a device using Kamailio. In my scenario I need to access a Kamailio located in my DMZ over a firewall that is only open in LAN > DMZ direction. It is close in the other way round. In my current configuration I keep the link open using Asterisk qualify option in SIP. I would like to replace Astersik located in LAN with Kamailio thus I need to keep the link open. I know there is an option using dispatch but I would like to avoid using this module in order to keep it simple. Is there a "simple embedded function" in order to accomplish this. thank you

2 1

WARNING: tm [t_lookup.c:1537]: t_unref(): WARNING: script writer didn't release transaction
by Roberto Fichera 19 Jun '15

19 Jun '15

Hi All, I'm getting the warning in the subject after replacing a route() with a async_task_route(). In the called route() I'm doing some db query and send notify messages using the uac module. The async_task_route() is called once received a register message. Any idea what's is going on? Cheers, Roberto Fichera.

2 3

users with or without asterisk integration on the same kamailio database
by asterisk asterisk 19 Jun '15

19 Jun '15

I wish to know how I can have two kinds of users 1. some users with real time integration with asterisk 2. other users do not How can I achieve in the same kamailio instance?

3 3

Dispatcher and options_reply
by jay binks 19 Jun '15

19 Jun '15

Hello, So Im using dispatcher to poll SBC's behind my kamailio load balancer. What I need to do is get a count of how many "active" SBC's there are in the dispatcher module. ( how many SBC's are actively responding to their options ping ) so that I can then use this info when I reply to an OPTIONS on the load balancer. effectively , if there are no SBC's active, the LB needs to not reply to options. I COULD use ds_select_dst or something , but is that going to be overkill from a CPU perspective ? ( And I dont want ds_select_dst to do any modification to my otpions packets ) just wandering what everyone thinks the best way to achieve this would be. -- Sincerely Jay

2 1

siremis with postgres
by Tomas Zanet 19 Jun '15

19 Jun '15

Hi all, I need to install Kamailio, and Siremis as well, with Postgres backend. My current versions are the following: - Kamailio 4.2.1 - Siremis 4.2.0 - PHP 5.3.10-1ubuntu3.18 - postgres 9.1.3 To do so, I installed libpq-dev and compiled Kamailio with db_postgres module. After that, I create the Kamailio database using kamdbctl create No problems so far....while I’m facing some problem when I tried to install Siremis with Postgres. To do so, I installed php5-pgsql before and then I did the steps described here: http://kb.asipto.com/siremis:install42x:main#dokuwiki__top After clicking on “Setup Now” and “Next” buttons, I’m facing with the following problem: ERROR: SQLSTATE[08006] [7] FATAL: password authentication failed for user "www-data" FATAL: password authentication failed for user "www-data" Though the username I specified into web page is not www-data, but siremis. As far as I know www-data is a Debian user name under which Apache/PHP is running. Can anyone help me please? Thanks in advance Best Regards, Tomas Tomas Zanet Software Design Department tzanet(a)came.com CAME S.p.A.

2 4

kamcmd
by asterisk asterisk 18 Jun '15

18 Jun '15

I am compiling the kamailio 4.3 in raspberry Pi 2. It compiled and worked. However, kamcmd did not work as it looks for the file /var/run/kamailio/kamailio.ctl which is not present. Similar issue found for kamctl but I fixed it by change the /tmp directory to /var/run/kamailio Did I miss something?

2 1

404 Not found error when login Siremis-4.2.0
by Chathura Senanayake 18 Jun '15

18 Jun '15

Hello, I need a help with siremis-4.2.0. I have installed Kamailio-4.3 and have it running. Then I installed Siremis-4.2.0. When I try to login to the Siremis by using login name "admin" and password "admin", it shows a 404 Not found error message as follow. Not Found The requested URL /siremis/system/general_default was not found on this server. ------------------------------ *Apache/2.4.7 (Ubuntu) Server at 172.16.2.9 Port 80* I have run the #a2enmod > rewrite and Restart the apache2. But the problem is still not solved. Please let me know how to resolve this. *Chathura Senanayake* Associate Software Engineer (DA) hSenid Mobile Solutions *Head Office* (Singapore)*P*: +65 65 332 140*M*: *F*: +65 62 910 023 *R&D* (Sri Lanka)*P*: +94 11 268 3751*M*: +94 71 757 9646*F*: +94 11 268 3951 <http://www.hsenidmobile.com/> <http://www.hsenidmobile.com/> <https://www.facebook.com/hSenidMobileSolutions> <https://twitter.com/hSenidMobile> <http://www.linkedin.com/company/hsenid-mobile-solutions-pvt-ltd> <https://plus.google.com/u/0/116242841237814090103/posts> <http://www.youtube.com/user/hSenidmobile>*www.hSenidMobile.com* <http://www.hsenidmobile.com/>

3 7

Kamailio number of matches of specific character in a string
by Ali Taher 18 Jun '15

18 Jun '15

Hello, I just need to count the number of a specific character in a string. For example I want to count the number of "a" in the following string : "bbbaddaa34" I searched among transformations but didn't find such tool. Any help would be appreciated. Thanks Ali

2 1

using uac_replace_from( )
by Al S 18 Jun '15

18 Jun '15

Hi, I am trying to change only the "display name" of the FROM header using uac_replace_from().However, the value is not being changed (see below) request_route { .... xlog ("L_INFO", "fu=$fu"); uac_replace_from("pbx", ""); xlog ("L_INFO", "fu=$fu"); # authentication route(AUTH); ....} Am I missing anything? Any help in this regard would be greatly appreciated. Thanks,Al

3 2

Re: [SR-Users] loose_route to find transport=tcp
by Ryan Kendrick 17 Jun '15

17 Jun '15

After enabling and deciphering debugging it appears there may be a bug. I also reviewed https://tools.ietf.org/html/rfc5658#section-6 I cross-referenced my pcap to ensure I was looking at the reINVITE and see: Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: rr [loose.c:785]: after_loose(): *Topmost route URI: 'sip:xx.xxx.x.xx;lr;r2=on;ftag=a30a720a;did=b75.65a1;nat=yes' is me* Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==11 && [xx.xxx.x.xx] == [xx.xxx.x.xx] Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5060 (advertise 0) matches port 5061 Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==11 && [xx.xxx.x.xx] == [xx.xxx.x.xx] Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5070 (advertise 0) matches port 5061 Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==11 && [xx.xxx.x.xx] == [xx.xxx.x.xx] Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5090 (advertise 0) matches port 5061 Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 11==11 && [xx.xxx.x.xx] == [xx.xxx.x.xx] Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 5061 (advertise 0) matches port 5061 Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: rr [loose.c:181]: find_next_route(): *No next Route HF found* Jun 16 15:01:29 xxxxxxxxxxxxx /usr/sbin/kamailio[643]: DEBUG: rr [loose.c:847]: after_loose(): no next URI found There is definitely another Route header immediately below the one found above, but find_next_route() doesn't find it . I added my own debugging to loose.c and if ((_m->last_header->type!=HDR_ROUTE_T) || (_m->last_header==*_hdr)) { LM_DBG("No next Route HF found\n"); LM_DBG("_m->last_header->type: %d\n", _m->last_header->type); return 1; } logs find_next_route(): _m->last_header->type: 12 which is HDR_CONTENTLENGTH_T which is indeed the LAST header in the message. We have done very little work in the Kamailio source...just some database escaping in odbc for things to work properly with our database engine...but unless I'm missing something isn't it very wrong to be looking at the last header right here? I may attempt to figure out the message and/or hdr_field data structures and change it. It may also be that the issue doesn't occur when find_next_route is called with a valid _hdr which does seem to search for the "next" one vs going straight to the final header in the entire message. If this is getting overly complicated for this mailing list please let me know... Ryan On Tue, Jun 16, 2015 at 11:40 AM, Ryan Kendrick <kendrick.ryan.c(a)gmail.com> wrote: > > We are using Kamailio 4.2.5 as a registrar and proxy between many > dispersed end-users of a soft phone app and our calling platform / switch. > > Until now we have used udp exclusively but are trying to introduce tcp > between end-users and Kamailio, leaving udp between Kam and our > switch...while maintaining the ability for some end-users to use udp to Kam. > > With some simple address checks I am able to always send to our switch > over udp. If all end-users used tcp I could send everything else tcp, but I > need to maintain udp support. > > The specific problem I am having is on a reINVITE such as this one from > our platform to the a-leg: > > INVITE sip:xxxxxx@xxxxxxxxxxxxx:42679;user=phone SIP/2.0 > Via: SIP/2.0/UDP > xxxxxxxxxxxxx:5060;branch=z9hG4bK218cc8e12ll5035f67INV6a67885312aad > Max-Forwards: 35 > Route: <sip:xxxxxxxxxxx;lr;r2=on;ftag=daba971c;did=b57.4872;nat=yes> > Route: > <sip:xxxxxxxxxxx:5070;transport=tcp;lr;r2=on;ftag=daba971c;did=b57.4872;nat=yes> > Contact: <sip:xxxxxxxxxx@xxxxxxxxxxxxx:5060> > To: "xxxxxx"<sip:xxxxxx@xxxxxxxxxxxxxxxxxxxxxxxxxxx:5070>;tag=daba971c > From: <sip:xxxxxxxxxx@xxxxxxxxxxxxxxxxxxxxxxxxxxx > :5070>;tag=6a678853-co76461-INS002 > Call-ID: MDI4ZmFjNmZhN2Y1NWE2ZTViNTkyZGUwNWE2YzUzYmU > CSeq: 7646101 INVITE > Allow: > INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,INFO,UPDATE > Content-Type: application/sdp > Date: Mon, 15 Jun 2015 20:10:18 GMT > User-Agent: xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx > Content-Length: 262 > > As you might notice, we have rr:enable_double_rr set: > > *There are some situations when the server needs to insert two > Record-Route header fields instead of one. For example when using two > disconnected networks or doing cross-protocol forwarding from UDP->TCP. > This parameter enables inserting of 2 Record-Routes. The server will later > remove both of them. * > > and I believe it is necessary to keep this way. Without it Kamailio > doesn't even see the reINVITE...the switch probably tries tcp and that's > not setup between the two. > > The invite above is sent to the a-leg over udp but I would expect and need > it to be tcp in this case. The reINVITE is part of an existing dialog. We > call loose_route() followed by some simple bflag setting and flag checking, > t_on_reply(), ... then t_relay(). > > I do have a functional workaround but would prefer to avoid such manual > handling by utilizing built-in functionality properly. > > # > # relay the message > # > if(route(TEST_TOGW)) { > if (!t_relay_to_udp()) { > sl_reply_error(); > } > } > else { > if ($(hdr(Route)[-1]) =~ "tcp") { > if(!t_relay_to_tcp()) { > sl_reply_error(); > } > } > else if (!t_relay()) { > sl_reply_error(); > } > } > > I'm not 100% sure how reliable or fast this will be, but it does work so > far in my simple tests. > > Is loose_route supposed to see and use the transport=tcp but isn't for > some reason? It seems like the right thing to do to me. If not, is there > anything else I can/should be doing in the tm and/or rr modules to make > Kamailio realize it needs to send this message over TCP? If not in those > two modules is there some recommended way perhaps via registrar or usrloc > etc. to make Kamailio remember/store when a user is connected via TCP and > be able to do a quick lookup before sending to them? Anything else I'm > missing or not thinking of? > > Please let me know if I can further explain and rest assured any > assistance will be much appreciated!!! >

2 2

Performance Tests/Hard Recommendations kamailio 4.2 -200CPS
by Jonathan Hunter 17 Jun '15

17 Jun '15

Hi Guys, Has anyone recently performed tests on kamailio 4.2 running up to 200 CPS? It would be great to know the hardware requirements for this setup. Thanks Jon

5 8

siptrace: how to log sent ACK within dialog
by Seudin Kasumovic 17 Jun '15

17 Jun '15

Hi, TM forward ACK statelessly (see log below), so there is no call of installed callback on TMCB_REQUEST_(IN)SENT in siptrace module, so I can't see way how to log ACK sent packet. Any work around to get sent ACK logged? DEBUG: sl [sl_funcs.c:388]: sl_filter_ACK(): DEBUG : sl_filter_ACK: to late to be a local ACK! DEBUG: <script>: 408:172.16.23.130:5060 <-- 198.51.100.137:52150 ACK sip:8807@127.0.0.1:5080;transport=udp DEBUG: <script>: 425: start trace ACK DEBUG: siptrace [siptrace.c:1865]: pipport2su(): the port string is 52150 DEBUG: siptrace [siptrace.c:1865]: pipport2su(): the port string is 5060 DEBUG: siptrace [siptrace.c:1677]: trace_send_hep_duplicate(): setting up the socket_info DEBUG: <script>: 542: WITHINDLG: loose_route set du='sip:172.16.23.130:5070 ;lr;ftag=1029663405' DEBUG: <script>: 369: RELAY ACK sip:8807@127.0.0.1:5080;transport=udp [4194308] DEBUG: tm [t_lookup.c:1312]: t_newtran(): DEBUG: t_newtran: msg id=4 , global msg id=3 , T on entrance=0xffffffff DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=34522, isACK=1 DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: *RFC3261 transaction matching failed* DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found DEBUG: tm [t_funcs.c:279]: t_relay_to(): SER: forwarding ACK statelessly DEBUG: <script>: 821:172.16.23.130:5060 --> 172.16.23.130:5070 ACK sip:8807@127.0.0.1:5080;transport=udp Regards, Seudin Kasumovic

2 2

Getting 407 response when calling out (on bulk registration)
by Al S 17 Jun '15

17 Jun '15

Hi, A bulk registration has been added to the "location" table:*************************** 1. row *************************** id: 113 ruid: uloc-55808d5a-33bc-1 username: pbx domain: NULL contact: sip:pbx@10.10.10.157:5060;bnc received: NULL path: NULL expires: 2015-06-17 10:08:42 q: -1.00 callid: 1-15061(a)10.10.10.157 cseq: 2last_modified: 2015-06-17 09:08:42 flags: 0 cflags: 0 user_agent: SIPp/Win32 socket: udp:10.10.10.160:5060 methods: NULL instance: NULL reg_id: 01 row in set (0.00 sec) An outgoing call (SIPOUT) is made from a 10-digit number of the pbx: Request-Line: INVITE sip:9191111111@example.net:5060 SIP/2.0From: sipp <sip:8181111111@home.net>;tag=1To: <sip:9191111111@example.net:5060>Proxy-Authorization: Digest username="pbx",realm="myhome.net",uri="sip:10.10.10.160:5060",nonce="???",response="??",algorithm=MD5 Even though I changed the value of $fU to "pbx", Kamailio does not authenticate the username and response with a 407 "Proxy Auth Required" response: route[AUTH] {... if (is_method("REGISTER") || from_uri==myself) { # authenticate requests $var(fU) = $fU; $fU = "pbx"; if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } ... $fU = $var(fU); #!endif return;} Is this the correct approach to process outgoing call for bulk registration username?Any help in this regard would be greatly appreciated Thanks,Al

1 0

kamailio sending 408 request time out
by Priyaranjan Nayak 17 Jun '15

17 Jun '15

Hi All, I am using kamailio-4.2.5 for testing setup. After starting the kamailio I am getting below message /usr/local/sbin/kamailio -f /usr/local/etc/kamailio/kamailio.cfg -N 20 -m 512 -M 8 /usr/local/sbin/kamailio -f /usr/local/etc/kamailio/kamailio.cfg -E loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(24471) WARNING: <core> [socket_info.c:1392]: fix_hostname(): could not rev. resolve 192.168.1.118 0(24471) WARNING: <core> [socket_info.c:1392]: fix_hostname(): could not rev. resolve 192.168.1.118 Listening on udp: 192.168.1.118:4060 tcp: 192.168.1.118:4060 Aliases: 0(24471) INFO: <core> [tcp_main.c:4748]: init_tcp(): using epoll_lt as the io watch method (auto detected) 0(24473) INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob 0(24473) INFO: rr [rr_mod.c:160]: mod_init(): outbound module not available 0(24473) INFO: usrloc [hslot.c:53]: ul_init_locks(): locks array size 1024 0(24473) INFO: <core> [udp_server.c:175]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(24473) INFO: <core> [udp_server.c:225]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 8(24481) INFO: ctl [io_listener.c:225]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) Once I sent the request to kamailio server, I am not getting 408 request time out from kamailio and getting below log 12(24485) ERROR: <core> [tcp_read.c:299]: tcp_read_data(): error reading: Connection reset by peer (104) 12(24485) ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading 9(24482) ERROR: <core> [tcp_read.c:299]: tcp_read_data(): error reading: Connection reset by peer (104) 9(24482) ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading 12(24485) ERROR: <core> [tcp_read.c:299]: tcp_read_data(): error reading: Connection reset by peer (104) 12(24485) ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading 10(24483) ERROR: <core> [tcp_read.c:299]: tcp_read_data(): error reading: Connection reset by peer (104) 10(24483) ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading 13(24486) ERROR: <core> [tcp_main.c:4162]: handle_tcpconn_ev(): connect 192.168.1.114:61945 failed 13(24486) ERROR: <core> [tcp_main.c:4162]: handle_tcpconn_ev(): connect 192.168.1.114:19129 failed 13(24486) ERROR: <core> [tcp_main.c:4162]: handle_tcpconn_ev(): connect 192.168.1.114:6050 failed Could you please give me any idea how to resolve this issue ? Thanks Priyaranjan

2 1

http_query failed to perform curl (28) errors
by Armen Babikyan 17 Jun '15

17 Jun '15

Hello, I am running across the same issue as described here: http://lists.sip-router.org/pipermail/sr-users/2015-February/086796.html Was there a resolution to this issue? I come across this issue fairly regularly. Unlike the OP, however, I do not have any luck by retrying my query: when I place my http_query within a while loop and keep 1-second delays between calls, all consistently fail with error 28. After long pauses, http_query()s start to work as expected. I think that previous thread mentions DNS as a possible culprit. Is http_query using libcurl's preferred resolver, or is it using Kamailio's built-in DNS cache? I really doubt my local resolver is cause of the issue here. Any insight is appreciated. Thanks! Armen

2 1

Not found error when Siremis login
by Chathura Senanayake 17 Jun '15

17 Jun '15

Hello, I need a help with siremis-4.2.0. When I try to login the the Siremis, it shows a 404 Not found error message as follow. Not Found The requested URL /siremis/system/general_default was not found on this server. ------------------------------ *Chathura Senanayake* Associate Software Engineer (DA) hSenid Mobile Solutions *Head Office* (Singapore)*P*: +65 65 332 140*M*: *F*: +65 62 910 023 *R&D* (Sri Lanka)*P*: +94 11 268 3751*M*: +94 71 757 9646*F*: +94 11 268 3951 <http://www.hsenidmobile.com/> <http://www.hsenidmobile.com/> <https://www.facebook.com/hSenidMobileSolutions> <https://twitter.com/hSenidMobile> <http://www.linkedin.com/company/hsenid-mobile-solutions-pvt-ltd> <https://plus.google.com/u/0/116242841237814090103/posts> <http://www.youtube.com/user/hSenidmobile>*www.hSenidMobile.com* <http://www.hsenidmobile.com/>

1 0

Kamailio Crash
by Marc Soda 17 Jun '15

17 Jun '15

Is it normal for Kamailio to segfault on a duplicate key in the usrloc DB? Jun 15 18:15:23 vproxy2 /usr/sbin/kamailio[13646]: ERROR: db_mysql [km_dbase.c:122]: db_mysql_submit_query(): driver error on query: Duplicate entry 'uloc-557a96c1-3577-4e5f1' for key 'ruid_idx' Jun 15 18:15:23 vproxy2 /usr/sbin/kamailio[13646]: ERROR: <core> [db_query.c:337]: db_do_update(): error while submitting query Jun 15 18:15:23 vproxy2 /usr/sbin/kamailio[13646]: ERROR: usrloc [ucontact.c:800]: db_update_ucontact_addr(): updating database failed Jun 15 18:15:23 vproxy2 /usr/sbin/kamailio[13646]: ERROR: usrloc [urecord.c:368]: wb_timer(): updating contact in db failed (aor: sip305_isibmp) I believe that there should never be a duplicate key. I'm not sure how that's happening either. I have 3 Kamailio boxes in a cluster replicating registrations to each other and using a MySQL Galara cluster for the usrloc DB. This is Kamailio 4.0.6. Thanks, Marc

4 4

loose_route to find transport=tcp
by Ryan Kendrick 16 Jun '15

16 Jun '15

We are using Kamailio 4.2.5 as a registrar and proxy between many dispersed end-users of a soft phone app and our calling platform / switch. Until now we have used udp exclusively but are trying to introduce tcp between end-users and Kamailio, leaving udp between Kam and our switch...while maintaining the ability for some end-users to use udp to Kam. With some simple address checks I am able to always send to our switch over udp. If all end-users used tcp I could send everything else tcp, but I need to maintain udp support. The specific problem I am having is on a reINVITE such as this one from our platform to the a-leg: INVITE sip:xxxxxx@xxxxxxxxxxxxx:42679;user=phone SIP/2.0 Via: SIP/2.0/UDP xxxxxxxxxxxxx:5060;branch=z9hG4bK218cc8e12ll5035f67INV6a67885312aad Max-Forwards: 35 Route: <sip:xxxxxxxxxxx;lr;r2=on;ftag=daba971c;did=b57.4872;nat=yes> Route: <sip:xxxxxxxxxxx:5070;transport=tcp;lr;r2=on;ftag=daba971c;did=b57.4872;nat=yes> Contact: <sip:xxxxxxxxxx@xxxxxxxxxxxxx:5060> To: "xxxxxx"<sip:xxxxxx@xxxxxxxxxxxxxxxxxxxxxxxxxxx:5070>;tag=daba971c From: <sip:xxxxxxxxxx@xxxxxxxxxxxxxxxxxxxxxxxxxxx :5070>;tag=6a678853-co76461-INS002 Call-ID: MDI4ZmFjNmZhN2Y1NWE2ZTViNTkyZGUwNWE2YzUzYmU CSeq: 7646101 INVITE Allow: INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,INFO,UPDATE Content-Type: application/sdp Date: Mon, 15 Jun 2015 20:10:18 GMT User-Agent: xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx Content-Length: 262 As you might notice, we have rr:enable_double_rr set: *There are some situations when the server needs to insert two Record-Route header fields instead of one. For example when using two disconnected networks or doing cross-protocol forwarding from UDP->TCP. This parameter enables inserting of 2 Record-Routes. The server will later remove both of them. * and I believe it is necessary to keep this way. Without it Kamailio doesn't even see the reINVITE...the switch probably tries tcp and that's not setup between the two. The invite above is sent to the a-leg over udp but I would expect and need it to be tcp in this case. The reINVITE is part of an existing dialog. We call loose_route() followed by some simple bflag setting and flag checking, t_on_reply(), ... then t_relay(). I do have a functional workaround but would prefer to avoid such manual handling by utilizing built-in functionality properly. # # relay the message # if(route(TEST_TOGW)) { if (!t_relay_to_udp()) { sl_reply_error(); } } else { if ($(hdr(Route)[-1]) =~ "tcp") { if(!t_relay_to_tcp()) { sl_reply_error(); } } else if (!t_relay()) { sl_reply_error(); } } I'm not 100% sure how reliable or fast this will be, but it does work so far in my simple tests. Is loose_route supposed to see and use the transport=tcp but isn't for some reason? It seems like the right thing to do to me. If not, is there anything else I can/should be doing in the tm and/or rr modules to make Kamailio realize it needs to send this message over TCP? If not in those two modules is there some recommended way perhaps via registrar or usrloc etc. to make Kamailio remember/store when a user is connected via TCP and be able to do a quick lookup before sending to them? Anything else I'm missing or not thinking of? Please let me know if I can further explain and rest assured any assistance will be much appreciated!!!

1 0

Keep-Alive in dialog "freeing a free fragment"
by Dirk Teurlings - SIGNET B.V. 16 Jun '15

16 Jun '15

Hi, (Kamailio 4.2.3) Working to get dialogs closed and cleaned up nicely I'm looking into the ka_interval and ka_timer parameter of the dialog module. The following is now happening. Scenario: 1. Call gets setup 2. Callee's internet connection drops 3. Call gets terminated (nicely) due to ka_timer/ka_interval 4. Log fills with "freeing a free fragment" after about 10 seconds Digging into this I saw that the dialog is being closed and cleaned properly in step 3. But it fails to remove/clean the timers used for the keep-alive. That's what triggers step 4, which in turn tries to clean the (or some of the) dialog space again. Is this expected behaviour when using the keep-alive options, or am I running into somekind of (known) bug? Thanks in advance for any help. -- Cheers, Dirk Teurlings

2 12

CANCEL request and subsequent 487 response to INVTE configuration problem
by Joao Alves 16 Jun '15

16 Jun '15

Hi, I'm creating a configuration is for a SIP <> HTTP gateway (performing protocol conversion) and thus all SIP messages and responses needs to be generated from the kamailio config file. I'm failing to create the case where the calling user abandons the session, and thus, following the reception of a CANCEL a 200 "OK" needs to be sent to this transaction and later the 487 "Request Terminated" to the SIP INVITE original session. On the traces I see: SIP INVITE (Cseq 1 INVITE) --> 100 Trying (Cseq 1 INVITE) <-- 180 Ringing (Cseq 1 INVITE) <-- SIP CANCEL (Cseq 1 CANCEL) --> 200 Ok (CSeq 1 CANCEL) <-- 487 Request Terminated (CSeq 1 CANCEL) Have then tried several options, trying to force the CSeq 1 INVITE but without success. t_reply_callid("$ci", "$cs", "487", " Request Terminated "); t_reply_callid("$ci", "$rm", "487", "Request Terminated"); t_reply_callid("$ci", "1", "487", "Request Terminated"); For instance, for the last I get the following log. 6(60226) exec: *** cfgtrace:request_route=[REQINIT] c=[/etc/kamailio/kamailio.cfg] l=705 a=28 n=t_reply_callid 6(60226) DEBUG: tm [t_lookup.c:1715]: t_lookup_callid(): created comparable call_id header field: >Call-ID: 76589ZTJmNTAwZDhlOTZiM2I3MjhhYzllNjgyOWVjZGZmMzk < 6(60226) DEBUG: tm [t_lookup.c:1719]: t_lookup_callid(): created comparable cseq header field: >CSeq: 1 INVITE< 6(60226) DEBUG: tm [t_lookup.c:1722]: t_lookup_callid(): just locked hash index 18668, looking for transactions there: 6(60226) DEBUG: tm [t_lookup.c:1749]: t_lookup_callid(): DEBUG: t_lookup_callid: transaction not found. 6(60226) DEBUG: tmx [tmx_mod.c:500]: t_reply_callid(): Lookup failed - no transaction Could you assist here? Thanks, Joao Joao Alves Solution Architect, Unified Communications +351 214094660 (desk) +351 912783702 (mobile) AMDOCS | EMBRACE CHALLENGE EXPERIENCE SUCCESS This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp

2 4

Kamailio license usage
by Alexandru Covalschi 15 Jun '15

15 Jun '15

Maybe it may be an offtopic, but I'm not really into legal issues - so I'm sorry if this message is not fully related to this mailing list. Can I use Kamailio to provide VoIP backend for kind of CRM system in case of SaaS? --- Alexandru Covalschi ABRISS-Solutions VoIP engineer and system administrator phone: +37367398493 web: http://abs-telecom.com/

3 3

Re: [SR-Users] Kamailio number of LCR rules
by Juha Heinanen 15 Jun '15

15 Jun '15

Henry Fernandes writes: > But we didn¹t want to add hundreds of thousands or millions of extra rules > just to deal with this problem. We didn¹t consider that a good > workaround. OK, I got it. Now priorities choose from equally long prefixes instead of all matching prefixes. I'll take a look at the code to check if the latter could be easily added as an option. -- Juha

1 1

Re: [SR-Users] Kamailio number of LCR rules
by Juha Heinanen 15 Jun '15

15 Jun '15

Henry Fernandes writes: > Let’s say you have the following info in your LCR. > > Prefix, Carrier/Route, Rate > 1306, A, 1.5 > 1306343, A, 1.7 > 1306, B, 1.4 > > If you have phone number 13063431111, the LCR module matches it to the > longest prefix (1306343) and routes the call to Carrier A. However, that > is incorrect. It should match it to Carrier B which is the cheapest. > Carrier B has provided a rate of 1.4 for all calls with prefix 1306. why can't you add rule 1306343, B, 1.4 to solve the problem? -- juha

2 1

Kamailio number of LCR rules
by Ali Taher 15 Jun '15

15 Jun '15

Hello, I'm testing LCR module in Kamailio and everything works as expected so far. Before going further , I just want to ask about how much rule in LCR_rule table and target in LCR_rule_target can Kamailio support . We are dealing with hundreds of millions of rules and targets. Kindly advise if Kamailio can support this number of rules. Thanks Ali

3 4

Removing parameters from user part of URI
by Alex Balashov 15 Jun '15

15 Jun '15

Hi, If I have an RURI like this: INVITE sip:4045551212;param=xyz@domain:port SIP/2.0 which transformations can I best string together in order to remove the ';param=xyz' bit and extract just the core user part value (4045551212)? {uri.user} returns '4045551212;param=xyz'. There are numerous transformations to help one access the parameters. What if I want to exclude the parameters? I've already scoured the available transformations, just hoping a second pair of eyes can tell me I've missed something useful. Cheers, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 303 Perimeter Center North, Suite 300 Atlanta, GA 30346 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

5 19

SIP-over-Websocket Load Balancing
by Murugan Pandian 15 Jun '15

15 Jun '15

HI, how to handle sip-over-websocket load balancing (WebRTC)

4 15

Websocket Load Balancing
by W5RTC 14 Jun '15

14 Jun '15

HI, how to handle SIP-OVER-Websocket loab balancing (WebRTC)

3 2

ASYNC Module
by Aaron Hamstra 13 Jun '15

13 Jun '15

We just updated our development environment from 4.2.2 to 4.2.5 and started noticing the server sending a 500 I'm terribly sorry, server error occurred (1/TM) error back as soon as the async_route block gets called. I have removed everything from our async route other than an xlog statement trying to determine if something in that route block was causing the 500 to occur. It still happens even with only the xlog. If the async_route is not in the config, then the server does not send the 500 response. We rolled back to 4.2.2 and the problem no longer occurs. I have attached a sip trace of the messages. I believe the relevant parts of our config are below, please let me know if you need more from the config however. Any ideas? async_route("PUSHJOIN", "2"); -------------------------- route[PUSHJOIN] { xlog("L_INFO", "Does it get here?\n"); exit; } Thanks in Advance, Aaron

2 11

lookup("location") question
by Al S 12 Jun '15

12 Jun '15

Hi, lookup("location") functions extracts username from Request-URI and relay INVITE without any username, causing UAC sends a CANCEL message. Is the function suppose to replace uri with "contact" field in the found entry in "location" table? INVITE uri arrives at Kamailio:Request-Line: INVITE sip:8181111111@example.net:5060 SIP/2.0 INVITE uri leaves Kamilio:Request-Line: INVITE sip:10.10.10.1:5060 SIP/2.0 lookup("location") function finding match and returns 1: $rc=1 entry in "location" table for this call:*************************** 1. row *************************** id: 104 ruid: uloc-557ad93a-2d6-1 username: 8181111111 domain: NULL contact: sip:8181111111@10.10.10.1:5060 received: NULL path: NULL Any help in this regard would be greatly appreciated Al,Thanks

2 2

How to prevent multiple registration entries in module ims_registrar_scscf
by jyaim 12 Jun '15

12 Jun '15

Hi all, I have an issue with mobile devices which get a different IP address each time they lose network. The scscf keeps the registration datas from the previous registrations, and forks requests toward all known identities. Is it possible to override the contact address at each register and keep only the last registration address per impi ? Thanks Regards JM

2 1

Problems with http_query(POST)
by Joao Alves 12 Jun '15

12 Jun '15

Hi, I'm finding a problem submitting a HTTP POST using the http_query(). Currently running on kamailio version: 4.1.8 on Debian wheezy. Normally this command has been worked fine, but since the body of the message increased (the POST data size is around 2000 bytes ) I start to get the following problem: 8(49381) ERROR: <core> [pvapi.c:1338]: pv_printf(): no more space for spec value 8(49381) ERROR: <core> [pvapi.c:1347]: pv_printf(): buffer overflow -- increase the buffer size... 8(49381) ERROR: <core> [sr_module.c:1696]: get_str_fparam(): Could not convert the PV-formated string to str 8(49381) ERROR: utils [functions.c:112]: http_query(): cannot get post value I've searched on how to increase buffer and found some global settings that could be directly, however the problem persist despite having set very high values for those: pv_buffer_size=32768 tcp_rd_buf_size=32768 maxbuffer=262144 Can someone assist and suggest a solution. Thanks, Joao Alves This message and the information contained herein is proprietary and confidential and subject to the Amdocs policy statement, you may review at http://www.amdocs.com/email_disclaimer.asp

3 7

Mark CANCEL of one fork when another answers the call - old Kamailio
by Sebastian Damm 11 Jun '15

11 Jun '15

Hi, I'm looking for an option to mark CANCELs which are generated locally by Kamailio after receiving a 200 OK for another fork of the same call. In the tm module documentation I found the option local_cancel_reason which enables this. This option is available (and turned on by default) in newer Kamailios. But we have a legacy system still running 1.5 (and we don't want to upgrade it anymore, it will be obsolete pretty soon), which doesn't have this option yet. Is there a possibility to add a header to this locally generated CANCEL via some function in the config file, too? Thanks in advance. Best Regards, Sebastian

2 1

Kamailio LCR module memory concern
by Ali Taher 11 Jun '15

11 Jun '15

Hello, I'm using LCR module where I have 44 million rules and 13 million rule targets (not all rules have destinations). Noting that I'm running Ubuntu on virtual machine with 8GB allocated memory. When I'm setting lcr_rule_hash_size parameter to 268435456 (2 power 28) I got the following error : "no memory for rules hash table" in the log , noting that when I run ipcs -lm command I got max total shared memory shared memory (kbytes) : 8388608. When I'm setting lcr_rule_hash_size parameter to 1024,meaning 1024 rules will be fetched from the database (please correct if I'm wrong) Kamailio takes around 2 minutes and 30 seconds to start , which is worrying for me :( Please advise how many rules Kamailio can support (if 8 GB RAM are allocated to the server) , and what size every rule takes in the hash table built in the shared memory. Thanks, Ali

5 24

Kamailio v4.3.0 Released - new major version is out
by Daniel-Constantin Mierla 10 Jun '15

10 Jun '15

Kamailio v4.3.0 is out – a new major release, bringing more features, flexibility and improvements. You can read detailed release notes at: * http://www.kamailio.org/w/kamailio-v4-3-0-release-notes/ Many thanks to all developers and community members that made possible this release. Enjoy Kamailio v4.2.0! Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 0

Releasing Kamailio v4.3.0
by Daniel-Constantin Mierla 10 Jun '15

10 Jun '15

Hello, I am considering to release next major version of Kamailio (v4.3.0) next week, on Wednesday (June 10) or Thursday (June 11). The pool of things to fix is minor, Victor already updated deb packaging for 4.3, left to do: add docs for 4.3 series in wiki, prepare RPM packaging and build proper 'what is new' list of features. Testing didn't reveal new issues for me in the recent past as I run in two servers with live traffic. If you have issues with 4.3 branch, be sure you report them on Github tracker: - https://github.com/kamailio/kamailio/issues Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

2 4

several uac_replace_from in one dialog
by Никитенко Виталий 10 Jun '15

10 Jun '15

Hi! Making calls via different trunks through kamailio for simply so uac_replace_from ("sip:regname@ipaddress"); if (! t_relay ()) { uac_replace_from ("sip:regname2@ipaddress"); t_relay () } exit and run tcpdump i see in fisrt INVITE From: "testname" <sip: regname @ ipaddress>; tag = as21aea6e0 in second INVITE From: "testname" <sip: regname @ ipaddress sipregname2 @ ipaddress>; tag = as21aea6e0 and in the log file EROOR: can not parse From URI, and then can not find a BYE to terminate the call I try do remove_hf insert_hf but then i do second remove_hf it realy not delete and i have two field From. Who can tell me how few times to change the From field in a one dialog?

2 1

kamcmd permissions problem
by Uri Shacked 10 Jun '15

10 Jun '15

Hi, I am using exec_avp() command from kamailio. The command execute a shell script that runs kamctl and kamcmd commands. kamailio runs with user 'kamailio' and group 'kamailio' (defaults). There is not problem with the kamctl commands. But, the kamcmd commands fails due to permissions issues. Any ideas? Cheers, Uri

2 1

403 not relaying
by jaflong jaflong 09 Jun '15

09 Jun '15

Hi, I am getting a 403 not relaying response. This is the part of the config How would I print the value of var myself, from_uri and uri # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; }

2 1

IP fragment
by Tuan Viet Nguyen 09 Jun '15

09 Jun '15

Hello, I would like to know if Kamailio handles the case where we receive a big SIP message fragmented in IP layer? Thanks,

2 2

Testing RPMs for v4.3.0
by Daniel-Constantin Mierla 09 Jun '15

09 Jun '15

Hello, in preparation for releasing v4.3.0 next week, I updated the specs for building RPMs from branch 4.3. However, I am not able to test the results, just made sure the build is ok. Packages are available at: - https://build.opensuse.org/package/show/home:kamailio:v4.3.x-rpms/kamailio43 I hope people here using RPMs will do some tests and provide feedback, interested mainly in: - it installs ok - after install, it starts ok with default configuration file - fifo and unix sockets are created now in /var/run/kamailio, so it has to be checked they are there - kamctl works ok for commands: ps, uptime and 'mi version' Some modules are not packaged in various versions, as I could not spot what are the packages needed for building them and if they are available with different names. For example, apparently jansson lib is on redhat, but not on centos, libev doesn't seem to be there. Also, I haven't had time to look at packaging erlang module. Help with the above is appreciated as well as helping building RPMs genrally. Maybe the init.d script needs to be reviewed and updated as well. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Book: SIP Routing With Kamailio - http://www.asipto.com

2 4

Need information regarding kamilio setup to support IPSec
by Priyaranjan Nayak 09 Jun '15

09 Jun '15

Hi All, I have a ims client endpoint supporting rfc 3329(IPSec).Now I am sending register message to kamailio server with Security-Client header and not getting any Security-Server header in response. Could you please tell me below queries - Whether kamailio server supporting rfc 3329 (IPSec) or not ? - If yes,how can we setup kamailio server to support Security-Client,Security-Server and Security-varify(as per rfc 3329) header ? Thanks Priyaranjan

3 3

Private IP in SDP packet headers
by Alex 08 Jun '15

08 Jun '15

Hello! Please help to fix problem with sdp headers UAC Inet -> (X.X.X.X) Kamailio (192.168.30.250) -> Asterisk (192.168.30.2) When i call from UAC to 9002 i received INVITE/SDP from kamailio SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.52:27080 ;received=10.10.101.50;branch=z9hG4bK-d8754z-027c786dac17bf68-1---d8754z-;rport=27080 Record-Route: <sip :192.168.30.2;line=sr-mYtaP6eErk-dx6VfrLzfr6BaPGj0OHFfPYd0OHFfPYIQpHmFr9mQPKDEx9VlvZ8QO4ttma**> Record-Route: <sip:X.X.X.X;r2=on;lr=on;ftag=0748d948;nat=yes> From: <sip:user4@X.X.X.X>;tag=0748d948 To: <sip:9002@X.X.X.X>;tag=as3914e1d1 Call-ID: ZWU5YmFiNTNhNmNmYWQzYzhkZWUzZDNjOTU3MDFiNGU. CSeq: 2 INVITE Server: Virtel.net Node2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:192.168.30.2;line=sr-mYtaP62ar9nzrg20y6eYPA-LrA-0P6Bax6z*> Content-Type: application/sdp Content-Length: 278 v=0 o=root 732368067 732368067 IN IP4 X.X.X.X s=Asterisk PBX 11.17.1 c=IN IP4 X.X.X.X t=0 0 m=audio 15768 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 a=nortpproxy:yes Why Record-Route and Contact fields contain private IP of asterisk ?

2 1

Need to add extra header in 200 OK message by using kamailio.cfg file
by Priyaranjan Nayak 08 Jun '15

08 Jun '15

Hi All, I wanted to send a extra header(i.e. Security-Server ) in the 200 OK from kamailio server. Could you please suggest me how can I add extra header in 200 OK message by using kamailio.cfg file ? Thanks Priyaranjan

3 4

msilo storage of message id
by Shouvanik Chakrabarti 07 Jun '15

07 Jun '15

Hello. We have been using the msilo module to handle offline SIP messaging. I have noticed that the m_store function ends up storing all the records with the default message id which is zero. However m_dump uses this id to figure which messages have already been sent. Thus on calling m_dump on registration, no messages are dumped. We simply get a 'message already sent' debug log. Is there a way around this? Any help would be deeply appreciated. Thanks, Shouvanik.

2 2

Kamailio second contact header
by Ali Taher 05 Jun '15

05 Jun '15

Hello, I'm trying to build new contact header that will be sent from Kamailio as below : load_gws(1, $rU, $fu); append_to_reply("Contact:"); while (next_gw()) { append_to_reply("$ru"); } append_to_reply("\r\n"); sl_send_reply("300","Multiple Choices"); exit; yet I'm getting two Contact headers as shown in below SIP message. How can I remove second contact header (highlighted below in gray) Via: SIP/2.0/UDP 192.168.26.3:42528;branch=z9hG4bK-d8754z-4c2cc56c3145b446-1---d8754z-;rport= 42528;received=192.168.26.3 To: "9612"<sip:9612@192.168.110.181>;tag=b27e1a1d33761e85846fc98f5f3a7e58.1a8d From: "test"<sip:test@192.168.110.181>;tag=8571af74 Call-ID: Y2MzOTNiOGRhMTc2NDkwZjIxNzZhYTAzZmM0OGE3NDY CSeq: 2 INVITE Contact: sip:9612@C001,sip:9612@192.168.26.1,sip:9612@192.168.26.6,sip:9612@192.168.2 6.8 Contact: sip:9612@192.168.26.8 Server: kamailio (4.1.8 (x86_64/linux)) Content-Length: 0 Thanks, Ali

2 12

Misc Radius can't parse sip-avps?
by Yasin CANER 05 Jun '15

05 Jun '15

Hello all; i am working on misc_radius modulü. i configured system like below. and when i try to write debug , it gives <null> Could you take a look what i am doing wrong Thanks -----FREERADIUS----- 101(a)192.168.0.188 Auth-Type := Accept, Service-Type == "SIP-Caller-AVPs" Sip-Avp = "#3#1" --------------------------- -----------Kamailio.cfg modparam("misc_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("misc_radius", "group_service_type", 12) modparam("misc_radius", "uri_service_type", 10) modparam("misc_radius", "caller_service_type", 30) modparam("misc_radius", "callee_service_type", 31) modparam("misc_radius", "caller_extra", "Calling-Station-Id=$fu") modparam("misc_radius", "callee_extra", "SIP-URI-User=$rU;SIP-URI-Host=$rd") #modparam("misc_radius", "group_extra", "My-Special=$avp(i:100)") modparam("misc_radius", "uri_extra", "Called-Station-Id=$tu") modparam("misc_radius", "common_response", 1) #"dont forget to change caller_service_type to 30 and callee_service_type to 31 as written in dictionary.kamailio" if(is_method("INVITE")){ if(!radius_load_caller_avps("$fU(a)192.168.0.188")){ xlog("L_ALERT","Bulamadim Bulamadim"); sl_send_reply("403", "Forbidden - Bulamadim"); exit; } xlog("L_ALERT"," i3 $avp(i:3) $avp(s:3) s6 $avp(6) $avp(session-protocol) $avp(Sip-Avp)"); } -------------------------- ---------RESULT-------- 1(21867) DEBUG: misc_radius [functions.c:453]: radius_is_user_in(): success 1(21867) DEBUG: misc_radius [functions.c:268]: radius_load_caller_avps(): success 1(21867) DEBUG: misc_radius [functions.c:137]: generate_avps_rad(): AVP 'SIP-AVP'/0='#3#1'/0 has been added May 25 15:05:28 group1-imslab11 /usr/local/sbin/kamailio[21587]: ALERT: <script>: i3 <null> <null> s6 <null> <null> 3#1 --------------------------

3 6

Kamailio fails to start if cannot connect to MySQL
by Al S 05 Jun '15

05 Jun '15

Hi, We have verified if MySQL server is offline when Kamailio is starting, Kamailio fails to start correctly.Is this a bug? Jun 4 10:53:22 ape /usr/sbin/kamailio[10630]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2)Jun 4 10:53:22 ape /usr/sbin/kamailio[10630]: ERROR: <core> [db.c:320]: could not add connection to the poolJun 4 10:53:22 ape /usr/sbin/kamailio[10630]: ERROR: usrloc [dlist.c:518]: failed to open database connectionJun 4 10:53:22 ape /usr/sbin/kamailio[10630]: ERROR: registrar [reg_mod.c:472]: failed to register domainJun 4 10:53:22 ape /usr/sbin/kamailio[10630]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:858Jun 4 10:53:22 ape /usr/sbin/kamailio[10630]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:861Jun 4 10:53:22 ape /usr/sbin/kamailio[10630]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:863 We are running kamailio 3.3.7. Thanks,Al

3 2

Private IP in SDP packet headers
by Alex 05 Jun '15

05 Jun '15

Hello! Please help to fix problem with sdp headers UAC Inet -> (X.X.X.X) Kamailio (192.168.30.250) -> Asterisk (192.168.30.2) When i call from UAC to 9002 i received INVITE/SDP from kamailio SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.52:27080 ;received=10.10.101.50;branch=z9hG4bK-d8754z-027c786dac17bf68-1---d8754z-;rport=27080 Record-Route: <sip:192.168.30.2;line=sr-mYtaP6eErk-dx6VfrLzfr6BaPGj0OHFfPYd0OHFfPYIQpHmFr9mQPKDEx9VlvZ8QO4ttma**> Record-Route: <sip:X.X.X.X;r2=on;lr=on;ftag=0748d948;nat=yes> From: <sip:user4@X.X.X.X>;tag=0748d948 To: <sip:9002@X.X.X.X>;tag=as3914e1d1 Call-ID: ZWU5YmFiNTNhNmNmYWQzYzhkZWUzZDNjOTU3MDFiNGU. CSeq: 2 INVITE Server: Virtel.net Node2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:192.168.30.2;line=sr-mYtaP62ar9nzrg20y6eYPA-LrA-0P6Bax6z*> Content-Type: application/sdp Content-Length: 278 v=0 o=root 732368067 732368067 IN IP4 X.X.X.X s=Asterisk PBX 11.17.1 c=IN IP4 X.X.X.X t=0 0 m=audio 15768 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 a=nortpproxy:yes Why Record-Route and Contact fields contain private IP of asterisk ?

2 1

How should a PBX behave when a user dials a DID configured for the same or another tenant on that PBX?
by Antonio Gómez Soto 04 Jun '15

04 Jun '15

Hello, This is question on PBX behavior, what is the right thing to do, and how do PBX's generally behave. If a user on a phone, dials a number, which happens to be configured on the same phone system (for example another tenant), there are two options: 1. The PBX notices this, and directly connects the phone to the DID on that system 2. The PBX sends the call out on the SIP trunk, and the provider-routing sends the call back as an incoming call. What are the pros and cons of each option? How do PBX's generally behave? Thanks, Antonio

2 1

Getting 404 From Kamailio for PRACK
by Austin Einter 04 Jun '15

04 Jun '15

Hi All I am trying ti implement a IMS client. For testing I am using Kamailio proxy. During call setup, I get 404 for Prack message. Whats going wrong not able to figure out. Here is the Kamailio log below, can somebody help me too find whats wrong her. *11(25376) DEBUG: sl [sl.c:296]: send_reply(): reply in stateless mode (sl)11(25376) DEBUG: <core> [msg_translator.c:206]: check_via_address(): check_via_address(192.168.1.195, 192.168.1.195, 0)11(25376) DEBUG: <core> [tcp_main.c:2281]: tcpconn_send_put(): tcp_send: send from reader (25376 (11)), reusing fd11(25376) DEBUG: <core> [tcp_main.c:2516]: tcpconn_do_send(): tcp_send: sending...11(25376) DEBUG: <core> [tcp_main.c:2550]: tcpconn_do_send(): tcp_send: after real write: c= 0x7fe2ebd05390 n=386 fd=511(25376) DEBUG: <core> [tcp_main.c:2551]: tcpconn_do_send(): tcp_send: buf=SIP/2.0 200 OKVia: SIP/2.0/TCP 192.168.1.195:40611;branch=z9hG4bK1186661565From: <sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>>;tag=1307796831To: <sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>>;tag=25fbdd2b0c9b93b0791296269f5ae411.38a0Call-ID: 835380265(a)192.168.1.195 <835380265(a)192.168.1.195>CSeq: 2 REGISTERContact: <sip:linux_ua_1@192.168.1.195:40611;transport=tcp>;expires=3600Server: kamailio (4.2.3 (x86_64/linux))Content-Length: 011(25376) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)11(25376) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)11(25376) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)11(25376) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)11(25376) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)11(25376) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)11(25376) DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil)11(25376) DEBUG: <core> [receive.c:298]: receive_msg(): receive_msg: cleaning up11(25376) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0xa33180, 5, 2, 0x7fe2ebd05390), fd_no=111(25376) DEBUG: <core> [tcp_read.c:1362]: tcp_read_req(): tcp_read_req: content-length= 011(25376) DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request:11(25376) DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <SUBSCRIBE>11(25376) DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: <sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>>11(25376) DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0>11(25376) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK1577674881>; state=1611(25376) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=511(25376) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=211(25376) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via11(25376) DEBUG: <core> [receive.c:154]: receive_msg(): After parse_msg...11(25376) DEBUG: <core> [receive.c:197]: receive_msg(): preparing to run routing scripts...11(25376) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=1011(25376) DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [28]; uri=[sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>] 11(25376) DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>>]11(25376) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <2> <SUBSCRIBE>11(25376) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 11(25376) DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=011(25376) DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header11(25376) DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=133567903511(25376) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=2911(25376) DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks result: 111(25376) DEBUG: siputils [checks.c:103]: has_totag(): no totag11(25376) DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=1 T start=(nil)11(25376) DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start searching: hash=34123, isACK=011(25376) DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction matching failed11(25376) DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found11(25376) DEBUG: tm [t_lookup.c:1141]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=2 T end=(nil)11(25376) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 13==13 && [192.168.1.195] == [192.168.1.195]11(25376) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 4060 (advertise 0) matches port 506011(25376) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 13==13 && [192.168.1.195] == [192.168.1.195]11(25376) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 4060 (advertise 0) matches port 506011(25376) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 13==13 && [192.168.1.195] == [192.168.1.195]11(25376) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 4090 (advertise 0) matches port 506011(25376) DEBUG: registrar [lookup.c:203]: lookup_helper(): contact for [linux_ua_1] cannot handle the SIP method11(25376) DEBUG: registrar [lookup.c:211]: lookup_helper(): 'linux_ua_1' has no valid contact in usrloc11(25376) DEBUG: tm [t_lookup.c:1373]: t_newtran(): DEBUG: t_newtran: msg id=2 , global msg id=2 , T on entrance=(nil)11(25376) DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start searching: hash=34123, isACK=011(25376) DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction matching failed11(25376) DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found11(25376) DEBUG: tm [t_hooks.c:380]: run_reqin_callbacks_internal(): DBG: trans=0x7fe2ebd10e68, callback type 1, id 0 entered11(25376) DEBUG: <core> [md5utils.c:67]: MD5StringArray(): DEBUG: MD5 calculated: c17c6f2bd5d60787072aef8ffcf5bfe611(25376) DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=2 T start=0x7fe2ebd10e6811(25376) DEBUG: tm [t_lookup.c:1144]: t_check_msg(): DEBUG: t_check_msg: T already found!11(25376) DEBUG: <core> [msg_translator.c:206]: check_via_address(): check_via_address(192.168.1.195, 192.168.1.195, 0)11(25376) DEBUG: <core> [mem/shm_mem.c:111]: _shm_resize(): WARNING:vqm_resize: resize(0) called11(25376) DEBUG: tm [t_reply.c:1653]: cleanup_uac_timers(): DEBUG: cleanup_uac_timers: RETR/FR timers reset11(25376) DEBUG: <core> [tcp_main.c:2281]: tcpconn_send_put(): tcp_send: send from reader (25376 (11)), reusing fd11(25376) DEBUG: <core> [tcp_main.c:2516]: tcpconn_do_send(): tcp_send: sending...11(25376) DEBUG: <core> [tcp_main.c:2550]: tcpconn_do_send(): tcp_send: after real write: c= 0x7fe2ebd05390 n=334 fd=511(25376) DEBUG: <core> [tcp_main.c:2551]: tcpconn_do_send(): tcp_send: buf=SIP/2.0 405 Method Not AllowedVia: SIP/2.0/TCP 192.168.1.195:40611;branch=z9hG4bK1577674881From: <sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>>;tag=1335679035To: <sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>>;tag=ff272440259f71af2c767e7942717037-95d5Call-ID: 1397060040(a)192.168.1.195 <1397060040(a)192.168.1.195>CSeq: 2 SUBSCRIBEServer: kamailio (4.2.3 (x86_64/linux))Content-Length: 011(25376) DEBUG: tm [t_reply.c:728]: _reply_light(): DEBUG: reply sent out. buf=0x7fe30fea65c0: SIP/2.0 405 Method N..., shmem=0x7fe2ebd135a8: SIP/2.0 405 Method N11(25376) DEBUG: tm [t_reply.c:738]: _reply_light(): DEBUG: _reply_light: finished11(25376) DEBUG: sl [sl.c:288]: send_reply(): reply in stateful mode (tm)11(25376) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)11(25376) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)11(25376) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)11(25376) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)11(25376) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)11(25376) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)11(25376) DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil)11(25376) DEBUG: <core> [receive.c:298]: receive_msg(): receive_msg: cleaning up 9(25374) DEBUG: websocket [ws_conn.c:459]: wsconn_get_list(): wsconn_get_list 9(25374) DEBUG: websocket [ws_conn.c:503]: wsconn_get_list(): wsconn_get_list returns list [(nil)] with [0] members10(25375) DEBUG: <core> [tcp_read.c:1362]: tcp_read_req(): tcp_read_req: content-length= 35910(25375) DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request:10(25375) DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <INVITE>10(25375) DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: <sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>>10(25375) DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0>10(25375) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK1970975121>; state=1610(25375) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=510(25375) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=210(25375) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via10(25375) DEBUG: <core> [receive.c:154]: receive_msg(): After parse_msg...10(25375) DEBUG: <core> [receive.c:197]: receive_msg(): preparing to run routing scripts...10(25375) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=1010(25375) DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [28]; uri=[sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>] 10(25375) DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>>]10(25375) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <2> <INVITE>10(25375) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 10(25375) DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=35910(25375) DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header10(25375) DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=60426366410(25375) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=2910(25375) DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks result: 110(25375) DEBUG: siputils [checks.c:103]: has_totag(): no totag10(25375) DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg: msg id=3 global id=2 T start=0xffffffffffffffff10(25375) DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start searching: hash=12870, isACK=010(25375) DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction matching failed10(25375) DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found10(25375) DEBUG: tm [t_lookup.c:1141]: t_check_msg(): DEBUG: t_check_msg: msg id=3 global id=3 T end=(nil)10(25375) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 13==13 && [192.168.1.195] == [192.168.1.195]10(25375) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 4060 (advertise 0) matches port 506010(25375) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 13==13 && [192.168.1.195] == [192.168.1.195]10(25375) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 4060 (advertise 0) matches port 506010(25375) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 13==13 && [192.168.1.195] == [192.168.1.195]10(25375) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 4090 (advertise 0) matches port 506010(25375) DEBUG: registrar [lookup.c:198]: lookup_helper(): contact for [linux_ua_1] found by address10(25375) DEBUG: tm [t_lookup.c:1373]: t_newtran(): DEBUG: t_newtran: msg id=3 , global msg id=3 , T on entrance=(nil)10(25375) DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start searching: hash=12870, isACK=010(25375) DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction matching failed10(25375) DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found10(25375) DEBUG: tm [t_hooks.c:380]: run_reqin_callbacks_internal(): DBG: trans=0x7fe2ebd13848, callback type 1, id 0 entered10(25375) DEBUG: <core> [md5utils.c:67]: MD5StringArray(): DEBUG: MD5 calculated: c4c3d26baa62dea619271f186d6344fc10(25375) DEBUG: tm [t_funcs.c:347]: t_relay_to(): SER: new INVITE10(25375) DEBUG: <core> [msg_translator.c:206]: check_via_address(): check_via_address(192.168.1.195, 192.168.1.195, 0)10(25375) DEBUG: <core> [mem/shm_mem.c:111]: _shm_resize(): WARNING:vqm_resize: resize(0) called10(25375) DEBUG: <core> [tcp_main.c:2281]: tcpconn_send_put(): tcp_send: send from reader (25375 (10)), reusing fd10(25375) DEBUG: <core> [tcp_main.c:2516]: tcpconn_do_send(): tcp_send: sending...10(25375) DEBUG: <core> [tcp_main.c:2550]: tcpconn_do_send(): tcp_send: after real write: c= 0x7fe2ebd096b0 n=307 fd=510(25375) DEBUG: <core> [tcp_main.c:2551]: tcpconn_do_send(): tcp_send: buf=SIP/2.0 100 trying -- your call is important to usVia: SIP/2.0/TCP 192.168.1.195:19009;branch=z9hG4bK1970975121From: <sip:linux_ua_0@192.168.1.195 <sip%3Alinux_ua_0(a)192.168.1.195>>;tag=604263664To: <sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>>Call-ID: 734015646(a)192.168.1.195 <734015646(a)192.168.1.195>CSeq: 2 INVITEServer: kamailio (4.2.3 (x86_64/linux))Content-Length: 010(25375) DEBUG: tm [t_reply.c:728]: _reply_light(): DEBUG: reply sent out. buf=0x7fe30fea6968: SIP/2.0 100 trying -..., shmem=0x7fe2ebd163f0: SIP/2.0 100 trying -10(25375) DEBUG: tm [t_reply.c:738]: _reply_light(): DEBUG: _reply_light: finished10(25375) DEBUG: <script>: new branch [0] to sip:linux_ua_1@192.168.1.195:40611;transport=tcp10(25375) DEBUG: <core> [msg_translator.c:2893]: create_via_hf(): create_via_hf: id added: <;i=2>, rcv proto=210(25375) DEBUG: <core> [msg_translator.c:206]: check_via_address(): check_via_address(192.168.1.195, 192.168.1.195, 0)10(25375) DEBUG: <core> [tcp_main.c:1818]: tcp_send(): tcp_send: no open tcp connection found, opening new one10(25375) DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 192.168.1.19510(25375) DEBUG: <core> [tcp_main.c:1073]: tcpconn_new(): tcpconn_new: on port 40611, type 210(25375) DEBUG: <core> [tcp_main.c:1382]: tcpconn_add(): tcpconn_add: hashes: 3453:3998:0, 310(25375) INFO: <core> [tcp_main.c:2753]: tcpconn_1st_send(): quick connect for 0x7fe2ebd1703010(25375) DEBUG: tm [t_funcs.c:394]: t_relay_to(): SER: new transaction fwd'ed10(25375) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)10(25375) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)14(25379) DEBUG: <core> [tcp_main.c:3565]: handle_ser_child(): handle_ser_child: read response= 7fe2ebd17030, 6, fd 28 from 10 (25375)10(25375) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)10(25375) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)10(25375) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)10(25375) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)10(25375) DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil)14(25379) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0x9eeee0, 28, 2, 0x7fe2ebd17030), fd_no=1910(25375) DEBUG: <core> [receive.c:298]: receive_msg(): receive_msg: cleaning up14(25379) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9eeee0, 28, -1, 0x0) fd_no=20 called14(25379) DEBUG: <core> [tcp_main.c:4219]: handle_tcpconn_ev(): tcp: DBG: sending to child, events 114(25379) DEBUG: <core> [tcp_main.c:3902]: send2child(): selected tcp worker 2 12(25377) for activity on [tcp:192.168.1.195:4060 <http://192.168.1.195:4060>], 0x7fe2ebd1703012(25377) DEBUG: <core> [tcp_read.c:1554]: handle_io(): received n=8 con=0x7fe2ebd17030, fd=512(25377) DEBUG: <core> [tcp_read.c:1362]: tcp_read_req(): tcp_read_req: content-length= 38812(25377) DEBUG: <core> [parser/msg_parser.c:633]: parse_msg(): SIP Reply (status):12(25377) DEBUG: <core> [parser/msg_parser.c:635]: parse_msg(): version: <SIP/2.0>12(25377) DEBUG: <core> [parser/msg_parser.c:637]: parse_msg(): status: <183>12(25377) DEBUG: <core> [parser/msg_parser.c:639]: parse_msg(): reason: <Session Progress>12(25377) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK6423.c4c3d26baa62dea619271f186d6344fc.0>; state=612(25377) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 236, <i> = <2>; state=1612(25377) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=512(25377) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=212(25377) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via12(25377) DEBUG: <core> [receive.c:154]: receive_msg(): After parse_msg...12(25377) DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg: msg id=1 global id=0 T start=0xffffffffffffffff12(25377) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK1970975121>; state=1612(25377) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=512(25377) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=6212(25377) DEBUG: <core> [parser/msg_parser.c:526]: parse_headers(): parse_headers: this is the second via12(25377) DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=165643220412(25377) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=2912(25377) DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [43]; uri=[sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>] 12(25377) DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>>]12(25377) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <2> <INVITE>12(25377) DEBUG: tm [t_lookup.c:949]: t_reply_matching(): DEBUG: t_reply_matching: hash 12870 label 0 branch 012(25377) DEBUG: tm [t_lookup.c:1004]: t_reply_matching(): DEBUG: t_reply_matching: reply matched (T=0x7fe2ebd13848)!12(25377) DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x7fe2ebd13848, callback type 2, id 0 entered12(25377) DEBUG: acc [acc_logic.c:644]: tmcb_func(): acc callback called for t(0x7fe2ebd13848) event type 2, reply code 18312(25377) DEBUG: tm [t_lookup.c:1141]: t_check_msg(): DEBUG: t_check_msg: msg id=1 global id=1 T end=0x7fe2ebd1384812(25377) DEBUG: tm [t_reply.c:2200]: reply_received(): DEBUG: reply_received: org. status uas=100, uac[0]=0 local=0 is_invite=1)12(25377) DEBUG: <script>: incoming reply12(25377) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=100, new_code=18312(25377) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=012(25377) DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=38812(25377) DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header12(25377) DEBUG: <core> [msg_translator.c:2278]: generate_res_buf_from_sip_res(): old size: 948, new size: 85312(25377) DEBUG: <core> [msg_translator.c:2296]: generate_res_buf_from_sip_res(): copied size: orig:125, new: 30, rest: 823 msg=SIP/2.0 183 Session ProgressVia: SIP/2.0/TCP 192.168.1.195:19009;branch=z9hG4bK1970975121Record-Route: <sip:192.168.1.195:4060;transport=tcp;lr>From: <sip:linux_ua_0@192.168.1.195 <sip%3Alinux_ua_0(a)192.168.1.195>>;tag=604263664To: <sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>>;tag=1656432204Call-ID: 734015646(a)192.168.1.195 <734015646(a)192.168.1.195>CSeq: 2 INVITEContact: <sip:linux_ua_1@192.168.1.195:40611;transport=TCP>Content-Type: application/sdpUser-Agent: Linux 64 US-UA 1.0.1Require: 100relRSeq: 1Content-Length: 388v=0o=- 0 0 IN IP4 192.168.1.195s=IMS_Callc=IN IP4 192.168.1.195t=0 0m=audio 21720 RTP/AVP 99 0 8 101 b=AS:64a=curr:qos remote nonea=curr:qos local nonea=des:qos mandatory remote sendrecva=conf:qos remote sendrecva=des:qos mandatory local sendrecva=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=rtpmap:99 AMR/8000a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/800012(25377) DEBUG: <core> [tcp_main.c:2297]: tcpconn_send_put(): tcp_send: tcp connection found (0x7fe2ebd096b0), acquiring fd12(25377) DEBUG: <core> [tcp_main.c:2307]: tcpconn_send_put(): tcp_send, c= 0x7fe2ebd096b0, n=1614(25379) DEBUG: <core> [tcp_main.c:3565]: handle_ser_child(): handle_ser_child: read response= 7fe2ebd096b0, 2, fd -1 from 12 (25377)12(25377) DEBUG: <core> [tcp_main.c:2337]: tcpconn_send_put(): tcp_send: after receive_fd: c= 0x7fe2ebd096b0 n=8 fd=812(25377) DEBUG: <core> [tcp_main.c:2516]: tcpconn_do_send(): tcp_send: sending...12(25377) DEBUG: <core> [tcp_main.c:2550]: tcpconn_do_send(): tcp_send: after real write: c= 0x7fe2ebd096b0 n=853 fd=812(25377) DEBUG: <core> [tcp_main.c:2551]: tcpconn_do_send(): tcp_send: buf=SIP/2.0 183 Session ProgressVia: SIP/2.0/TCP 192.168.1.195:19009;branch=z9hG4bK1970975121Record-Route: <sip:192.168.1.195:4060;transport=tcp;lr>From: <sip:linux_ua_0@192.168.1.195 <sip%3Alinux_ua_0(a)192.168.1.195>>;tag=604263664To: <sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>>;tag=1656432204Call-ID: 734015646(a)192.168.1.195 <734015646(a)192.168.1.195>CSeq: 2 INVITEContact: <sip:linux_ua_1@192.168.1.195:40611;transport=TCP>Content-Type: application/sdpUser-Agent: Linux 64 US-UA 1.0.1Require: 100relRSeq: 1Content-Length: 388v=0o=- 0 0 IN IP4 192.168.1.195s=IMS_Callc=IN IP4 192.168.1.195t=0 0m=audio 21720 RTP/AVP 99 0 8 101 b=AS:64a=curr:qos remote nonea=curr:qos local nonea=des:qos mandatory remote sendrecva=conf:qos remote sendrecva=des:qos mandatory local sendrecva=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=rtpmap:99 AMR/8000a=rtpmap:0 PCMU/8000a=rtpmap:8 PCMA/800012(25377) DEBUG: tm [t_hooks.c:288]: run_trans_callbacks_internal(): DBG: trans=0x7fe2ebd13848, callback type 512, id 0 entered12(25377) DEBUG: acc [acc_logic.c:644]: tmcb_func(): acc callback called for t(0x7fe2ebd13848) event type 512, reply code 18312(25377) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)12(25377) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)12(25377) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)12(25377) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)12(25377) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)12(25377) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)12(25377) DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil)12(25377) DEBUG: <core> [receive.c:298]: receive_msg(): receive_msg: cleaning up12(25377) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0xa33180, 5, 2, 0x7fe2ebd17030), fd_no=110(25375) DEBUG: <core> [tcp_read.c:1362]: tcp_read_req(): tcp_read_req: content-length= 39210(25375) DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request:10(25375) DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <PRACK>10(25375) DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: <sip:linux_ua_1@192.168.1.195:40611;transport=TCP>10(25375) DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0>10(25375) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK1576634775>; state=1610(25375) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=510(25375) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=210(25375) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via10(25375) DEBUG: <core> [receive.c:154]: receive_msg(): After parse_msg...10(25375) DEBUG: <core> [receive.c:197]: receive_msg(): preparing to run routing scripts...10(25375) DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=165643220410(25375) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=2910(25375) DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [43]; uri=[sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>] 10(25375) DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>>]10(25375) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <3> <PRACK>10(25375) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 10(25375) DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=39210(25375) DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header10(25375) DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=60426366410(25375) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=2910(25375) DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks result: 110(25375) DEBUG: siputils [checks.c:106]: has_totag(): totag found10(25375) DEBUG: rr [loose.c:90]: is_preloaded(): is_preloaded: No10(25375) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 13==13 && [192.168.1.195] == [192.168.1.195]10(25375) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 4060 (advertise 0) matches port 4061110(25375) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 13==13 && [192.168.1.195] == [192.168.1.195]10(25375) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 4060 (advertise 0) matches port 4061110(25375) DEBUG: <core> [socket_info.c:583]: grep_sock_info(): grep_sock_info - checking if host==us: 13==13 && [192.168.1.195] == [192.168.1.195]10(25375) DEBUG: <core> [socket_info.c:587]: grep_sock_info(): grep_sock_info - checking if port 4090 (advertise 0) matches port 4061110(25375) DEBUG: rr [loose.c:674]: after_strict(): Next hop: 'sip:192.168.1.195:4060;transport=tcp;lr' is loose router10(25375) DEBUG: rr [loose.c:724]: after_strict(): The last route URI: 'sip:192.168.1.195:4060;transport=tcp;lr'10(25375) DEBUG: tm [t_lookup.c:1373]: t_newtran(): DEBUG: t_newtran: msg id=4 , global msg id=3 , T on entrance=0xffffffffffffffff10(25375) DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start searching: hash=12871, isACK=010(25375) DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction matching failed10(25375) DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found10(25375) DEBUG: tm [t_hooks.c:380]: run_reqin_callbacks_internal(): DBG: trans=0x7fe2ebd1b780, callback type 1, id 0 entered10(25375) DEBUG: <core> [md5utils.c:67]: MD5StringArray(): DEBUG: MD5 calculated: 0dd1edaf37f3874c5176befe76ed5b3e10(25375) DEBUG: <core> [msg_translator.c:2893]: create_via_hf(): create_via_hf: id added: <;i=2>, rcv proto=210(25375) DEBUG: <core> [msg_translator.c:206]: check_via_address(): check_via_address(192.168.1.195, 192.168.1.195, 0)10(25375) DEBUG: <core> [tcp_main.c:1818]: tcp_send(): tcp_send: no open tcp connection found, opening new one10(25375) DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 192.168.1.19510(25375) DEBUG: <core> [tcp_main.c:1073]: tcpconn_new(): tcpconn_new: on port 4060, type 210(25375) DEBUG: <core> [tcp_main.c:1382]: tcpconn_add(): tcpconn_add: hashes: 3360:0:0, 410(25375) INFO: <core> [tcp_main.c:2753]: tcpconn_1st_send(): quick connect for 0x7fe2ebd1e5c010(25375) DEBUG: tm [t_funcs.c:394]: t_relay_to(): SER: new transaction fwd'ed14(25379) DEBUG: <core> [ip_addr.c:243]: print_ip(): tcpconn_new: new tcp connection: 192.168.1.19510(25375) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)14(25379) DEBUG: <core> [tcp_main.c:1073]: tcpconn_new(): tcpconn_new: on port 19785, type 210(25375) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)10(25375) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)10(25375) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)14(25379) DEBUG: <core> [tcp_main.c:1382]: tcpconn_add(): tcpconn_add: hashes: 3888:3539:528, 510(25375) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)10(25375) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)10(25375) DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil)10(25375) DEBUG: <core> [receive.c:298]: receive_msg(): receive_msg: cleaning up14(25379) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0x9eeee0, 29, 2, 0x7fe2ebd228e0), fd_no=1914(25379) DEBUG: <core> [tcp_main.c:3565]: handle_ser_child(): handle_ser_child: read response= 7fe2ebd1e5c0, 6, fd 30 from 10 (25375)14(25379) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0x9eeee0, 30, 2, 0x7fe2ebd1e5c0), fd_no=2014(25379) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9eeee0, 29, -1, 0x0) fd_no=21 called14(25379) DEBUG: <core> [tcp_main.c:4219]: handle_tcpconn_ev(): tcp: DBG: sending to child, events 114(25379) DEBUG: <core> [tcp_main.c:3902]: send2child(): selected tcp worker 3 13(25378) for activity on [tcp:192.168.1.195:4060 <http://192.168.1.195:4060>], 0x7fe2ebd228e013(25378) DEBUG: <core> [tcp_read.c:1554]: handle_io(): received n=8 con=0x7fe2ebd228e0, fd=513(25378) DEBUG: <core> [tcp_read.c:1362]: tcp_read_req(): tcp_read_req: content-length= 39213(25378) DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request:13(25378) DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <PRACK>13(25378) DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: <sip:192.168.1.195:4060;transport=tcp;lr>13(25378) DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0>13(25378) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK7423.0dd1edaf37f3874c5176befe76ed5b3e.0>; state=613(25378) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 236, <i> = <2>; state=1613(25378) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=513(25378) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=213(25378) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via13(25378) DEBUG: <core> [receive.c:154]: receive_msg(): After parse_msg...13(25378) DEBUG: <core> [receive.c:197]: receive_msg(): preparing to run routing scripts...13(25378) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK1576634775>; state=1613(25378) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=513(25378) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=10013(25378) DEBUG: <core> [parser/msg_parser.c:526]: parse_headers(): parse_headers: this is the second via13(25378) DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=165643220413(25378) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=2913(25378) DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [43]; uri=[sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>] 13(25378) DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>>]13(25378) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <3> <PRACK>13(25378) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 69 13(25378) DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=39213(25378) DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header13(25378) DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=60426366413(25378) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=2913(25378) DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks result: 113(25378) DEBUG: siputils [checks.c:106]: has_totag(): totag found13(25378) DEBUG: rr [loose.c:113]: find_first_route(): No Route headers found13(25378) DEBUG: rr [loose.c:929]: loose_route(): There is no Route HF13(25378) DEBUG: <core> [msg_translator.c:206]: check_via_address(): check_via_address(192.168.1.195, 192.168.1.195, 0)13(25378) DEBUG: <core> [tcp_main.c:2281]: tcpconn_send_put(): tcp_send: send from reader (25378 (13)), reusing fd13(25378) DEBUG: <core> [tcp_main.c:2516]: tcpconn_do_send(): tcp_send: sending...13(25378) DEBUG: <core> [tcp_main.c:2550]: tcpconn_do_send(): tcp_send: after real write: c= 0x7fe2ebd228e0 n=386 fd=514(25379) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9eeee0, 30, -1, 0x0) fd_no=20 called13(25378) DEBUG: <core> [tcp_main.c:2551]: tcpconn_do_send(): tcp_send: buf=SIP/2.0 404 Not hereVia: SIP/2.0/TCP 192.168.1.195:4060;branch=z9hG4bK7423.0dd1edaf37f3874c5176befe76ed5b3e.0;i=2Via: SIP/2.0/TCP 192.168.1.195:19009;branch=z9hG4bK1576634775From: <sip:linux_ua_0@192.168.1.195 <sip%3Alinux_ua_0(a)192.168.1.195>>;tag=604263664To: <sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>>;tag=1656432204Call-ID: 734015646(a)192.168.1.195 <734015646(a)192.168.1.195>CSeq: 3 PRACKServer: kamailio (4.2.3 (x86_64/linux))Content-Length: 014(25379) DEBUG: <core> [tcp_main.c:4219]: handle_tcpconn_ev(): tcp: DBG: sending to child, events 114(25379) DEBUG: <core> [tcp_main.c:3898]: send2child(): WARNING: send2child: no free tcp receiver, connection passed to the least busy one (1)14(25379) DEBUG: <core> [tcp_main.c:3902]: send2child(): selected tcp worker 0 10(25375) for activity on [tcp:192.168.1.195:4060 <http://192.168.1.195:4060>], 0x7fe2ebd1e5c013(25378) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)13(25378) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)10(25375) DEBUG: <core> [tcp_read.c:1554]: handle_io(): received n=8 con=0x7fe2ebd1e5c0, fd=1013(25378) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)13(25378) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)13(25378) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)13(25378) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)10(25375) DEBUG: <core> [tcp_read.c:1362]: tcp_read_req(): tcp_read_req: content-length= 013(25378) DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil)10(25375) DEBUG: <core> [parser/msg_parser.c:633]: parse_msg(): SIP Reply (status):13(25378) DEBUG: <core> [receive.c:298]: receive_msg(): receive_msg: cleaning up10(25375) DEBUG: <core> [parser/msg_parser.c:635]: parse_msg(): version: <SIP/2.0>10(25375) DEBUG: <core> [parser/msg_parser.c:637]: parse_msg(): status: <404>10(25375) DEBUG: <core> [parser/msg_parser.c:639]: parse_msg(): reason: <Not here>10(25375) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK7423.0dd1edaf37f3874c5176befe76ed5b3e.0>; state=613(25378) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0xa33180, 5, 2, 0x7fe2ebd228e0), fd_no=110(25375) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 236, <i> = <2>; state=1610(25375) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=510(25375) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=210(25375) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via10(25375) DEBUG: <core> [receive.c:154]: receive_msg(): After parse_msg...10(25375) DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg: msg id=5 global id=4 T start=0xffffffffffffffff10(25375) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK1576634775>; state=1610(25375) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=510(25375) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=6210(25375) DEBUG: <core> [parser/msg_parser.c:526]: parse_headers(): parse_headers: this is the second via10(25375) DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=165643220410(25375) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=2910(25375) DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [43]; uri=[sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>] 10(25375) DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [<sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>>]10(25375) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <3> <PRACK>10(25375) DEBUG: tm [t_lookup.c:949]: t_reply_matching(): DEBUG: t_reply_matching: hash 12871 label 0 branch 010(25375) DEBUG: tm [t_lookup.c:1004]: t_reply_matching(): DEBUG: t_reply_matching: reply matched (T=0x7fe2ebd1b780)!10(25375) DEBUG: tm [t_lookup.c:1141]: t_check_msg(): DEBUG: t_check_msg: msg id=5 global id=5 T end=0x7fe2ebd1b78010(25375) DEBUG: tm [t_reply.c:2200]: reply_received(): DEBUG: reply_received: org. status uas=0, uac[0]=0 local=0 is_invite=0)10(25375) DEBUG: tm [t_reply.c:1294]: t_should_relay_response(): ->>>>>>>>> T_code=0, new_code=40410(25375) DEBUG: tm [t_reply.c:1812]: relay_reply(): DEBUG: relay_reply: branch=0, save=0, relay=0 icode=010(25375) DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=010(25375) DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header10(25375) DEBUG: <core> [msg_translator.c:2278]: generate_res_buf_from_sip_res(): old size: 386, new size: 29110(25375) DEBUG: <core> [msg_translator.c:2296]: generate_res_buf_from_sip_res(): copied size: orig:117, new: 22, rest: 269 msg=SIP/2.0 404 Not hereVia: SIP/2.0/TCP 192.168.1.195:19009;branch=z9hG4bK1576634775From: <sip:linux_ua_0@192.168.1.195 <sip%3Alinux_ua_0(a)192.168.1.195>>;tag=604263664To: <sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>>;tag=1656432204Call-ID: 734015646(a)192.168.1.195 <734015646(a)192.168.1.195>CSeq: 3 PRACKServer: kamailio (4.2.3 (x86_64/linux))Content-Length: 010(25375) DEBUG: <core> [mem/shm_mem.c:111]: _shm_resize(): WARNING:vqm_resize: resize(0) called10(25375) DEBUG: <core> [tcp_main.c:2281]: tcpconn_send_put(): tcp_send: send from reader (25375 (10)), reusing fd10(25375) DEBUG: <core> [tcp_main.c:2516]: tcpconn_do_send(): tcp_send: sending...10(25375) DEBUG: <core> [tcp_main.c:2550]: tcpconn_do_send(): tcp_send: after real write: c= 0x7fe2ebd096b0 n=291 fd=510(25375) DEBUG: <core> [tcp_main.c:2551]: tcpconn_do_send(): tcp_send: buf=SIP/2.0 404 Not hereVia: SIP/2.0/TCP 192.168.1.195:19009;branch=z9hG4bK1576634775From: <sip:linux_ua_0@192.168.1.195 <sip%3Alinux_ua_0(a)192.168.1.195>>;tag=604263664To: <sip:linux_ua_1@192.168.1.195 <sip%3Alinux_ua_1(a)192.168.1.195>>;tag=1656432204Call-ID: 734015646(a)192.168.1.195 <734015646(a)192.168.1.195>CSeq: 3 PRACKServer: kamailio (4.2.3 (x86_64/linux))Content-Length: 010(25375) DEBUG: tm [t_reply.c:1653]: cleanup_uac_timers(): DEBUG: cleanup_uac_timers: RETR/FR timers reset10(25375) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)10(25375) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)10(25375) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)10(25375) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)10(25375) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)10(25375) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)10(25375) DEBUG: <core> [xavp.c:448]: xavp_destroy_list(): destroying xavp list (nil)10(25375) DEBUG: <core> [receive.c:298]: receive_msg(): receive_msg: cleaning up10(25375) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0xa33180, 10, 2, 0x7fe2ebd1e5c0), fd_no=214(25379) CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 1314(25379) DEBUG: <core> [tcp_main.c:3537]: handle_ser_child(): DBG: handle_ser_child: dead child 4, pid 25369 (shutting down?)14(25379) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9eeee0, 13, -1, 0x0) fd_no=19 calledroot@Kamal:/usr/local/etc/kamailio# *

2 1

Increase Simultaneous Registration
by Austin Einter 04 Jun '15

04 Jun '15

Hi All I am using Kamailio 4.0.4 (x86_64/linux). I wanted to do simultaneous registrations of large number of endpoints. Many times I see only 2048 endpoints registered and for other endpoints Kamailio did not respond for REGISTER message. I am tcp as transport. Now my questions are 1. What are the different configurations available to increase Kamailio performance 2. If I put Kamailio in a Linux 64 Bit, 2GHz, 8GB RAM machine, how much simultaneous and total registrations can be achieved. I have tuned system file FDs, tcp related configurations like ephemeral port range, and other required configurations. I hope there should be some configurations in Kamailio to achieve higher throughput , can somebody help me out. Thanks Austin

2 1

Problem with command kamcmd pkg.stats on Kamailio 4.2.5
by Dao Hai Dang 04 Jun '15

04 Jun '15

Hi, I need help with kamcmd pkg.stats on Kamailio 4.2.5. When I used this command to check kamailio process, I get an entry with no pid and total_size param. I used kamcmd ps and the process error was MI FIFO. Same thing happened when I added snmpstats module, I get process SNMP AgentX with no pid and total_size param. This didn't happen when I use kamailio 4.1.x . Below are logs : [root@localhost ~]# kamcmd pkg.stats { entry: 0 pid: 1532 rank: 0 used: 376224 free: 3718760 real_used: 475544 total_size: 4194304 total_frags: 4 } { entry: 1 pid: 1534 rank: 1 used: 384496 free: 3709368 real_used: 484936 total_size: 4194304 total_frags: 22 } { entry: 2 pid: 1535 rank: -1 used: 384552 free: 3710096 real_used: 484208 total_size: 4194304 total_frags: 8 } { entry: 3 pid: 1536 rank: -1 used: 384552 free: 3710096 real_used: 484208 total_size: 4194304 total_frags: 8 } { entry: 4 pid: 0 rank: 0 used: 393744 free: 3701072 real_used: 493232 total_size: 0 total_frags: 4 } { entry: 5 pid: 1543 rank: -2 used: 383592 free: 3710552 real_used: 483752 total_size: 4194304 total_frags: 8 } { entry: 6 pid: 1544 rank: -4 used: 454392 free: 3640424 real_used: 553880 total_size: 4194304 total_frags: 4 } [root@localhost ~]# kamcmd ps 1532 attendant 1534 udp receiver child=0 sock=10.84.11.116:5060 1535 slow timer 1536 timer 1537 MI FIFO 1543 ctl handler 1544 tcp main process Thanks & Regards, Dao Hai Dang -------------------------------------------------------------------------------- Solutions & Software Development Center | VNPT Technology A member of VietNam Posts & Telecomunications Group Addr : 124 Hoang Quoc Viet, Ha Noi Mobile : (+84 )912 363 579 - Skype : cnoob89 Office : (+84) 438 36 20 94 - ext 1222

2 6

Strange core.tcp_list / tls.list output
by Mikko Lehto 04 Jun '15

04 Jun '15

Hi, I noticed thath command core.tcp_list in kamcmd gives output that does not make sense to me: --- kamcmd> core.tcp_list { id: 1 type: TLS state: CONN_OK timeout: 3599 ref_count: 1 src_ip: 192.168.47.132 src_port: 5061 dst_ip: 10.10.235.9 dst_port: 54769 } { id: 2 type: TLS state: CONN_OK timeout: 3599 ref_count: 1 src_ip: 192.168.47.132 src_port: 32854 dst_ip: 10.10.235.9 dst_port: 5041 } --- In my scenario I am connecting via TLS to host 192.168.47.132:5061. And host 192.168.47.132 is connecting to my proxy host 10.10.235.9:5041. pcap verifies this. Also tls.list gives similar output. Values regarding incoming connection (id=2) makes total sense. It seems that values regarding outgoing connection (id=1) have dst_* values swapped with src_* values. To debug, I placed this logging to tcp_main.c: --- $ git diff tcp_main.c diff --git a/tcp_main.c b/tcp_main.c index 5830c8e..3717cf6 100644 --- a/tcp_main.c +++ b/tcp_main.c @@ -1207,6 +1207,7 @@ again: } from=&my_name; /* update from with the real "from" address */ su2ip_addr(&ip, &my_name); + LM_WARN("after getsockname() my_name=%s server=%s ip=%s\n", su2a(&my_name, sizeof(my_name)), su2a(server, sizeof(*server)), ip_addr2a(&ip)); find_socket: #ifdef USE_TLS if (unlikely(type==PROTO_TLS)) @@ -1223,6 +1224,7 @@ find_socket: else *res_si=sendipv6_tcp; } *res_local_addr=*from; + LM_WARN("before return my_name=%s server=%s ip=%s\n", su2a(&my_name, sizeof(my_name)), su2a(server, sizeof(*server)), ip_addr2a(&ip)); return s; error: if (s!=-1) tcp_safe_close(s); --- This gives following output: WARNING: <core> [tcp_main.c:1210]: tcp_do_connect(): after getsockname() my_name=192.168.47.132:5061 server=192.168.47.132:5061 ip=10.10.235.9 WARNING: <core> [tcp_main.c:1227]: tcp_do_connect(): before return my_name=192.168.47.132:5061 server=192.168.47.132:5061 ip=10.10.235.9 Are tcp endpoint values really wrong way around or did I undertand something wrong about their meaning? I am running Kamailio 4.2.5 (x86_64/linux) 61d84c-dirty from git. -- Mikko Lehto

2 3

Kamailio Contact header read-only
by Ali Taher 04 Jun '15

04 Jun '15

Hello, I'm trying to change contact header in SIP response message by stripping first characters (for testing only) as below : $ct = $(ct{s.strip,25}); But I got below error saying that $ct is read-only: 0(15410) : <core> [cfg.y:3419]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 575, column 5-7: read only pvar in assignment left side ERROR: bad config file (1 errors) The question is how can I change contact header before it's sending of Kamailio? Thanks Ali

2 7

SIP Auth Realm - From domain or Request domain ?
by jay binks 04 Jun '15

04 Jun '15

I have a questions in regards to SIP Auth, and specifically how Asterisk 13 with PJSIP appears to send Invites. It seems that the typical Kamailio config sends the SIP auth challenge with the from domain as the auth realm. however in the case of the Asterisk 13 ( PJSIP ) invite, the from domain is an RFC1918 IP address and I would have to use the domain from the request. What im wanting to know is what the consensus is on where the auth realm comes from. does everyone 100% agree that you send auth using the SIP From header, or is this something that some do Request domain ? What im considering doing is to use the from domain, but if from domain = RFC1918 then use request domain. Im just after everyone's opinion on how to best handle this invite, from a auth_challenge perspective. ( not so much if the Asterisk 13 PJSIP module is doing the right or wrong thing ) INVITE sip:555111222333@sip.mytestdomain.com:5060 SIP/2.0 Via: SIP/2.0/UDP 180.111.222.333:5060;rport;branch=z9hG4bKPjafb111a9-aaaf-4c9e-b456-748514855a06 From: <sip:555111222333@172.18.90.1 >;tag=88317259-052b-4997-bd2e-9c8836e27b47 To: <sip:555111222333@sip.netsip.net.au> Contact: <sip:9d43c8df-6378-4266-89c1-0a0bb0e1d0d1@180.111.222.333:5060> Call-ID: b7da52e2-dfd1-434e-a2c2-b28e1683ce0e -- Sincerely Jay

2 1

Fwd: Kamailio start failed
by Gopalakrishnan N 03 Jun '15

03 Jun '15

Hi, I have added the ASTERISK integration in kamailio.cfg, after adding those, am not able to start kamailio service, its getting failed, attached is the script. Any help would be appreciated. Thank you with regards, Gopalkrishnan N. Mob: +91 99404 91346 VoIP call - sip:saigop@gtalk2voip.com

2 4

Need help to configure rtpporyx
by Mir Jee 03 Jun '15

03 Jun '15

Hi! Please help me to configure rtpproxy on centos + kamailio.

2 1

lookup() to fork execution of routing script?
by Andrey Utkin 03 Jun '15

03 Jun '15

Hi! Currently we have the following logic in our Kamailio routing script. On incoming INVITE, we lookup() the location of called party, then we analyse the updated Request-URI ($ru) regarding callee contact protocol: is it websocket-based or not, to distinguish web-based useragents from classic SIP agents and provide RTP profiles interconnection with rtpengine (RTP/SAVPF for web-based, RTP/AVPF for the rest). After that, we relay the mangled INVITE to callee location(s) with t_relay(). The issue arises when callee has both websocket-based and classic useragents registered simultaneously. According to lookup() manual, it replaces Req-URI with just one contact, and somehow also attach metainformation about additional locations, so that t_relay() forks the call. lookup() does not provide the possibility to fork the execution of routing script, so that we could execute our routing logic described above. But I would love this to be possible, because the only alternative way for calling to multiple locations seems to be the serial forking described in TM module manual. But we don't need serial calling contact-by-contact, waiting for call failures, we need parallel calling to all available user locations. So we need an advice how to go. Is there a way to implement what we need at scripting level, or should this feature be implemented in Kamailio code (or is it already there?). Thanks in advance for any help. -- Andrey Utkin

2 2

Kamailio LCR Multiple choices
by Ali Taher 03 Jun '15

03 Jun '15

Hello, I'm using LCR module to send multiple choices. Everything works fine so far , yet I have a problem that last gateway comes first, then gateways are ordered correctly. Below is correct sorting of gateways: First gateway : 192.168.26.4 Second gateway: 192.168.26.1 Third gateway : 192.168.26.6 Below is Invite response I'm getting, as you can see 192.168.26.6 (last gateway) is included as the first gateway , then 192.168.26.4 (first gateway) comes next. Via: SIP/2.0/UDP 192.168.26.3:37066;branch=z9hG4bK-d8754z-885c37410729cc3b-1---d8754z-;rport= 37066;received=192.168.26.3 To: "9612"<sip:9612@192.168.110.181>;tag=b27e1a1d33761e85846fc98f5f3a7e58.8d8f From: "5000"<sip:5000@192.168.110.181>;tag=aa37be29 CSeq: 1 INVITE Call-ID: NTY0YjVmYzZiNzc3ZDA3ZDA3MmM0OGM1ZDBhZGZhNTM. Contact: <sip:9612@192.168.26.6>, <sip:9612@192.168.26.4>, <sip:9612@192.168.26.1>, <sip:9612@192.168.26.6> Below is routing block I'm using : if (is_method("INVITE")) { if (!load_gws(1, $rU, $fu)) { sl_send_reply("502", "Unable to load Gateways"); exit; } else while(next_gw()) { append_branch(); } sl_send_reply("300","Multiple Choices"); exit; } Appreciate your help. Regards, Ali

2 7

delay on "location" table update
by Al S 03 Jun '15

03 Jun '15

Hi, I was wondering if it's normal for "location" table to be updated almost 10 sec after receiving REGISTER. Thanks,Al

2 1

Potential CNXCC bug
by Jay Patel 02 Jun '15

02 Jun '15

I am trying to use cnxcc for credit control . It seems like dialog and cnxcc modules are getting out of sync. For some reason , I see cnxcc.active_clients gets stuck and they never clear up. Interestingly only calls that gets failure 5xx and 4xx from PSTN gateway gets stuck, all successful calls are cleared . I have made sure , I have setflag(DLG_FLAG) 1st line in my routing logic. If you need my routing script, I can share with you . Am I missing anything obvious? As you can see , "dlg.list" is empty, ( no calls in progress in system.) kamcmd> dlg.list kamcmd> But , two user are stuck ( 1 with 6 calls in progress and another with 2 calls in progress) kamcmd> cnxcc.active_clients client_id:917044056840,number_of_calls:6,concurrent_calls:0,type:1,max_amount:0.000000,consumed_amount:0.000000; client_id:971554578191,number_of_calls:2,concurrent_calls:0,type:1,max_amount:0.000000,consumed_amount:0.000000; Here are the call dumps for both users. kamcmd> cnxcc.check_client s:971554578191 id:0,confirmed:no,local_consumed_amount:0.588000,global_consumed_amount:0.000000,local_max_amount:7.690000,global_max_amount:0.00000 id:1,confirmed:no,local_consumed_amount:0.588000,global_consumed_amount:0.000000,local_max_amount:7.690000,global_max_amount:0.00000 kamcmd> kamcmd> kamcmd> cnxcc.check_client s:917044056840 id:0,confirmed:no,local_consumed_amount:0.588000,global_consumed_amount:0.000000,local_max_amount:9.990000,global_max_amount:0.00000 id:1,confirmed:no,local_consumed_amount:14.100000,global_consumed_amount:0.000000,local_max_amount:9.990000,global_max_amount:0.0000 id:2,confirmed:no,local_consumed_amount:14.100000,global_consumed_amount:0.000000,local_max_amount:9.990000,global_max_amount:0.0000 id:3,confirmed:no,local_consumed_amount:14.100000,global_consumed_amount:0.000000,local_max_amount:9.990000,global_max_amount:0.0000 id:4,confirmed:no,local_consumed_amount:14.100000,global_consumed_amount:0.000000,local_max_amount:9.990000,global_max_amount:0.0000 id:5,confirmed:no,local_consumed_amount:0.588000,global_consumed_amount:0.000000,local_max_amount:9.990000,global_max_amount:0.00000 kamcmd>

1 0

Kamailio LCR rule matching
by Ali Taher 02 Jun '15

02 Jun '15

Hi, I'm using LCR module in Kamailio , where matching of the rule (From-URI in LCR-rule table) is currently based on user part of sip address in From field (highlighted below in blue). While we need to change the identification to be on the otg value sent in the sip address in From field (highlighted below in yellow). INVITE sip:86532917453592@172.16.5.108:5060 <http://sip:86532917453592@172.16.5.108:5060> SIP/2.0 Via: SIP/2.0/UDP 172.16.5.112:5060;branch=z9hG4bK1sansay9906915rdb2243 Via: SIP/2.0/UDP 216.53.4.5:5060;branch=z9hG4bK_1852370774_4741_1 Record-Route: <sip:sansay9906915rdb2243@172.16.5.112:5060;lr;transport=udp> To: <sip:86532917453592@69.41.186.186:5060 <http://sip:86532917453592@69.41.186.186:5060> > From: <sip: <http://sip:+249964642906@216.53.4.5:5060> +249964642906@216.53.4.5:5060;otg=024050>;tag=1852370774_C Call-ID: sbcsipuac.2_169.132.137.51_b55sb12_1_1_2015052909455960_1852370774_487698 CSeq: 1 INVITE Contact: <sip:+249964642906@216.53.4.5:5060 <http://sip:+249964642906@216.53.4.5:5060> > P-Asserted-Identity: <sip:+249964642906@216.53.4.5:5060 <http://sip:+249964642906@216.53.4.5:5060> > Privacy: none User-Agent: VCS 5.9.2.42-01 Allow: INVITE, ACK, CANCEL, BYE Max-Forwards: 69 Content-Length: 0 How can we achieve this ? Any help will be appreciated. Regards, Ali

3 2

RTPengine+Kamailio, 200 OK without DTLS fingerprint
by Vasiliy Ganchev 02 Jun '15

02 Jun '15

Hello list! Have some difficulties with RTPengine: My test scheme is: UA(WS) -> Kamailio (WS proxy) -> Kamailio (SIP REGISTRAR, proxy) -> ASTERISK (media server) -> UA (SIP) Kamailio (WS) use RTPengine. Kamailio (WS proxy) - 4.2.3 Kamailio (SIP REGISTRAR, proxy) - 4.1.4 RTPengine - 3.3.0.0+0~mr3.8.0.0 I make a test call and have troubles in negotiation between UA(WS) -> Kamailio (WS proxy): UA(WS) -> INVITE -> Kamailio WS ...... rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force"); ...... this INVITE consists in SDP section a=fingerprint:sha-256...... When Kamailio (WS) receives 200 OK, it is also handled by RTPengine ....... rtpengine_manage("trust-address replace-origin replace-session-connection ICE=force"); ....... but UA (WS) receives 200OK without fingerprint, and log an error: Failed to set remote answer sdp: Called with SDP without DTLS fingerprint. Currently spend a lot of time reading Kamailio/RTPengine documentation/mail-list - but without success If someone have some hints, I would appreciate any help. -- View this message in context: http://sip-router.1086192.n5.nabble.com/RTPengine-Kamailio-200-OK-without-D… Sent from the Users mailing list archive at Nabble.com.

3 8

RFC 6140
by Al S 02 Jun '15

02 Jun '15

Hi, I was wondering if Kamailio support RFC 6140 to register Addresses of Record (AORs) in bulk map to a unique set of contacts. REGISTER sip:kamailio.example.com SIP/2.0 Via: SIP/2.0/UDP 198.51.100.3:5060;branch=z9hG4bKnashds7 Max-Forwards: 70 To: <sip:pbx@kamailio.example.com> From: <sip:pbx@kamailio.example.com>;tag=a23589 Call-ID: 843817637684230@998sdasdh09 CSeq: 1826 REGISTER Proxy-Require: gin Require: gin Supported: path Contact: <sip:198.51.100.3:5060;bnc> Expires: 7200 Content-Length: 0 Thanks,Al

2 3

next via branch value
by Juha Heinanen 01 Jun '15

01 Jun '15

when kamailio forwards invite request, it adds to it a new via header. for example, incoming invite: SIP/2.0/TCP 192.168.43.192:48089;branch=z9hG4bK4e1ab219cbde6190;rport outgoing invite: SIP/2.0/TCP 192.98.102.10;branch=z9hG4bK27b9.597e97c43da7ddd6be8520b662d616b9.0;i=5 SIP/2.0/TCP 192.168.43.192:48089;received=192.98.102.10;branch=z9hG4bK4e1ab219cbde6190;rport=49582 is it somehow possible to find out in the config script, what will the branch value of the new via header be? if that value would be known, it could be assigned to rtpengine extra_id_pv pseudo variable before making the offer and could then be used in rtpengine_delete to delete the right branch. if the new via branch value cannot be known, any suggestions on how to uniquely identify the outgoing branched of the invite? -- juha

4 14

Request of advice about TAS for IMS
by dfretes＠ing.una.py 01 Jun '15

01 Jun '15

I am trying to setup a really simple (I hope so) IMS platform, with some basic services (like voicemail, IVR, videoconference), but I'm kind of stuck. I have installed Kamailio IMS on debian and a OpenIMS VM also. But my little problem begins when I try to configure a application server (in this case, a TAS). I have tried with Elastix 3.0 (kamailio+asterisk), asterisk alone, and read some about Restcomm. For Elastix and asterisk, I get stuck trying to setup the trigger point. For Restcomm, I only find documentation for integration with Clearwater IMS. So, I need your help. Somebody has something like that working? What TAS (or components for a TAS) do you recommend? Tutorials, examples, manuals that you know can help me? Thanks for reading, and many many thanks for any advice. David

2 1

Select a network
by Nelson Migliaro 01 Jun '15

01 Jun '15

Hello everybody, I need to apply a specific configuration to softphones registering from a specific network range. Is there something like:

2 2

Invalid SDP line (no '=' delimiter) error !
by Badri Ranganathan 01 Jun '15

01 Jun '15

Hi all, I am answering a SIP call made to my softphone. The softphone rings and also sends a 200 OK correctly. But the other end that made the call is seeing a failure in the SDP of the 200 OK packet. This error is seen in SDP header => Invalid SDP line (no '=' delimiter) error w.r.t w_frame4_public_ip.pcap It looks like the failure as seen in wireshark is because line "audio 5000 RTP/AVP 8 101" should really be "m= audio 5000 RTP/AVP 8 101" so it doesn't see the '=' delimiter. But if you look at w_frame52_local_ip.pcap, the packets sent from the local IP has the contents correct. It is only after passing the public IP and at the network premises, that this failure is reported by the Session Border controller. I am not able to understand why a correctly formed packet sent out is seen "corrupted" at the network end. Any ideas really appreciated. Thanks, Badri.

2 1

Re: [SR-Users] Regarding presence conference
by Daniel-Constantin Mierla 01 Jun '15

01 Jun '15

Hello, the conference mixer is supposed to send PUBLISH requests with information about who is in the conference room, who is speaking at that moment and so. That is not the job of the presence_conference module. Eventually, that could be a pua_conference module that has to be implemented. The presence_conference module was implemented as part of GSoC and the PUA was supposed to be inside SEMS, but the other project failed. If you want to get it all working, you have to implement the PUA either in your version of rtpproxy or part of kamailio as well. When all in kamailio, you can see the pairs of modules such as presence_dialoginfo and pua_dialoginfo. You already have presence_confgerece. Cheers, Daniel On 28/05/15 14:28, Surendra Pullaiah wrote: > > Hello Daniel, > > > > I am implementing conference bridge using kamalio and > RTP porxy without media mixing. Since for now we are ok with this. > My question is as per the flow of RFC 4579, UA may send SUBSCRIBE to > the conference URI with event as conference. In kamalio modules > presence_conference where they mentioned such way that it will support > the same what I am looking for. But I simulated using sipP by sending > subscriber with conference event. I am not seeing any notify XML which > will have the conference users list. One more doubt where it will pick > the conference users, since I am not seeing any specific data base > table for the same. > > > > Can you please clarify what exactly for > presence_conference module. > > > > Note: I have loaded three modules presence, presence_xml, > presence_conference, in route config I am calling handle_subscribe > when SUBCRIBE method receives. > > > > Regards > > Surendra > > India: +91-8124625001 > > > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

1 0

Forward register to multiple Asterisk
by Nelson Migliaro 01 Jun '15

01 Jun '15

Hello, I am trying to forward the register to multiple Asterisk using Kamailio. The basic Idea is: Softphone A---->Kamailio-----> Asterisk 1 ------> Asterisk 2. I follow Asipto howto http://lylix.net/kamailio, but the problem I see in my case, is that registration IP in Asterisk is Kamailio´s IP. This causes calls going from kamailio to asterisk and then back to kamailio. In my scenario, I need that softphone A IP appeard in Asterisk realtime contact IP. The call path would be: Softphone B -> kamailio -> Asterisk 1 or Asterisk 2 (depending on dispatcher) -> Softphone A Following Asipto how to the call goes Softphone A -> kamailio -> asterisk -> kamailio Softphone B Hope I explain myself clearly. Thank you in advance.

2 1

Kamailio , RTPProxy and NAT
by Mir Jee 01 Jun '15

01 Jun '15

HI! I need help to configure RTPProxy. I have working Kamailio on Centos 6.5, On LAN everything is working fine. Now I want my other users to connect using 3G or any other network. Please help me with that. Statis Public IP (DSL Modem) ----- Proxy Box with 2 Lan cards --- 192.168.2.0 and 192.168.0.0 (Lan Side)

2 1

Kamailio 300 Multiple choices
by Ali Taher 29 May '15

29 May '15

Hello, We need to integrate Kamailio with an exchange switch in order to have routing decision. Kamailio must send to the switch the list of available gateways in the contact tag of SIP response as shown in the below example: SIP/2.0 300 Multiple choices Via: SIP/2.0/UDP 172.16.5.112:5060;branch=z9hG4bK1sansay9906915rdb2243 Via: SIP/2.0/UDP 216.53.4.5:5060;branch=z9hG4bK_1852370774_4741_1 To: <sip:86532917453592@69.41.186.186:5060 <http://sip:86532917453592@69.41.186.186:5060> > From: <sip:+249964642906@216.53.4.5:5060;otg=024050>;tag=1852370774_C Call-ID: sbcsipuac.2_169.132.137.51_b55sb12_1_1_2015052909455960_1852370774_487698 CSeq: 1 INVITE Contact: <sip:2917453592@036310>, <sip:2917453592@016500>, <sip:138295#2917453592@044610>, <sip:2917453592@020100> Server: Invoice SIP-server Content-Length: 0 Is this can be done through drouting module , or we need to use another module ? Your help is highly appreciated. Regards, Ali

2 7

"Multiple Phone Numbers" registration
by Al S 29 May '15

29 May '15

Hi, Does Kamilio have a predefined module to support "Multiple Phone Numbers" registration with a single Registration?I am trying to register multiple numbers on behalf of IP-PBXs. Thanks,Al

2 3

hash table not loading
by Olli Attila 29 May '15

29 May '15

Hello, I am trying to load a test hash table on kamailio startup but I keep getting the following error with kamctl after i restart Kamailio and try to list all the tables in memory: kamcmd> htable.listTables error: 500 - No htables I have the following configured in kamailio.cfg: # ---- loading the htable module loadmodule "htable.so" # ----- htable params ----- modparam("htable", "htable", "abc=>size=8;autoexpire=0;dbtable=htable;") modparam("htable", "db_url", DBURL) modparam("htable", "fetch_rows", 1000) The DBURL is pointed to the kamailio database on the same machine. #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!endif I have also added the following values for testing purposes to the kamailio.htable: id key_name key_type value_type key_value expires ------ ---------- -------- ---------- --------- --------- 1 test 0 0 12345 0 Is there anything else that i might be missing here? Im running kamailio v.4.2.4. Cheers, -- Olli

3 8

Kamailio V4.2.5 append prefix to Request Uri problem.
by Yasin CANER 29 May '15

29 May '15

Hello; i try to append prefix to $rU like below in Kamailio V4.2.5 . i get error even if i set string value. it works only when i append like that $rU="3002"+$rU; --------- 1 test------------ $var(prefix)=$avp(s:69); $rU=$var(prefix)+$rU; -FAILED ---------2 test ------------ $rU=$avp(s:69)+$rU; -FAILED ------- 3 test ------------ $rU=$avp(i:69)+$rU; -FAILED -------ERROR--------- May 29 12:06:57 group1-imslab11 /usr/local/sbin/kamailio[26196]: ERROR: pv [pv_core.c:2115]: pv_set_ruri_user(): str value required to set R-URI user May 29 12:06:57 group1-imslab11 /usr/local/sbin/kamailio[26196]: ERROR: <core> [lvalue.c:363]: lval_pvar_assign(): setting pvar failed May 29 12:06:57 group1-imslab11 /usr/local/sbin/kamailio[26196]: ERROR: <core> [lvalue.c:416]: lval_assign(): assignment failed at pos: (790,7-790,22) it work on Version 4.1.7 like below correctly; $rU=$avp(i:69)+$rU; Thanks Yasin

3 3

Kamailio users creation mechanism
by Alexandru Covalschi 29 May '15

29 May '15

Hello! My goal is to create a mechanism for dynamic creation/deletion of SIP users on Kamailio. First question: is it possible to create users not via kamctl command, but directly inserting new entries to this tables? And the second one: if it is possible, which hash alghorythm Kamailio uses when creating a new user on subscriber table? Becase I see 2 fields with hashes. Thank you -- Alexandru Covalschi VoIP engineer and system administrator phone: +37367398493 <http://abs-telecom.com/>

2 1

Elastix MT (3.0) issue with NAT (ACK is not received for 200 OK)
by Mainul Haque 28 May '15

28 May '15

Hello. I think I understand the problem, but I do not know how to fix it. I have done the following when the problem accord: -Download Elastix MT 64bit (downloaded from - http://www.elastix.com/en/downloads/) -Install it in VMware ESXi 5.1. Elastix works fine within the LAN -Then I pushed the VM to EC2 by exporting VMware OVF -I noticed that I could not registered my softphone (eyebeam 1.5) -After doing research online, I had to modify /etc/kamailio/kamailio-mhomed-elastix.cfg to know about the new IP of EC2 instance. See below of what I had to change: Old settings: [root@new-host-2 endpoint_configurator]# grep 192 /etc/kamailio/kamailio-mhomed-elastix.cfg if (is_in_subnet($var(target_remote_ip), "192.168.1.0/24")) { $var(rtpproxy_if) = "192.168.1.62"; $var(rtpproxy_if) = "192.168.1.6"; To New Settings: [root@new-host-2 ~]# grep 172 /etc/kamailio/kamailio-mhomed-elastix.cfg if (is_in_subnet($var(target_remote_ip), "172.31.48.0/18")) { $var(rtpproxy_if) = "172.31.55.101"; $var(rtpproxy_if) = "172.31.55.101"; -Now I can register my eyebeam softphone. -When ever I dial anything, the connection would *drop after 10 to 30 seconds.* -When I enable SIP debug, I get the following: ############################################################# [May 28 16:07:58] VERBOSE[2688] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> INVITE sip:*65@voiptosave.com SIP/2.0 Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Record-Route: <sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0499.3cc1dc7f932835a200ec1f8feafd0895.0 Via: SIP/2.0/UDP 192.168.1.6:29470 ;branch=z9hG4bK-d87543-182127345f1d7b4f-1--d87543-;rport=29470 Max-Forwards: 69 Contact: <sip:110@192.168.1.6:29470> To: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080> From: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 1011w stamp 42237 Content-Length: 457 v=0 o=- 2 2 IN IP4 192.168.1.62 s=CounterPath eyeBeam 1.5 c=IN IP4 192.168.1.62 t=0 0 m=audio 18598 RTP/AVP 100 106 6 0 105 8 18 3 5 101 a=alt:1 1 : KjAnzMl+ lOdYVIyR 192.168.1.6 15880 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:60C3A67CD0E643D8A6945053F94B1D3E a=nortpproxy:yes <-------------> [May 28 16:07:58] VERBOSE[2688] chan_sip.c: --- (15 headers 17 lines) --- [May 28 16:07:58] VERBOSE[2688] chan_sip.c: Sending to 127.0.0.1:5060 (NAT) [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Sending to 127.0.0.1:5060 (NAT) [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Using INVITE request as basis request - Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found peer ' 110_voiptosave.com' for '110_voiptosave.com' from 127.0.0.1:5060 [May 28 16:07:58] VERBOSE[2688][C-00000021] netsock2.c: == Using SIP RTP TOS bits 184 [May 28 16:07:58] VERBOSE[2688][C-00000021] netsock2.c: == Using SIP RTP CoS mark 5 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 100 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 106 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 6 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 0 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 105 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 8 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 18 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 3 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 5 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found RTP audio format 101 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found audio description format SPEEX for ID 100 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found unknown media description format SPEEX-FEC for ID 106 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found unknown media description format SPEEX-FEC for ID 105 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found audio description format G729 for ID 18 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Found audio description format telephone-event for ID 101 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(gsm|ulaw|alaw|adpcm|g729|speex16)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw) [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Peer audio RTP is at port 192.168.1.62:18598 [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: Looking for *65 in voiptosave.com-from-internal (domain voiptosave.com) [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: list_route: hop: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: list_route: hop: <sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> [May 28 16:07:58] VERBOSE[2688][C-00000021] chan_sip.c: <--- Transmitting (NAT) to 127.0.0.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0499.3cc1dc7f932835a200ec1f8feafd0895.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.6:29470 ;branch=z9hG4bK-d87543-182127345f1d7b4f-1--d87543-;rport=29470 Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Record-Route: <sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> From: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e To: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080> Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 2 INVITE Server: Asterisk PBX 11.13.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:*65@127.0.0.1:5080> Content-Length: 0 <------------> [May 28 16:07:58] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:1] Answer("SIP/110_voiptosave.com-0000001a", "") in new stack [May 28 16:07:58] VERBOSE[9110][C-00000021] chan_sip.c: Audio is at 17370 [May 28 16:07:58] VERBOSE[9110][C-00000021] chan_sip.c: Adding codec 100003 (ulaw) to SDP [May 28 16:07:58] VERBOSE[9110][C-00000021] chan_sip.c: Adding codec 100004 (alaw) to SDP [May 28 16:07:58] VERBOSE[9110][C-00000021] chan_sip.c: Adding codec 100002 (gsm) to SDP [May 28 16:07:58] VERBOSE[9110][C-00000021] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP [May 28 16:07:58] VERBOSE[9110][C-00000021] chan_sip.c: <--- Reliably Transmitting (NAT) to 127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0499.3cc1dc7f932835a200ec1f8feafd0895.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.6:29470 ;branch=z9hG4bK-d87543-182127345f1d7b4f-1--d87543-;rport=29470 Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Record-Route: <sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> From: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e To: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080>;tag=as3a404cf6 Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 2 INVITE Server: Asterisk PBX 11.13.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:*65@127.0.0.1:5080> Content-Type: application/sdp Content-Length: 275 v=0 o=root 153043023 153043023 IN IP4 127.0.0.1 s=Asterisk PBX 11.13.0 c=IN IP4 127.0.0.1 t=0 0 m=audio 17370 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> [May 28 16:07:58] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:2] Wait("SIP/110_voiptosave.com-0000001a", "1") in new stack [May 28 16:07:58] VERBOSE[2688] chan_sip.c: Retransmitting #1 (NAT) to 127.0.0.1:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0499.3cc1dc7f932835a200ec1f8feafd0895.0;received=127.0.0.1;rport=5060 Via: SIP/2.0/UDP 192.168.1.6:29470 ;branch=z9hG4bK-d87543-182127345f1d7b4f-1--d87543-;rport=29470 Record-Route: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Record-Route: <sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> From: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e To: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080>;tag=as3a404cf6 Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 2 INVITE Server: Asterisk PBX 11.13.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:*65@127.0.0.1:5080> Content-Type: application/sdp Content-Length: 275 v=0 o=root 153043023 153043023 IN IP4 127.0.0.1 s=Asterisk PBX 11.13.0 c=IN IP4 127.0.0.1 t=0 0 m=audio 17370 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [May 28 16:07:58] VERBOSE[2688] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> ACK sip:*65@127.0.0.1:5080 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0499.1131e459ea9f93c78896b0fdaf99b3be.0 Via: SIP/2.0/UDP 192.168.1.6:29470 ;branch=z9hG4bK-d87543-9f74bc13426dd510-1--d87543-;rport=29470 Max-Forwards: 69 Contact: <sip:110@192.168.1.6:29470> To: "*65"<sip:*65@voiptosave.com>;tag=as3a404cf6 From: "Manny"<sip:110@voiptosave.com>;tag=f133137e Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 2 ACK Proxy-Authorization: Digest username="110",realm="voiptosave.com ",nonce="VWd2ylVndZ5Yo3kaAdjLh/W6zc+CeoE1",uri="sip:*65@voiptosave.com ",response="ed5ad3dd093d74414340aa68dca6a246",algorithm=MD5 User-Agent: eyeBeam release 1011w stamp 42237 Content-Length: 0 <-------------> [May 28 16:07:58] VERBOSE[2688] chan_sip.c: --- (12 headers 0 lines) --- [May 28 16:07:58] VERBOSE[2688] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> ACK sip:*65@127.0.0.1:5080 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1;branch=z9hG4bK0499.1131e459ea9f93c78896b0fdaf99b3be.0 Via: SIP/2.0/UDP 192.168.1.6:29470 ;branch=z9hG4bK-d87543-9f74bc13426dd510-1--d87543-;rport=29470 Max-Forwards: 69 Contact: <sip:110@192.168.1.6:29470> To: "*65"<sip:*65@voiptosave.com>;tag=as3a404cf6 From: "Manny"<sip:110@voiptosave.com>;tag=f133137e Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 2 ACK Proxy-Authorization: Digest username="110",realm="voiptosave.com ",nonce="VWd2ylVndZ5Yo3kaAdjLh/W6zc+CeoE1",uri="sip:*65@voiptosave.com ",response="ed5ad3dd093d74414340aa68dca6a246",algorithm=MD5 User-Agent: eyeBeam release 1011w stamp 42237 Content-Length: 0 <-------------> [May 28 16:07:58] VERBOSE[2688] chan_sip.c: --- (12 headers 0 lines) --- [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:3] Macro("SIP/110_voiptosave.com-0000001a", "voiptosave.com-user-callerid,") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:1] Set("SIP/110_voiptosave.com-0000001a", "EXTUSER=110") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:2] GotoIf("SIP/110_voiptosave.com-0000001a", "0?report") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:3] ExecIf("SIP/110_voiptosave.com-0000001a", "1?Set(REALCALLERIDNUM=110)") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:4] Set("SIP/110_voiptosave.com-0000001a", "EXTUSER=110") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:5] Set("SIP/110_voiptosave.com-0000001a", "EXTUSERCIDNAME=User_110") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:6] GotoIf("SIP/110_voiptosave.com-0000001a", "0?report") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:7] Set("SIP/110_voiptosave.com-0000001a", "EXTUSERCID=110") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:8] Set("SIP/110_voiptosave.com-0000001a", "CALLERID(all)="User_110" <110>") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:9] ExecIf("SIP/110_voiptosave.com-0000001a", "1?Set(CHANNEL(language)=en)") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:10] ExecIf("SIP/110_voiptosave.com-0000001a", "1?Set(CHANNEL(language)=en)") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:11] GotoIf("SIP/110_voiptosave.com-0000001a", "0?continue") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:12] Set("SIP/110_voiptosave.com-0000001a", "__TTL=64") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:13] GotoIf("SIP/110_voiptosave.com-0000001a", "1?continue") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-user-callerid,s,20) [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:20] Set("SIP/110_voiptosave.com-0000001a", "CALLERID(number)=110") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:21] Set("SIP/110_voiptosave.com-0000001a", "CALLERID(name)=User_110") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-user-callerid:22] NoOp("SIP/110_voiptosave.com-0000001a", "Using CallerID "User_110" <110>") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:4] Playback("SIP/110_voiptosave.com-0000001a", "your") in new stack [May 28 16:07:59] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'your.gsm' (language 'en') [May 28 16:08:00] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:5] Playback("SIP/110_voiptosave.com-0000001a", "extension") in new stack [May 28 16:08:00] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'extension.gsm' (language 'en') [May 28 16:08:01] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:6] Playback("SIP/110_voiptosave.com-0000001a", "number") in new stack [May 28 16:08:01] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'number.gsm' (language 'en') [May 28 16:08:02] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:7] Playback("SIP/110_voiptosave.com-0000001a", "is") in new stack [May 28 16:08:02] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'is.gsm' (language 'en') [May 28 16:08:03] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:8] SayDigits("SIP/110_voiptosave.com-0000001a", "110") in new stack [May 28 16:08:03] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'digits/1.gsm' (language 'en') [May 28 16:08:03] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'digits/1.gsm' (language 'en') [May 28 16:08:04] VERBOSE[9110][C-00000021] file.c: -- <SIP/110_voiptosave.com-0000001a> Playing 'digits/0.gsm' (language 'en') [May 28 16:08:05] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:9] Wait("SIP/110_voiptosave.com-0000001a", "2") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [*65@voiptosave.com-from-internal:10] Hangup("SIP/110_voiptosave.com-0000001a", "") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: == Spawn extension (voiptosave.com-from-internal, *65, 10) exited non-zero on 'SIP/110_voiptosave.com-0000001a' [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [h@voiptosave.com-from-internal:1] Macro("SIP/110_voiptosave.com-0000001a", "voiptosave.com-hangupcall") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:1] ExecIf("SIP/110_voiptosave.com-0000001a", "1?Set(CDR(organization_domain)= voiptosave.com)") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:2] GotoIf("SIP/110_voiptosave.com-0000001a", "1?endmixmoncheck") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,7) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:7] NoOp("SIP/110_voiptosave.com-0000001a", "End of MIXMON check") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:8] GotoIf("SIP/110_voiptosave.com-0000001a", "1?nomeetmemon") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,14) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:14] NoOp("SIP/110_voiptosave.com-0000001a", "MEETME_RECORDINGFILE=") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:15] GotoIf("SIP/110_voiptosave.com-0000001a", "1?noautomon") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,23) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:23] NoOp("SIP/110_voiptosave.com-0000001a", "TOUCH_MONITOR_OUTPUT=") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:24] GotoIf("SIP/110_voiptosave.com-0000001a", "1?noautomon2") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,31) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:31] NoOp("SIP/110_voiptosave.com-0000001a", "MONITOR_FILENAME=") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:32] GotoIf("SIP/110_voiptosave.com-0000001a", "1?skiprg") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,35) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:35] GotoIf("SIP/110_voiptosave.com-0000001a", "1?skipblkvm") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,38) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:38] GotoIf("SIP/110_voiptosave.com-0000001a", "1?theend") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Goto (macro-voiptosave.com-hangupcall,s,40) [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: -- Executing [s@macro-voiptosave.com-hangupcall:40] Hangup("SIP/110_voiptosave.com-0000001a", "") in new stack [May 28 16:08:07] VERBOSE[9110][C-00000021] app_macro.c: == Spawn extension (macro-voiptosave.com-hangupcall, s, 40) exited non-zero on 'SIP/110_voiptosave.com-0000001a' in macro 'voiptosave.com-hangupcall' [May 28 16:08:07] VERBOSE[9110][C-00000021] pbx.c: == Spawn extension (voiptosave.com-from-internal, h, 1) exited non-zero on 'SIP/110_voiptosave.com-0000001a' [May 28 16:08:07] VERBOSE[9110][C-00000021] chan_sip.c: Scheduling destruction of SIP dialog 'Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA.' in 6400 ms (Method: ACK) [May 28 16:08:07] VERBOSE[9110][C-00000021] chan_sip.c: set_destination: Parsing <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> for address/port to send to [May 28 16:08:07] VERBOSE[9110][C-00000021] chan_sip.c: set_destination: set destination to 127.0.0.1:5060 [May 28 16:08:07] VERBOSE[9110][C-00000021] chan_sip.c: Reliably Transmitting (NAT) to 127.0.0.1:5060: BYE sip:110@192.168.1.6:29470 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK2355deaa;rport Route: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes>,<sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Max-Forwards: 70 From: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080>;tag=as3a404cf6 To: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 102 BYE User-Agent: Asterisk PBX 11.13.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [May 28 16:08:07] VERBOSE[2688] chan_sip.c: Retransmitting #1 (NAT) to 127.0.0.1:5060: BYE sip:110@192.168.1.6:29470 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK2355deaa;rport Route: <sip:127.0.0.1;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes>,<sip:192.168.1.62;r2=on;lr=on;ftag=f133137e;vsf=BhoZSgsAAW82GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;vst=HxoZShAcA2o2GQYZBBscEhcTS00MAi0xMjcuMC4wLjE6NTA4MA--;nat=yes> Max-Forwards: 70 From: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080>;tag=as3a404cf6 To: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 102 BYE User-Agent: Asterisk PBX 11.13.0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [May 28 16:08:07] VERBOSE[2688] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK2355deaa;rport=5080 Contact: <sip:110@192.168.1.6:29470> To: "Manny"<sip:110_voiptosave.com@127.0.0.1:5080>;tag=f133137e From: "*65"<sip:*65_voiptosave.com@127.0.0.1:5080>;tag=as3a404cf6 Call-ID: Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA. CSeq: 102 BYE User-Agent: eyeBeam release 1011w stamp 42237 Content-Length: 0 <-------------> [May 28 16:08:07] VERBOSE[2688] chan_sip.c: --- (9 headers 0 lines) --- [May 28 16:08:07] VERBOSE[2688][C-00000021] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived [May 28 16:08:07] VERBOSE[2688] chan_sip.c: Really destroying SIP dialog 'Mjk2YjBjZjA2NDYzOTUxZDVmOGZjNTgwYzVmMjg3YjA.' Method: ACK [May 28 16:08:08] VERBOSE[2688] chan_sip.c: Reliably Transmitting (NAT) to 127.0.0.1:5060: OPTIONS sip:110@192.168.1.6:29470;rinstance=3180b40a89f2e3f4 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK0bcd981c;rport Max-Forwards: 70 From: "asterisk" <sip:asterisk@127.0.0.1:5080>;tag=as68358641 To: <sip:110@192.168.1.6:29470;rinstance=3180b40a89f2e3f4> Contact: <sip:asterisk@127.0.0.1:5080> Call-ID: 03d50e985a2f08445ad315c83a591412@127.0.0.1:5080 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 11.13.0 Date: Thu, 28 May 2015 20:08:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [May 28 16:08:08] VERBOSE[2688] chan_sip.c: <--- SIP read from UDP:127.0.0.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK0bcd981c;rport=5080 Contact: <sip:192.168.1.6:29470> To: <sip:110@192.168.1.6:29470;rinstance=3180b40a89f2e3f4>;tag=86756729 From: "asterisk"<sip:asterisk@127.0.0.1:5080>;tag=as68358641 Call-ID: 03d50e985a2f08445ad315c83a591412@127.0.0.1:5080 CSeq: 102 OPTIONS Accept: application/sdp Accept-Language: en Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: eyeBeam release 1011w stamp 42237 Content-Length: 0 <-------------> [May 28 16:08:08] VERBOSE[2688] chan_sip.c: --- (12 headers 0 lines) --- [May 28 16:08:08] VERBOSE[2688] chan_sip.c: Really destroying SIP dialog ' 03d50e985a2f08445ad315c83a591412@127.0.0.1:5080' Method: OPTIONS ################################################################## Please let me know if you need any settings. I am more than happy to provide it. Thank you.

1 1

log the value of "from_uri" and "myself"
by Al S 28 May '15

28 May '15

Hi, I would like to know how to log the value of "from_uri" and "myself" attributes on kamailio 3.3.7 ?None of the following works: xlog("L_INFO","from_uri: $dlg(from_uri)"); $var(x) = from_uri;xlog("L_INFO","$var(x)"); Thanks,Al

2 3

Kamailio not receiving BYE request
by Ali Taher 28 May '15

28 May '15

Hello, Kamailio is not receiving the BYE request when the call is ended by any of call parties. Noting that when the call is connected, a direct RTP session is established between the user and the gateway , and kamailio doesn't receive since then any message related to this call. Is there any trick to do in order to let kamailio gets the BYE request? Attached is the used configuration. Appreciate your help. Regards, Ali

3 11

Regarding presence conference
by Surendra Pullaiah 28 May '15

28 May '15

Hello Daniel, I am implementing conference bridge using kamalio and RTP porxy without media mixing. Since for now we are ok with this. My question is as per the flow of RFC 4579, UA may send SUBSCRIBE to the conference URI with event as conference. In kamalio modules presence_conference where they mentioned such way that it will support the same what I am looking for. But I simulated using sipP by sending subscriber with conference event. I am not seeing any notify XML which will have the conference users list. One more doubt where it will pick the conference users, since I am not seeing any specific data base table for the same. Can you please clarify what exactly for presence_conference module. Note: I have loaded three modules presence, presence_xml, presence_conference, in route config I am calling handle_subscribe when SUBCRIBE method receives. Regards Surendra India: +91-8124625001

1 0

Kamailio drouting failure not working
by Ali Taher 28 May '15

28 May '15

Dears, I'm trying drouting module in kamailio , it's working good when first gateway in the list is up , yet when first gateway is unrecheable , it's not rerouted to second destination, meaning that fail over is not working. Kindly find attached wireshark trace I'm getting . Below is db entries I'm using : dr_gateways: Dr_gw_lists Dr_groups: Dr_rules: Below is routing block I'm using : request_route{ if (is_method("INVITE")) { if (!do_routing("0")) { send_reply("503", "Test No Rules matching the URI"); exit; } route(10); exit; } } route[10] { if (!do_routing("0")) { xlog("do_routing: No rules matching the URI\n"); send_reply("503","No rules matching the URI"); exit; } if (is_method("INVITE")) { t_on_failure("4"); } route(1); } failure_route[4] { xlog("DEBUG: DROUTING failure route active\n"); if (t_check_status("[34][0-9][0-9]")) { exit; } if (use_next_gw()) { t_relay(); exit; } else { t_reply ("503", "Service not available"); exit; } } route[1] { # send it out now; use stateful forwarding as it works # reliably even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } Appreciating your prompt reply and assistance. Ali Taher Technical Support Engineer T&F M W A +961-5-457437 Ext 215 +961-70-010600 <http://www.vanrise.com/> http://www.vanrise.com <https://maps.google.com/maps/ms?msa=0&msid=208243362929850217065.0004cbc8d6 12c5f9b4237&ie=UTF8&t=h&vpsrc=0&z=19&iwloc=0004cbc8e4652b4edf71cC:\Users\Mar ie\Documents\My%20Received%20Files> Vanrise Building, Facing Galerie Semaan Hazmiyeh Highway - Beirut, Lebanon Join us at <http://www.facebook.com/pages/Vanrise-Solutions-Offshore/128309743915533> <http://www.linkedin.com/company/vanrise-solutions> <http://twitter.com/VanriseSolution>

3 22

kamailio crash periodically in timer handler
by Péter Barabás 27 May '15

27 May '15

Hi, we have a serious problem with Kamailio v4.2.3, it crashes after some days. We have checked the core dump and the gdb output is the following for bt and full backtrace: #0 0x0000000000602291 in timer_list_expire (t=1432559962, h=0x7f64eb5b6bd0, slow_l=0x7f64eb5ba328, slow_mark=43873) at timer.c:877 #1 0x0000000000602746 in timer_handler () at timer.c:953 #2 0x0000000000602bb9 in timer_main () at timer.c:992 #3 0x00000000004a833c in main_loop () at main.c:1700 #4 0x00000000004ad857 in main (argc=13, argv=0x7fff96697d38) at main.c:2578 #0 0x0000000000602291 in timer_list_expire (t=1432559962, h=0x7f64eb5b6bd0, slow_l=0x7f64eb5ba328, slow_mark=43873) at timer.c:877 tl = 0x7f64ebc3a908 ret = 0 #1 0x0000000000602746 in timer_handler () at timer.c:953 saved_ticks = 1432559962 run_slow_timer = 0 i = 865 __FUNCTION__ = "timer_handler" #2 0x0000000000602bb9 in timer_main () at timer.c:992 No locals. #3 0x00000000004a833c in main_loop () at main.c:1700 i = 4 pid = 0 si = 0x0 si_desc = "udp receiver child=3 sock=127.0.0.1:5060\000\375\362\250/\313\345\v\004\000\000\000\000\000\000\000\004\257\263=\000\000\000\000\200TA\000\000\000\000\000\060}i\226\377\177", '\000' <repeats 19 times>, "{i\226\377\177\000\000\376\366R\000\000\000\000\000\377\377\377\377\377\177\000\000@\350\250\000\000\000\000" nrprocs = 4 __FUNCTION__ = "main_loop" #4 0x00000000004ad857 in main (argc=13, argv=0x7fff96697d38) at main.c:2578 cfg_stream = 0x1b94010 c = -1 r = 0 tmp = 0x7fff96699b17 "" tmp_len = 0 port = 1 proto = 32767 options = 0x703e08 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:" ret = -1 seed = 2591880768 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x0 p = 0x7fff96697c50 "" __FUNCTION__ = "main" In the logs it can be seen: Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4635]: CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 21 Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4612]: ALERT: <core> [main.c:784]: handle_sigs(): child process 4623 exited by a signal 11 Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4612]: ALERT: <core> [main.c:787]: handle_sigs(): core was generated Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4612]: INFO: <core> [main.c:799]: handle_sigs(): terminating due to SIGCHLD Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4631]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4628]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4634]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4615]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4617]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4626]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4625]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4624]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4632]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4622]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4621]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4620]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4619]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4618]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4616]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4635]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4614]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4629]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4627]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4633]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4630]: INFO: <core> [main.c:850]: sig_usr(): signal 15 received Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4612]: INFO: <core> [mem/f_malloc.c:599]: fm_free(): freeing a free fragment (0x7f64eca19950/0x7f64eca19988) - ignore Apr 23 08:07:31 ctsip1 /usr/local/sbin/kamailio[4612]: INFO: <core> [mem/f_malloc.c:599]: fm_free(): freeing a free fragment (0x7f64ed587f20/0x7f64ed587f58) - ignore Kamailio runs on Ubuntu trusty. The interesting thing that we run Kamailio 4.2.3 in an other environment (on Debian Squeeze) and it works perfect, no crashes. The main difference is that in Debian environment we have less number of users and the firewall is not so strength. But the configuration and the source is the same for both. Please help us to solve this problem. Thanks. Dr. Péter Barabás Director of Software Development Arenim Technologies AB. Mob: +36 70 314 56 57 Tel: +36 1 8 555 602 E-mail: peter.barabas(a)arenim.com<mailto:peter.barabas@arenim.com> Postal: Sweden - 11435 Stockholm, Stureplan 4c, 4th. Floor [Arenim_logo]<http://www.arenim.com/> [CryptTalk_Logo]<http://www.crypttalk.com/>

2 11

Kamailio with public ip, users behind nat
by MtK - SmartMtK 27 May '15

27 May '15

Hi, I'm trying to install Kamailio on a server with a public IP, but my users are mostly at their homes (a.k.a behind NAT). I understood I need to install RTPproxy, but all tutorials out there are for servers behind nat, and they talk about the server's private IP, which does not exists in this case. how would I configure RTPproxy and Kamailio to allow users to subscribe to the server and have audio calls between them? MtK

3 2

dbcassandra schema files
by jay binks 26 May '15

26 May '15

Hi all, Im wondering if anyone can share their schema files for dbcassandra. Ive managed to construct a bunch based on what ive found in git, wiki all over the place, I just want to compare my ones, with some others. specifically the location table. -- Sincerely Jay

2 1

Development open again for master branch
by Daniel-Constantin Mierla 26 May '15

26 May '15

Hello, as branch 4.3 was created to host the upcoming release series 4.3, new features can be now pushed to master branch. They will be part of the next future release, likely to be numbered 4.4. Any fixes that affect existing code in branches 4.3 or older version have to be backported - push first to master and then cherry pick -- some guidelines exemplified with branch 3.2 are available at: - https://www.kamailio.org/wiki/devel/backporting-to-3.2.x As usual, the next future release will be out as stable version in like 8-10 months. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

1 0

Branch 4.3 created
by Daniel-Constantin Mierla 26 May '15

26 May '15

Hello, the GIT branch 4.3 was created, it will host the release series 4.3.x. To get this branch from GIT, you can use: git clone https://github.com/kamailio/kamailio.git kamailio cd kamailio git checkout -b 4.3 origin/4.3 Notes about installing Kamailio from this branch are available at: - http://www.kamailio.org/wiki/install/4.3.x/git Hopefully in about two weeks or so the full release of 4.3.0 will be out. >From now on, any corresponding fix has to be pushed first to master branch and then cherry-picked to branch 4.3. No new features can get in branch 4.3. Enhancements to documentation or helping tools are still allowed. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

1 0

Using $avp in evapi:message-received
by DanB 25 May '15

25 May '15

Hey Guys, Was wondering what would be the recommended way to share information between $evapi(msg) received and transaction processors? I have tried $avp() but they do not appear to be available in evapi:message-received route. Trying to avoid here $sht() so I don't need to do extra management to destroy them. Is there any way to "activate" the $avp or $dlg_var based on dialog index, label maybe? Thanks in advance for any kind of tip! DanB //

3 7

info
by Walter Josef 25 May '15

25 May '15

Hi, Do you have any phonenumber for info about your iptel voip phone service ?

2 1

Creating branch 4.3
by Daniel-Constantin Mierla 25 May '15

25 May '15

Hello, I am planning to branch the code for upcoming major release on next Monday, May 25. Testing went fine so far in my side and I don't expect new consistent changes. Any fixes afterwards will be backported as usual to a stable branch. From that moment, master branch will be open again for new features. If someone else has different proposal, let's discuss it on sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

1 1

Re: [SR-Users] Getting 404 From Kamailio for PRACK
by DanB 24 May '15

24 May '15

Hi Austin, Kamailio will not reject by himself (nor handle) PRACK (or any other methods) unless you tell him so. The logic for handling it should be in your .cfg file. I suggest adding xlog statements through your config script and see where that is rejected. 404 would tell me you are trying to locate your endpoint and send the PRACK there but the endpoint is not registered. DanB

1 0

Re: [SR-Users] Increase Simultaneous Registration
by DanB 24 May '15

24 May '15

Hi Austin, Kamailio should be pretty fast on these things. Here are things which you can check: * Increase number of TCP workers/make sure you are using async TCP. * Use in-memory store for your registrations, not DB ones (at least dump async your contacts in DB). * Make sure you allow enough files to be opened in case of TCP connections (I suspect this is your issue there). Regarding performance tests, not sure if you saw already these links: http://www.kamailio.org/docs/openser-performance-tests/ https://sip-router.org/wiki/performance/v3.0-capacity Hope these help! DanB

1 0

disabling Server header in kamailio
by NITESH BANSAL 22 May '15

22 May '15

Hello, I noticed that Kamailio adds a Server header in the replies generated by it, is there anyway to disable it? Thanks in advance for your help, Nitesh

2 1

Find if client is connected.
by Dmytro Bogovych 22 May '15

22 May '15

Greetings. I want to modify INVITE answers processing in this way: 1) receive INVITE answer 2) check if recipient is registered but has no TCP/TLS connection now 3) suspend answer for 30 seconds (for example) 4) resume answer if REGISTER comes from lost client in these 30 seconds 5) terminate INVITE session if 30 seconds are expired What I do not understand - is it possible to find if client is connected now to kamailio? I talk about TCP/TLS connection only - not about registration record. Thank you!

3 7

issues with domain names in the record route
by NITESH BANSAL 22 May '15

22 May '15

Hello, I am trying to play with domain names in the Record-Route header. This is for a case when kamailio is doing the translation between UDP and TLS, so it is doing double record routing. I want to use the ip addresses on inbound UDP leg and domain names on the outbound TLS leg in record-route headers. I'm doing this by calling the following function -- record_route_preset("xyz.com;transport=tls", "1.1.1.1"); ---- This does the trick of adding a Record-Route in initial TLS invite on the outbound leg: ----------------- Record-Route: <sip:xyz.com;transport=tls;lr=on> Record-Route: <sip:1.1.1.1;lr=on> ---------------- My problems start when the calling party sends a ACK to kamailio over UDP with the Route header Route: <sip:1.1.1.1;lr=on>,<sip:xyz.com;transport=tls;lr=on> I call loose_route() function in config file and as per my expectation, Kamailio should see that both these Route headers belong to Kamailio and it should use Request URI for forwarding the ACK, but unfortunately, Kamailio thinks that xyz.com is some other entity and tries to forward the ACK to xyz.com doing a DNS query. What can i do to avoid this problem, i need kamailio to recognize that "xyz.com" is kamailio as i have a defined an alias in the config file too as follows: alias=xyz.com:5060 Thanks for your help in advance, Nitesh

3 3

rtpproxy 2.0
by Klaus Darilion 22 May '15

22 May '15

Hi! I just found out that there is a new rtpproxy release: http://www.rtpproxy.org/post/v2release/ Has anybody tested it and want to share some experiences? Or have people turned to rtpengine meanwhile? (I am still using rtpproxy 1.2 as I do not need new features). regards Klaus

3 2

Re: [SR-Users] Kamailio proxy for far-end nat traversal
by rahul.ultimate 21 May '15

21 May '15

Hi Daniel I have rtpproxy to perform media relaying. The case where it just timeouts: 1. Client A behind NAT (wifi) calling Client B on 3G Sent from Samsung Mobile -------- Original message -------- From: Daniel-Constantin Mierla <miconda(a)gmail.com> Date:05/05/2015 12:52 (GMT+05:30) To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.sip-router.org> Subject: Re: [SR-Users] Kamailio proxy for far-end nat traversal Hello, do you have rtpproxy or rtpengine for relaying rtp packets? Kamailio is routing only sip packets, you need the rtp relay application to help with media streams. Cheers, Daniel On 01/05/15 13:15, rahul.ultimate wrote: Hello I need a small guidance on creating a light weight proxy which only forwards the msgs to my sip server and also does supports symmetrical nated clients. The way I have created the configuration is a slight modification of : https://github.com/xlab1/sipfe_kamailio/blob/master/kamailio.cfg Problem is unless the sip clients use STUN media packets are not routed. And sometimes even the signalling does not pass through. Should i not use fix_nated_register and switch to fix _ nated_contact always ? Because in all those cases where signalling does not pass through, I see RTO. My best guess is kamailio trying to communicate with private ip. Anything which I can try that you could suggest from the top of yourmind woyld be muc appreciated. Thanks Rahul Sent from Samsung Mobile _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

4 5

XLOG Problem
by Nahum Nir 21 May '15

21 May '15

Hi All, I am using Kamailio 4.0.3 and can't logs. .cfg: #!define WITH_DEBUG ### LOG Levels: 3=DBG, 2=INFO, 1=NOTICE, 0=WARN, -1=ERR #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 When the server starts there outputs about xlog: 0(20266) DEBUG: <core> [cfg.y:1791]: yyparse(): loading module xlog.so 0(20266) DEBUG: <core> [sr_module.c:499]: load_module(): load_module: module file not found </usr/local/lib/kamailio/modules_k/xlog.so> 0(20266) DEBUG: <core> [sr_module.c:522]: load_module(): load_module: module file not found </usr/local/lib/kamailio/modules_k/xlog/xlog.so> 0(20266) DEBUG: <core> [sr_module.c:564]: load_module(): load_module: trying to load </usr/lib64/kamailio/modules/xlog.so> 0(20266) DEBUG: <core> [sr_module.c:343]: register_module(): register PV from: xlog 0(20266) DEBUG: <core> [sr_module.c:674]: find_mod_export_record(): find_export_record: found <xlog> in module xlog [/usr/lib64/kamailio/modules/xlog.so] 0(20266) DEBUG: <core> [sr_module.c:674]: find_mod_export_record(): find_export_record: found <xlog> in module xlog [/usr/lib64/kamailio/modules/xlog.so] 0(20266) DEBUG: <core> [sr_module.c:674]: find_mod_export_record(): find_export_record: found <xlog> in module xlog [/usr/lib64/kamailio/modules/xlog.so] 0(20266) DEBUG: <core> [sr_module.c:674]: find_mod_export_record(): find_export_record: found <xlog> in module xlog [/usr/lib64/kamailio/modules/xlog.so] 0(20266) DEBUG: <core> [sr_module.c:674]: find_mod_export_record(): find_export_record: found <xlog> in module xlog [/usr/lib64/kamailio/modules/xlog.so] 0(20266) DEBUG: <core> [sr_module.c:674]: find_mod_export_record(): find_export_record: found <xlog> in module xlog [/usr/lib64/kamailio/modules/xlog.so] 0(20266) DEBUG: <core> [sr_module.c:674]: find_mod_export_record(): find_export_record: found <xdbg> in module xlog [/usr/lib64/kamailio/modules/xlog.so] 0(20266) DEBUG: <core> [sr_module.c:674]: find_mod_export_record(): find_export_record: found <xdbg> in module xlog [/usr/lib64/kamailio/modules/xlog.so] 0(20268) DEBUG: <core> [sr_module.c:935]: init_mod(): DEBUG: init_mod: xlog 0(20268) DEBUG: <core> [route.c:928]: fix_actions(): fixing xlog() 0(20268) DEBUG: <core> [route.c:928]: fix_actions(): fixing xlog() 0(20268) DEBUG: <core> [route.c:928]: fix_actions(): fixing xlog() 0(20268) DEBUG: <core> [route.c:928]: fix_actions(): fixing xlog() 0(20268) DEBUG: <core> [route.c:928]: fix_actions(): fixing xlog() 0(20268) DEBUG: <core> [route.c:928]: fix_actions(): fixing xlog() At the beginning of the Main routing block I put: log("~~~~~~~~~~~~route~~~~~~~~~~~~~~~~~~~~\n"); xdbg("~~~~~~~~~ xdbg - route~~~~~~~~~~~~~~~~~~~~\n"); xlog("L_ALERT","~~~~~~~~~ xlog - route~~~~~~~~~~~~~~~~~~~~\n"); But nothing is logging when I send INVITEs Any help? Thanks, Nir

4 3

Kamailio 4.2.4 segfault and crash
by Federico San Martín 21 May '15

21 May '15

Hi all, I just experienced a sudden crash from one of my Kamailio servers. I'm using v4.2.4 built from git repository (branch remotes/origin/4.2). The error logs show this at the time of the crash (I'm using debug=1 on config file) May 15 15:59:53 KAMAILIO-ERS1 kamailio[21162]: CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 19 May 15 15:59:53 KAMAILIO-ERS1 kamailio[21141]: ALERT: <core> [main.c:784]: handle_sigs(): child process 21149 exited by a signal 11 May 15 15:59:53 KAMAILIO-ERS1 kamailio[21141]: ALERT: <core> [main.c:787]: handle_sigs(): core was not generated May 15 15:59:53 KAMAILIO-ERS1 kernel: [2156637.186585] kamailio[21149]: segfault at 20b6 ip 00000000000020b6 sp 00007ffeb0c2bc88 error 14 in kamailio[400000+3c3000] May 15 15:59:53 KAMAILIO-ERS1 kernel: [2156637.214933] kamailio[21141]: segfault at 2290 ip 0000000000002290 sp 00007ffeb0c2c4d8 error 14 in kamailio[400000+3c3000] The modules I'm using are: loadmodule "mi_fifo.so" loadmodule "db_mysql.so" loadmodule "sqlops.so" loadmodule "tm.so" loadmodule "sl.so" loadmodule "pv.so" loadmodule "xlog.so" loadmodule "maxfwd.so" loadmodule "siputils.so" loadmodule "sanity.so" loadmodule "textops.so" loadmodule "rr.so" loadmodule "statsd.so" Have you noticed this error before? This server has a very simple config, for every invite received, it queries a MySQL database to replace the called number from 3 digits to 8-10 digits (for example 135 is translated to 1152751135) and sends back a 300 Redirect with the new number. Another thing to notice is that this server was working at 3-4% CPU load and when the issue happen it spiked to 53%... Let me know I can provide any other info that could be useful. Thanks! Federico Federico San Martín Ingeniería de Telefonía TeleCentro S.A. Cnel. Apolinario Figueroa 254 Piso 4to CABA - C1414EDF Fijo : +54 11 3977 10 45 Móvil : +54 9 11 5824 3240 e-mail : fsanmartin(a)telecentro.net.ar

2 3

Clear To-tag on '181 Call Is Being Forwarded'
by John Murray 21 May '15

21 May '15

Hello, I am forwarding an incoming call to a destination and on time-out forward to another and them another. I need to send a '181 Call Is Being Forwarded' to my SBC to reset the early media stream when forwarding to the next destiation which I can do with sl_send_reply("181","Call Is Being Forwarded"). However the SBC vendor requires that the To-tag in the 181 is cleared for this to work. How can I clear the To-tag from this 181 response? Thanks John

2 1

setup dbtext issues
by jaflong jaflong 21 May '15

21 May '15

I need some instructions on how to setup dbtext properly I generated dbtext files through "./kamdbctl create" I then add user through "./kamctrl add user1 user1" It seems to have added the user because when I do the show './kamctl show user1' command it displays [1, ' user1', '10.1.1.1', ' user1', '', 'ee2820706921ce44a3f760e03ff19a22', '9e2c9e7df27f474d2b995c7e94ab32f2', ''] However viewing the subscriber file /usr/local/etc/kamailio/dbtext/subscriber id(int,auto) username(string) domain(string) password(string) email_address(string,null) ha1(string,null) ha1b(string,null) rpid(string,null) There is no entry for 'user1'. Where is it storing the user data and why is it not in the subscriber file?. user1 is not recognised when trying to register 3(20432) DEBUG: db_text [dbt_api.c:80]: dbt_get_columns(): allocate 16 bytes for RES_NAMES[0] at 0x7f9e0dd776c0 3(20432) DEBUG: auth_db [authorize.c:196]: get_ha1(): no result for user 'user1@' Regards

2 1

Debian package kamailio-tls-modules potential bug ?
by jay binks 20 May '15

20 May '15

I just found something odd in the Debian package for kamailio-als-modules and I'm looking for someone to verify it for me. I verified that I'm using Kamailio 4.2 # Kamailio -v version: kamailio 4.2.0 (x86_64/linux) 36e695 the docs http://kamailio.org/docs/modules/4.2.x/modules/tls.html#tls.p.tls_method show that I should be able to use "*TLSv1+" however Kamailio crashes on startup with * [tls_mod.c:287]: mod_init(): Invalid tls_method parameter value I have verified that TLSv1.1 and TLSv1.2 both work, but I can not use TLSv1.1+ or TLSv1+ I also tried TLSv1.0+ ( I found it somewhere on the web ) but this doesn't work either. I used strings against tls.so to see what was available, and this is what I found. strings ~/tls.so | grep TLSv1 TLSv1_client_method TLSv1_server_method TLSv1_method TLSv1_1_client_method TLSv1_1_server_method TLSv1_1_method TLSv1_2_client_method TLSv1_2_server_method TLSv1_2_method TLSv1 TLS method used (TLSv1, SSLv3, SSLv2, SSLv23) TLSv1.1 TLSv1.2 Any idea why this is differs from the docs, or what I'm doing wrong ? -- Sincerely Jay

2 1

Kamailio v4.2.5 Released
by Daniel-Constantin Mierla 19 May '15

19 May '15

Hello, Kamailio SIP Server v4.2.5 stable release is out. This is a maintenance release of the latest stable branch, 4.2, that includes fixes since release of v4.2.5. There is no change to database schema or configuration language structure that you have to do on installations of v4.2.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.2.5. For more details about version 4.2.5 (including links and guidelines to download the tarball or from GIT repository), visit: * http://www.kamailio.org/w/2015/05/kamailio-v4-2-5-released/ RPM, Debian/Ubuntu packages will be available soon as well. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

2 1

log_prefix not working in reply_route
by Klaus Darilion 19 May '15

19 May '15

Hi! Should log_prefix also work for xlog in reply routes? In my case it doesn't (Kamailio 4.2.2) Thanks Klaus

2 3

rtpproxy-ng and late SDP
by Sebastian Damm 19 May '15

19 May '15

Hi, I'm working on a setup, where I have to support late SDP. It is working right now, but I have to use an explicit rtpproxy_offer() when processing the reply to an INVITE without SDP. Is there a way to have rtpproxy_manage() handle those calls automatically? I was thinking of setting a flag in the request which is used by the rtpproxy_manage() function later. Best Regards, Sebastian

3 3

SIP Packet identifier
by jay binks 19 May '15

19 May '15

So im logging lots of information to Syslog to assist our helpdesk in diagnosing customer issues. Im seeing LOTS of user agents that re-use the sip Call-ID ( especially in registers ). What im wanting to be able to do is link a log message ( that I log at some point in my dialplan ) to a specific SIP packet ( viewed from voipmonitor, HOMER, wireshark ). I cant REALLY see anything in the SIP to uniquely identify a packet, however if kamailio provided the UDP packet checksum in a pseudo variable then we could combine that with the sip call-id to hopefully do what im after. does anyone else have any ideas about how do what im after ? -- Sincerely Jay

4 3

Planning to release Kamailio v4.2.5
by Daniel-Constantin Mierla 19 May '15

19 May '15

Hello, I am considering to release a new maintenance version from latest stable branch - to be v4.2.5 - next week, most likely on Tuesday (May 19) or Wednesday (May 20). If there is anything important to get in and not discussed yet, bring the topic on development mailing list. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World Conference, May 27-29, 2015 Berlin, Germany - http://www.kamailioworld.com

1 1

NO VOICE AFTER MSILO
by sscc 19 May '15

19 May '15

i have configured msilo module successful but there isn't any voice with msilo. i debug and compare the call flow with and without msilo. with msilo in call flow it didn't follow to relay and consequently didnt activate rtpproxy befor the call is answered. due to which during call there isn't any voice. please help -- View this message in context: http://sip-router.1086192.n5.nabble.com/NO-VOICE-AFTER-MSILO-tp137430.html Sent from the Users mailing list archive at Nabble.com.

3 8

sems-rtpengine-webrtc client audio problem
by Juha Heinanen 19 May '15

19 May '15

i have not been able to figure out, why i can't hear any audio when i call from webrtc client (using UDP/TLS/RTP/SAVPF profile) to sems echo server (using RTP/AVP profile): 1) webrtc client - UDP/TLS/RTP/SAVPF - rtpengine - RTP/AVP - sems echo server if i call from the same webrtc client to baresip, audio works fine: 2) webrtc client - UDP/TLS/RTP/SAVPF - rtpengine - RTP/AVP - baresip also, if i call from baresip using UDP/TLS/RTP/SAVPF profile to sems, audio works fine: 3) basesip - UDP/TLS/RTP/SAVPF - rtpengine - RTP/AVP - sems echo server below is level 7 debug from the problem case (1). stats at the bottom tell that audio packets were flowing both ways fine. what does not seem correct is that the webrtc client is constantly sending "Successful STUN binding requests" to rtpengine. any advise for further debugging this problem would be highly appreciated. -- juha ------------------------------------------------------------------------------ May 18 11:47:34 box /usr/bin/sip-proxy[18299]: INFO: ===== rtpengine_offer(ICE=force replace-session-connection replace-origin via-branch=1 rtcp-mux-demux RTP/AVP) May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Received command 'offer' from 127.0.0.1:43378 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Dump for 'offer' from 127.0.0.1:43378: { "sdp": "v=0#015#012o=- 4880355588178147449 2 IN IP4 127.0.0.1#015#012s=-#015#012t=0 0#015#012a=group:BUNDLE audio#015#012a=msid-semantic: WMS k2qrqNs5yzQcMjup0H0E85yNYAP5aA0w4SeM#015#012m=audio 35551 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 126#015#012c=IN IP4 93.106.207.11#015#012a=rtcp:35551 IN IP4 93.106.207.11#015#012a=candidate:1603072777 1 udp 2122260223 192.168.43.146 35551 typ host generation 0#015#012a=candidate:1603072777 2 udp 2122260223 192.168.43.146 35551 typ host generation 0#015#012a=candidate:2861 ... May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] ... 99801 1 tcp 1518280447 192.168.43.146 0 typ host tcptype active generation 0#015#012a=candidate:286199801 2 tcp 1518280447 192.168.43.146 0 typ host tcptype active generation 0#015#012a=candidate:2466368666 1 udp 1686052607 93.106.207.11 35551 typ srflx raddr 192.168.43.146 rport 35551 generation 0#015#012a=candidate:2466368666 2 udp 1686052607 93.106.207.11 35551 typ srflx raddr 192.168.43.146 rport 35551 generation 0#015#012a=ice-ufrag:U+gG6hyUzfrt859a#015#012a=ice-pwd:FQwGMInhUjWl8bbYIbSJS1pz#015#012a=fingerprint:sha-256 09:D6: ... May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] ... AA:5B:F2:9E:D6:A7:CE:69:D3:76:67:EF:36:A6:F4:87:94:B7:37:82:AF:17:9E:4B:E8:B7:F4:F1:8C:A0#015#012a=setup:actpass#015#012a=mid:audio#015#012a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level#015#012a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time#015#012a=sendrec… opus/48000/2#015#012a=fmtp:111 minptime=10; useinbandfec=1#015#012a=rtpmap:103 ISAC/16000#015#012a=rtpmap:104 ISAC/32000#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:106 CN/32000#015#012a=rtpmap:105 CN/16000#015 ... May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] ... #012a=rtpmap:13 CN/8000#015#012a=rtpmap:126 telephone-event/8000#015#012a=maxptime:60#015#012a=ssrc:2202238380 cname:zcLA/sUaZsuYHC73#015#012a=ssrc:2202238380 msid:k2qrqNs5yzQcMjup0H0E85yNYAP5aA0w4SeM 2053b0c2-35f2-48f8-ae79-419a078169f4#015#012a=ssrc:2202238380 mslabel:k2qrqNs5yzQcMjup0H0E85yNYAP5aA0w4SeM#015#012a=ssrc:2202238380 label:2053b0c2-35f2-48f8-ae79-419a078169f4#015#012", "ICE": "force", "replace": [ "session-connection", "origin" ], "transport-protocol": "RTP/AVP", "rtcp-mux": [ "demux" ], "call-id": "e91aj9gq39qr07b6dn6u", "vi ... May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] ... a-branch": "z9hG4bK1294663", "received-from": [ "IP4", "<sip-proxy-rtpengine-ip>" ], "from-tag": "0m9av72lp5", "command": "offer" } May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Creating new call May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Opened ports 8012..8013 for media relay May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Opened ports 8028..8029 for media relay May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Learning new ICE candidate 1603072777:1 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Created candidate pair MmPVRTRG3mIW6Esg:1603072777:1 between <sip-proxy-rtpengine-ip> and 192.168.43.146:35551, type host May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Learning new ICE candidate 1603072777:2 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Learning new ICE candidate 2466368666:1 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Created candidate pair MmPVRTRG3mIW6Esg:2466368666:1 between <sip-proxy-rtpengine-ip> and 93.106.207.11:35551, type srflx May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Learning new ICE candidate 2466368666:2 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Sending ICE/STUN request for candidate pair MmPVRTRG3mIW6Esg:1603072777:1 from <sip-proxy-rtpengine-ip> to 192.168.43.146:35551 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Replying to 'offer' from 127.0.0.1:43378 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Response dump for 'offer' to 127.0.0.1:43378: { "sdp": "v=0#015#012o=- 4880355588178147449 2 IN IP4 <sip-proxy-rtpengine-ip>#015#012s=-#015#012t=0 0#015#012a=msid-semantic: WMS k2qrqNs5yzQcMjup0H0E85yNYAP5aA0w4SeM#015#012m=audio 8012 RTP/AVP 111 103 104 9 0 8 106 105 13 126#015#012c=IN IP4 <sip-proxy-rtpengine-ip>#015#012a=rtpmap:111 opus/48000/2#015#012a=fmtp:111 minptime=10; useinbandfec=1#015#012a=rtpmap:103 ISAC/16000#015#012a=rtpmap:104 ISAC/32000#015#012a=rtpmap:9 G722/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:106 CN/32000#015#012a=rtpmap:105 CN/16000#015#012a=rtpmap:13 ... May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] ... CN/8000#015#012a=rtpmap:126 telephone-event/8000#015#012a=maxptime:60#015#012a=ssrc:2202238380 cname:zcLA/sUaZsuYHC73#015#012a=ssrc:2202238380 msid:k2qrqNs5yzQcMjup0H0E85yNYAP5aA0w4SeM 2053b0c2-35f2-48f8-ae79-419a078169f4#015#012a=ssrc:2202238380 mslabel:k2qrqNs5yzQcMjup0H0E85yNYAP5aA0w4SeM#015#012a=ssrc:2202238380 label:2053b0c2-35f2-48f8-ae79-419a078169f4#015#012a=sendrecv#015#012a=rtcp:8013#015#012a=ice-ufrag:aEaiKI4d#015#012a=ice-pwd:S0ET0fRwnGLZ6lYSdXuAfvCo9K#015#012a=candidate:MmPVRTRG3mIW6Esg 1 UDP 2130706431 <sip-proxy-rtpengine-ip> 8012 typ host#015#012a=candidate:MmP ... May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] ... VRTRG3mIW6Esg 2 UDP 2130706430 <sip-proxy-rtpengine-ip> 8013 typ host#015#012", "result": "ok" } May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u/0m9av72lp5/1] Sending ICE/STUN request for candidate pair MmPVRTRG3mIW6Esg:2466368666:1 from <sip-proxy-rtpengine-ip> to 93.106.207.11:35551 May 18 11:47:34 box /usr/bin/sip-proxy[18242]: INFO: ===== rtpengine_answer(ICE=force via-branch=2 trust-address) May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Received command 'answer' from 127.0.0.1:49396 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Dump for 'answer' from 127.0.0.1:49396: { "sdp": "v=0#015#012o=sems 1244512444 1372754 IN IP4 <sip-proxy-rtpengine-ip>#015#012s=sems#015#012c=IN IP4 <sip-proxy-rtpengine-ip>#015#012t=0 0#015#012m=audio 8064 RTP/AVP 0 8 126#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:126 telephone-event/8000#015#012a=sendrecv#015#012a=direction:both#015#012", "ICE": "force", "flags": [ "trust-address" ], "call-id": "e91aj9gq39qr07b6dn6u", "via-branch": "z9hG4bK1294663", "received-from": [ "IP4", "127.0.0.1" ], "from-tag": "0m9av72lp5", "to-tag": "627A71DC-5559A726000163C1 ... May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] ... -AA759700", "command": "answer" } May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Shutting down ICE agent (nothing to do) May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Replying to 'answer' from 127.0.0.1:49396 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Response dump for 'answer' to 127.0.0.1:49396: { "sdp": "v=0#015#012o=sems 1244512444 1372754 IN IP4 <sip-proxy-rtpengine-ip>#015#012s=sems#015#012c=IN IP4 <sip-proxy-rtpengine-ip>#015#012t=0 0#015#012m=audio 8028 UDP/TLS/RTP/SAVPF 0 8 126#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:126 telephone-event/8000#015#012a=direction:both#015#012a=sendrecv#015#012a=rtcp:8028#015#012a=rtcp-mux#015#012a=setup:active#015#012a=fingerprint:sha-1 B0:27:8A:76:DE:C4:2A:18:7D:C9:32:F6:78:79:5E:7B:1D:7C:DB:68#015#012a=ice-ufrag:joFSF896#015#012a=ice-pwd:AMtcYr30DYw7PhCsz1LU1Eoi12#015#012a=candidate:MmPVRTRG ... May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] ... 3mIW6Esg 1 UDP 2130706431 <sip-proxy-rtpengine-ip> 8028 typ host#015#012", "result": "ok" } May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8012] SRTP output wanted, but no crypto suite was negotiated May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u/0m9av72lp5/1] Sending ICE/STUN request for candidate pair MmPVRTRG3mIW6Esg:1603072777:1 from <sip-proxy-rtpengine-ip> to 192.168.43.146:35551 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u/0m9av72lp5/1] Sending ICE/STUN request for candidate pair MmPVRTRG3mIW6Esg:2466368666:1 from <sip-proxy-rtpengine-ip> to 93.106.207.11:35551 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Triggering check for MmPVRTRG3mIW6Esg:2466368666:1 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] ICE pair MmPVRTRG3mIW6Esg:2466368666:1 has been nominated by peer May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] ICE not completed yet May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u/0m9av72lp5/1] Sending ICE/STUN request for candidate pair MmPVRTRG3mIW6Esg:2466368666:1 from <sip-proxy-rtpengine-ip> to 93.106.207.11:35551 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Triggering check for MmPVRTRG3mIW6Esg:2466368666:1 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u/0m9av72lp5/1] Sending ICE/STUN request for candidate pair MmPVRTRG3mIW6Esg:2466368666:1 from <sip-proxy-rtpengine-ip> to 93.106.207.11:35551 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Received ICE/STUN response code 0 for candidate pair MmPVRTRG3mIW6Esg:2466368666:1 from 93.106.207.11:35551 to <sip-proxy-rtpengine-ip> May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Setting ICE candidate pair MmPVRTRG3mIW6Esg:2466368666:1 as succeeded May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Best succeeded ICE pair with all components is MmPVRTRG3mIW6Esg:2466368666:1 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] ICE completed, using pair MmPVRTRG3mIW6Esg:2466368666:1 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u/0m9av72lp5/1] Sending ICE/STUN request for candidate pair MmPVRTRG3mIW6Esg:1603072777:1 from <sip-proxy-rtpengine-ip> to 192.168.43.146:35551 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Received ICE/STUN response code 0 for candidate pair MmPVRTRG3mIW6Esg:2466368666:1 from 93.106.207.11:35551 to <sip-proxy-rtpengine-ip> May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] DTLS: Peer certificate accepted May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] DTLS handshake successful May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] DTLS-SRTP successfully negotiated May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] SRTP keys, incoming: May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] --- AES_CM_128_HMAC_SHA1_32 key jZlg8WlCjfEiFoilpz7BQg== salt gUcb/yOXBOIuNY1ItSw= May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] SRTP keys, outgoing: May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] --- AES_CM_128_HMAC_SHA1_32 key e3hQ8pCcc04HD8I7KWdl/g== salt mrNX5VDOk0+Tif2xv5I= May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] DTLS-SRTP successfully negotiated May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] SRTP keys, incoming: May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] --- AES_CM_128_HMAC_SHA1_32 key jZlg8WlCjfEiFoilpz7BQg== salt gUcb/yOXBOIuNY1ItSw= May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] SRTP keys, outgoing: May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] --- AES_CM_128_HMAC_SHA1_32 key e3hQ8pCcc04HD8I7KWdl/g== salt mrNX5VDOk0+Tif2xv5I= May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u/0m9av72lp5/1] Sending ICE/STUN request for candidate pair MmPVRTRG3mIW6Esg:1603072777:1 from <sip-proxy-rtpengine-ip> to 192.168.43.146:35551 May 18 11:47:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:35 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:35 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u/0m9av72lp5/1] Sending ICE/STUN request for candidate pair MmPVRTRG3mIW6Esg:1603072777:1 from <sip-proxy-rtpengine-ip> to 192.168.43.146:35551 May 18 11:47:35 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:36 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:36 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:37 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u/0m9av72lp5/1] Sending ICE/STUN request for candidate pair MmPVRTRG3mIW6Esg:1603072777:1 from <sip-proxy-rtpengine-ip> to 192.168.43.146:35551 May 18 11:47:37 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:37 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:38 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8012] Confirmed peer address as <sip-proxy-rtpengine-ip>:8064 May 18 11:47:38 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Confirmed peer address as 93.106.207.11:35551 May 18 11:47:38 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Kernelizing media stream May 18 11:47:38 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8012] Kernelizing media stream May 18 11:47:38 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:38 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:39 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:39 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:40 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:40 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u/0m9av72lp5/1] Sending ICE/STUN request for candidate pair MmPVRTRG3mIW6Esg:1603072777:1 from <sip-proxy-rtpengine-ip> to 192.168.43.146:35551 May 18 11:47:40 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:41 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:41 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:42 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:42 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:43 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:43 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:43 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:44 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:44 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:45 box proxy_monitor[24109]: SIP test succeeded with proxy 192.98.100.34! May 18 11:47:45 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:45 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:46 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:46 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u/0m9av72lp5/1] Sending ICE/STUN request for candidate pair MmPVRTRG3mIW6Esg:1603072777:1 from <sip-proxy-rtpengine-ip> to 192.168.43.146:35551 May 18 11:47:46 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:47 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:47 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:48 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:48 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:49 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:49 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:50 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:50 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:51 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:51 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:52 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:52 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:53 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:53 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:54 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:54 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:55 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:55 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:55 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:56 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:56 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:57 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:57 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:58 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:58 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:59 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:47:59 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u/0m9av72lp5/1] Sending ICE/STUN request for candidate pair MmPVRTRG3mIW6Esg:1603072777:1 from <sip-proxy-rtpengine-ip> to 192.168.43.146:35551 May 18 11:47:59 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:00 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:00 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:01 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:01 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:02 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:02 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:03 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:03 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:04 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:04 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:05 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:05 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:06 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:06 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:07 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:07 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:07 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:08 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:08 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:09 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:09 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:10 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:10 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:11 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:11 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:12 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:12 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:13 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:13 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:14 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:14 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:15 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:15 box proxy_monitor[24109]: SIP test succeeded with proxy 192.98.100.34! May 18 11:48:15 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:16 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:16 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:17 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:17 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:18 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:18 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:19 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:19 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:20 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:20 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:20 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:21 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:21 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:22 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:22 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:23 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:23 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:24 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:24 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:25 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u/0m9av72lp5/1] Setting ICE candidate pair MmPVRTRG3mIW6Esg:1603072777:1 as failed May 18 11:48:25 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:25 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:26 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:26 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:27 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:27 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:28 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:28 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:29 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:29 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:30 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:30 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:31 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:31 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:32 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:32 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:32 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:33 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:33 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:34 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:35 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:35 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u port 8028] Successful STUN binding request from 93.106.207.11:35551 May 18 11:48:36 box /usr/bin/sip-proxy[18297]: INFO: ===== rtpengine_delete(via-branch=1) May 18 11:48:36 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Received command 'delete' from 127.0.0.1:60722 May 18 11:48:36 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Dump for 'delete' from 127.0.0.1:60722: { "call-id": "e91aj9gq39qr07b6dn6u", "via-branch": "z9hG4bK3605983", "received-from": [ "IP4", "<sip-proxy-rtpengine-ip>" ], "from-tag": "0m9av72lp5", "command": "delete" } May 18 11:48:36 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Scheduling deletion of call branch '0m9av72lp5' in 30 seconds May 18 11:48:36 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Replying to 'delete' from 127.0.0.1:60722 May 18 11:48:36 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Response dump for 'delete' to 127.0.0.1:60722: { "created": 1431938854, "last signal": 1431938854, "tags": { "0m9av72lp5": { "tag": "0m9av72lp5", "created": 1431938854, "in dialogue with": "627A71DC-5559A726000163C1-AA759700", "medias": [ { "index": 1, "type": "audio", "protocol": "UDP/TLS/RTP/SAVPF", "streams": [ { "local port": 8028, "endpoint": { "family": "IPv4", "address": "93.106.207.11", "port": 35551 }, "advertised endpoint": { "family": "IPv4", "address": "93.106.207.11", "port": 35551 ... May 18 11:48:36 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] ... }, "crypto suite": "AES_CM_128_HMAC_SHA1_32", "last packet": 1431938916, "flags": [ "RTP", "RTCP", "filled", "confirmed", "kernelized", "DTLS fingerprint verified" ], "stats": { "packets": 3084, "bytes": 540700, "errors": 0 } }, { "local port": 8029, "endpoint": { "family": "IPv6", "address": "::", "port": 0 }, "advertised endpoint": { "family": "IPv6", "address": "::", "port": 0 }, "crypto suite": "AES_CM_128_HMAC_SHA1_32", "last packet": 1431938854, "flags": [ "RTCP", "fallback RTCP", "filled" ... May 18 11:48:36 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] ... ], "stats": { "packets": 0, "bytes": 0, "errors": 0 } } ], "flags": [ "initialized", "send", "recv", "rtcp-mux", "DTLS-SRTP", "DTLS role active", "ICE" ] } ] }, "627A71DC-5559A726000163C1-AA759700": { "tag": "627A71DC-5559A726000163C1-AA759700", "created": 1431938854, "in dialogue with": "0m9av72lp5", "medias": [ { "index": 1, "type": "audio", "protocol": "RTP/AVP", "streams": [ { "local port": 8012, "endpoint": { "family": "IPv4", "address": "<sip-proxy-rtpengine-ip>", "port": 8064 }, "advertised endpoin ... May 18 11:48:36 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] ... t": { "family": "IPv4", "address": "<sip-proxy-rtpengine-ip>", "port": 8064 }, "last packet": 1431938916, "flags": [ "RTP", "filled", "confirmed", "kernelized" ], "stats": { "packets": 3095, "bytes": 533040, "errors": 0 } }, { "local port": 8013, "endpoint": { "family": "IPv4", "address": "<sip-proxy-rtpengine-ip>", "port": 8065 }, "advertised endpoint": { "family": "IPv4", "address": "<sip-proxy-rtpengine-ip>", "port": 8065 }, "last packet": 1431938854, "flags": [ "RTCP", "filled" ], "stats": { "packets": 0, "bytes": 0, "error ... May 18 11:48:36 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] ... s": 0 } } ], "flags": [ "initialized", "send", "recv" ] } ] } }, "totals": { "RTP": { "packets": 6179, "bytes": 1073740, "errors": 0 }, "RTCP": { "packets": 0, "bytes": 0, "errors": 0 } }, "result": "ok" } May 18 11:49:06 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Call branch '0m9av72lp5' deleted, no more branches remaining May 18 11:49:06 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] Final packet stats: May 18 11:49:06 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] --- Tag '627A71DC-5559A726000163C1-AA759700', created 1:32 ago, in dialogue with '0m9av72lp5' May 18 11:49:06 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] ------ Media #1 (audio over RTP/AVP) using PCMU/8000 May 18 11:49:06 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] --------- Port 8012 <> <sip-proxy-rtpengine-ip>:8064 , 3110 p, 535620 b, 0 e, 1431938917 last_packet May 18 11:49:06 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] --------- Port 8013 <> <sip-proxy-rtpengine-ip>:8065 (RTCP), 0 p, 0 b, 0 e, 1431938854 last_packet May 18 11:49:06 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] --- Tag '0m9av72lp5', created 1:32 ago, in dialogue with '627A71DC-5559A726000163C1-AA759700' May 18 11:49:06 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] ------ Media #1 (audio over UDP/TLS/RTP/SAVPF) using PCMU/8000 May 18 11:49:06 box rtpengine[4025]: [e91aj9gq39qr07b6dn6u] --------- Port 8028 <> 93.106.207.11:35551, 3101 p, 543584 b, 0 e, 1431938917 last_packet

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Issue with Asterisk interconnection for VoiceMail
by Igor Potjevlesch 18 May '15

18 May '15

Hello, I experienced a strange issue with some of VoIP accounts. When the INVITE comes into MANAGE_FAILURE, after timeout, the config identifies, with "dialplan", the right Asterisk instance that should handle the call for voicemail. This part is okay, and results in a new INVITE with the Request-URI formed with the right domain (eg. sip: <sip:%3cNUMBER%3e@asterisk3> <NUMBER>@asterisk3). Then, the request goes to RELAY. Here is the issue: sometimes, the request is forwarded to the IP of the UA (the one initially contacted) instead of the IP of Asterisk. I can't figure out the difference between a succeeded call and a failed one. If someone has an idea. Here is the config that handles the VoiceMail: failure_route[MANAGE_FAILURE] { [.] if (isflagset(24)) { $avp(s:inv_timeout) = "5"; t_set_fr($avp(s:inv_timeout)*1000); if (avp_db_load("$to/username","$avp(s:vm_uri)/usr_vm")) { resetflag(24); avp_pushto("$ruri","$avp(s:vm_uri)"); # Dynamic routing if (avp_db_load("$ruri/username","$avp(s:client)/usr_fai")) { if (dp_translate("2","$avp(s:client)/$avp(s:dest)") == 1) { $ru = "sip:" + $rU + "@" + $avp(s:dest); } else { # Load default voicemail $avp(s:client) = "DEFAULT_VM"; dp_translate("2","$avp(s:client)/$avp(s:dest)"); $ru = "sip:" + $rU + "@" + $avp(s:dest); }; } else { # Load default voicemail $avp(s:client) = "DEFAULT_VM"; dp_translate("2","$avp(s:client)/$avp(s:dest)"); $ru = "sip:" + $rU + "@" + $avp(s:dest); } } else { xlog("L_WARN","time=[$Tf] call id=[$ci] call seq=[$cs] contact header=[$ct] from uri=[$fu] from tag=[$ft] request's method=[$rm] request's uri=[$ru] to uri=[$tu] to tag=[$tt] sip message id=[$mi] process id=[$pp] ip source=[$si] flags=[$mf], User have no mail box\n"); exit; }; prefix("710"); xlog("L_WARN","time=[$Tf] call id=[$ci] call seq=[$cs] contact header=[$ct] from uri=[$fu] from tag=[$ft] request's method=[$rm] request's uri=[$ru] to uri=[$tu] to tag=[$tt] sip message id=[$mi] process id=[$pp] ip source=[$si] flags=[$mf], failure route to Voice Mail\n"); route(RELAY); exit; } Regards, Igor.

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dbtext why is my user not recognised
by jaflong jaflong 18 May '15

18 May '15

cat > /usr/local/etc/kamailio/dbtext/subscriber username(str) password(str) ha1(str) domain(str) ha1b(str) user1:user1:xxx:10.1.1.1:xxx 9350) DEBUG: auth_db [authorize.c:498]: auth_check(): realm [10.31.8.1] table [subscriber] flags [1] 2(9350) DEBUG: auth [api.c:96]: pre_auth(): auth: digest-algo: MD5 parsed value: 1 2(9350) DEBUG: db_text [dbt_file.c:102]: dbt_load_file(): request for table [subscriber] 2(9350) DEBUG: db_text [dbt_file.c:109]: dbt_load_file(): db is [/usr/local/etc/kamailio/dbtext] 2(9350) DEBUG: db_text [dbt_file.c:124]: dbt_load_file(): loading file [/usr/local/etc/kamailio/dbtext/subscriber] 2(9350) DEBUG: db_text [dbt_tb.c:187]: dbt_table_new(): mtime is 1431951381 2(9350) DEBUG: db_text [dbt_file.c:480]: dbt_load_file(): error at row=0 col=0 c= 2(9350) ERROR: db_text [dbt_lib.c:307]: dbt_db_get_table(): could not load database from file [subscriber] 2(9350) ERROR: db_text [dbt_base.c:200]: dbt_query(): table subscriber does not exist! 2(9350) ERROR: auth_db [authorize.c:187]: get_ha1(): failed to query database 2(9350) exec: *** cfgtrace:request_route=[AUTH] c=[/usr/local/etc/kamailio/kamailio.cfg] l=498 a=26 n=auth_challenge 2(9350) DEBUG: auth [challenge.c:127]: get_challenge_hf(): build_challenge_hf: realm='10.31.8.1' 2(9350) DEBUG: auth [challenge.c:269]: get_challenge_hf(): auth: 'Proxy-Authenticate: Digest realm="10.31.8.1", nonce="VVnqGlVZ6O4l1T62BGADuI2p8uAnUYeu" regards

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Setting up a presence AS with Kamailio IMS.
by Mihail Dakov 18 May '15

18 May '15

Hi all, I have successfully setup a Kamailio IMS environment using just a single machine running three kamailio instances. I am using kamailio 4.2.4 compiling it myself. I followed this guide: http://nil.uniza.sk/instant-messaging/simple/configuring-im-and-presence-ka… to setup a presence AS using the scscf instance. I used kamdbctl script to create the presence tables with one inconsistence with the guide. I had to create a db in which I created a couple of columns (table_name, table_version) to satisfy the kamdbctl script as it was failing when running "kamdbctl presence". At the moment the problem is that my kamailio instance is failing with the following error: scscf/scscf[5256]: ERROR: <core> [db.c:422]: db_table_version(): invalid type (3) or nul (0) version columns for presentity ... and then some time later: scscf/scscf[5256]: ERROR: <core> [db.c:440]: db_check_table_version(): querying version for table presentity scscf/scscf[5256]: ERROR: presence [presence.c:322]: mod_init(): error during table version check scscf/scscf[5256]: ERROR: <core> [sr_module.c:968]: init_mod(): Error while initializing module presence (/home/ng40/kamailio-4.2.4/modules/presence/presence.so) scscf/scscf[5256]: DEBUG: cdp_avp [mod.c:224]: cdp_avp_destroy(): Destroying module cdp_avp I suspect that the problem relays on creating manually the db where I inserted the two columns mentioned above (table_name, table_version) so that I could execute "kamdbctl presence". Could someone assist me into solving this issue? br, md.

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critical crash with dbtext
by jaflong jaflong 18 May '15

18 May '15

HI When using dbtext setup I receive a CRITICAL error and shut down. This is the debug log. My config settings are described below. Please advise how I should fo the correct dbtext config 9(21196) DEBUG: <core> [sr_module.c:920]: init_mod_child(): DEBUG: init_mod_child (-4): acc 9(21196) DEBUG: <core> [sr_module.c:920]: init_mod_child(): DEBUG: init_mod_child (-4): auth_db 9(21196) DEBUG: <core> [sr_module.c:920]: init_mod_child(): DEBUG: init_mod_child (-4): debugger 9(21196) DEBUG: debugger [debugger_mod.c:179]: child_init(): rank is (-4) 9(21196) DEBUG: <core> [local_timer.c:74]: init_local_timer(): init_local_timer: timer_list between 0x9e7268 and 0xa2b268 9(21196) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0x9e70a0, 8, 4, 0x7f5aaa402364), fd_no=0 9(21196) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0x9e70a0, 9, 4, 0x7f5aaa4023f0), fd_no=1 9(21196) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0x9e70a0, 10, 4, 0x7f5aaa40247c), fd_no=2 9(21196) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0x9e70a0, 11, 4, 0x7f5aaa402508), fd_no=3 9(21196) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0x9e70a0, 12, 4, 0x7f5aaa402594), fd_no=4 9(21196) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0x9e70a0, 13, 4, 0x7f5aaa402620), fd_no=5 9(21196) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0x9e70a0, 14, 4, 0x7f5aaa4026ac), fd_no=6 9(21196) DEBUG: <core> [io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0x9e70a0, 15, 4, 0x7f5aaa402738), fd_no=7 9(21196) CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 8 9(21196) DEBUG: <core> [tcp_main.c:3537]: handle_ser_child(): DBG: handle_ser_child: dead child 1, pid 21188 (shutting down?) 9(21196) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9e70a0, 8, -1, 0x0) fd_no=8 called 9(21196) CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 9 9(21196) DEBUG: <core> [tcp_main.c:3537]: handle_ser_child(): DBG: handle_ser_child: dead child 2, pid 21189 (shutting down?) 9(21196) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9e70a0, 9, -1, 0x0) fd_no=7 called 9(21196) CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 10 9(21196) DEBUG: <core> [tcp_main.c:3537]: handle_ser_child(): DBG: handle_ser_child: dead child 3, pid 21190 (shutting down?) 9(21196) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9e70a0, 10, -1, 0x0) fd_no=6 called 9(21196) CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 11 9(21196) DEBUG: <core> [tcp_main.c:3537]: handle_ser_child(): DBG: handle_ser_child: dead child 4, pid 21191 (shutting down?) 9(21196) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9e70a0, 11, -1, 0x0) fd_no=5 called 9(21196) CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 12 9(21196) DEBUG: <core> [tcp_main.c:3537]: handle_ser_child(): DBG: handle_ser_child: dead child 5, pid 21192 (shutting down?) 9(21196) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9e70a0, 12, -1, 0x0) fd_no=4 called 9(21196) CRITICAL: <core> [pass_fd.c:293]: receive_fd(): EOF on 13 9(21196) DEBUG: <core> [tcp_main.c:3537]: handle_ser_child(): DBG: handle_ser_child: dead child 6, pid 21193 (shutting down?) 9(21196) DEBUG: <core> [io_wait.h:610]: io_watch_del(): DBG: io_watch_del (0x9e70a0, 13, -1, 0x0) fd_no=3 called 8(21195) INFO: db_text [dbt_base.c:99]: dbt_init(): using database at: /usr/local/etc/kamailio/dbtext 8(21195) DEBUG: db_text [dbt_lib.c:129]: dbt_cache_get_db(): looking for db /usr/local/etc/kamailio/dbtext! 8(21195) DEBUG: db_text [dbt_lib.c:139]: dbt_cache_get_db(): db already cached! 8(21195) DEBUG: <core> [sruid.c:101]: sruid_init(): root for sruid is [ulcx-5555c137-52cb-] (0 / 19) 8(21195) DEBUG: <core> [sr_module.c:920]: init_mod_child(): DEBUG: init_mod_child (-2): acc 8(21195) DEBUG: <core> [sr_module.c:920]: init_mod_child(): DEBUG: init_mod_child (-2): auth_db 8(21195) INFO: db_text [dbt_base.c:99]: dbt_init(): using database at: /usr/local/etc/kamailio/dbtext 8(21195) DEBUG: db_text [dbt_lib.c:129]: dbt_cache_get_db(): looking for db /usr/local/etc/kamailio/dbtext! 8(21195) DEBUG: db_text [dbt_lib.c:139]: dbt_cache_get_db(): db already cached! 8(21195) DEBUG: <core> [sr_module.c:920]: init_mod_child(): DEBUG: init_mod_child (-2): debugger 8(21195) DEBUG: debugger [debugger_mod.c:179]: child_init(): rank is (-2) 8(21195) DEBUG: ctl [ctl.c:325]: mod_child(): ctl: mod_child(0), fork_process=0, csl=0x2682080 8(21195) DEBUG: ctl [ctl.c:332]: mod_child(): ctl: 0 io_listen_loop(1, 0x2682080) 8(21195) INFO: ctl [io_listener.c:225]: io_listen_loop(): io_listen_loop: using epoll_lt io watch method (config) 8(21195) DEBUG: ctl [io_listener.c:257]: io_listen_loop(): io_listen_loop: adding socket 4, type 2, transport 3 (/tmp/kamailio_ctl) 8(21195) DEBUG: ctl [../../io_wait.h:388]: io_watch_add(): DBG: io_watch_add(0x7f5aaf2a36c0, 4, 2, 0x2682080), fd_no=0 --------------------------------------------- cd /usr/local/etc/kamailio mkdir dbtext cd dbtext --------------------------------------------- cat > /usr/local/etc/kamailio/dbtext/version table_name(str) table_version(int) subscriber:6 location:6 aliases:6 --------------------------------------------- cat > /usr/local/etc/kamailio/dbtext/location username(str) contact(str) expires(int) q(double) callid(str) cseq(int) --------------------------------------------- cat > /usr/local/etc/kamailio/dbtext/subscriber username(str) password(str) ha1(str) domain(str) ha1b(str) testuser:testuser:xxx:10.1.1.5:xxx --------------------------------------------- #!define DBURL "text:///usr/local/etc/kamailio/dbtext" #!endif --------------------------------------------- # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) ---------------------------------------------

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