- sr-users - kamailio.org

rtjson + mhomed and record route
by Diego Nadares 22 Dec '16

22 Dec '16

Hi list, We have two eth, one with a private ip and the other with a public ip. We have MHOMED configured. The call comes from private network and then is routed to a public network via rtjson routing. The is routed but the thing is that the record route is being set with the private one (It's the same when calls come from public and is routed to a private network, record route has the public and not the private). With mhomed configured shouldn't Record Route be updated as VIA header ".... When activated, sip-router will select a socket that can reach the destination (to be able to connect to the remote address). (sip-router opens a UDP socket to the destination, then it retrieves the local IP which was assigned by the operating system to the new UDP socket. Then this socket will be closed and the retrieved IP address will be used as IP address in the Via/Record-Route headers)..." This is part of ngrep U 2016/12/22 11:24:15.019643 172.16.213.21:5060 -> 172.16.213.38:5060 INVITE sip:11111111111@172.16.213.38:5060 SIP/2.0. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. From: "Test" <sip:2222222222@172.16.213.21:5060;user=phone>;tag=1. To: <sip:1111111111@172.16.213.38:5060>. Call-ID: 1-13024(a)172.16.213.21. CSeq: 1 INVITE. Contact: "2222222222 <sip:2222222222@172.16.213.21:5060>. .... U 2016/12/22 11:24:15.023923 172.16.213.38:5060 -> 172.16.213.21:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 172.16.213.21:5060;branch=z9hG4bK-13024-1-0. .... U 2016/12/22 11:24:15.038695 XXX.XXX.XXX.01:5060 -> XXX.XXX.XXX.02:5060 INVITE sip:1111111111@XXX.XXX.XXX.02:5060 SIP/2.0. *-----> Record-Route: <sip:172.16.213.38;lr;ftag=1;did=1111>. <----* Via: SIP/2.0/UDP XXX.XXX.XXX.01;branch=z9hG4bKc09f Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.NLOBF6 This is my cfg mhomed=1 request_route { # per request initial checks route(REQINIT); # NAT detection #route(NATDETECT); # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { route(RELAY); } exit; } # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # handle retransmissions if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); # authentication route(AUTH); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { record_route(); } Thanks in advance. Diego.

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Socket in rtjson module
by Diego Nadares 22 Dec '16

22 Dec '16

Hi Guys, Anybody knows if each 'socket' specified in routes struct should be use to send the messages? The socket is configured in kamailio. In kamailio.cfg listen=udp:172.16.213.38:5060 json struct "routes": [ { "uri": "sip:127.0.0.1:5080", "dst_uri": "sip:127.0.0.1:5082", "path": "<sip:127.0.0.1:5084>, <sip:127.0.0.1:5086>", ----> "socket": "udp:172.16.213.38:5060",<---- "headers": { "from": { "display": "Alice", "uri": "sip:alice@127.. In our kamailio has two eth and I want to select which one to use but I can't make it work through rtjson. In the code of the module I can't find where is being assigned the socket. Thanks in advance! Diego

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kamailio dispatcher can not forward call in Amazon EC2
by huan nguyen duy 22 Dec '16

22 Dec '16

Hi everybody, I have difficult time when configure load balancing for kamailio servers, i’m very appreciated if you help me figure out the problem. This email contain images, if you can see images please read problem_detail.pdf. I want to setup the system as following: Both Load Balancer (LB) and Sip Server (SIP) run on Amazon EC2, (LB) using dispatcher module. I enclose here config files for both (LB) and (SIP). Please view the files attached. Figure (1) is what i saw when run kamctl dispatcher dump: Figure (1) I created 2 users with following information (extract from mysql cluster database, table subscriber): Figure (2) Then use sip client (Linphone) connect to (LB), registration is success, below is records from location table: Figure (3) But i can’t make call between 2 users. Below is data i get from wireshark: Load Balancer: Figure(4) SIP Server: Figure (5) I also enclose 2 files export from wireshark for detail information. Steve Nguyen, --------------------------------------- Cell phone: (84)909585104 Mobile app developer Phan Thi Company Sai Gon, VietNam

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kamailio dispatcher can not forward call in Amazon EC2
by Huân Nguyễn Duy 22 Dec '16

22 Dec '16

Hi everybody, I have difficult time when configure load balancing for kamailio servers, i’m very appreciated if you help me figure out the problem. This email contain images, if you can see images please read problem_detail.pdf. I want to setup the system as following: Both Load Balancer (LB) and Sip Server (SIP) run on Amazon EC2, (LB) using dispatcher module. I enclose here config files for both (LB) and (SIP). Please view the files attached. Figure (1) is what i saw when run kamctl dispatcher dump: Figure (1) I created 2 users with following information (extract from mysql cluster database, table subscriber): Figure (2) Then use sip client (Linphone) connect to (LB), registration is success, below is records from location table: Figure (3) But i can’t make call between 2 users. Below is data i get from wireshark: Load Balancer: Figure(4) SIP Server: Figure (5) I also enclose 2 files export from wireshark for detail information.

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Memory leak in Kamailio 4.4.4 TLS Module
by raja raja 22 Dec '16

22 Dec '16

Hi, shared memory not freed once TLS connection close means shm available memory decreasing for new connection that leads to the following error *tls: ssl bug #1491 workaround: not enough memory for safe operation* *TestCase:* 1. start kamailio 2. use sample program to establish tls connection and close imediately. 3. Kamailio Log is For first TLS connection tls [tls_server.c:189]: tls_complete_init(): tls: enough memory for safe operation: shm=63604096 threshold1=12058624 tls [tls_server.c:649]: tls_h_tcpconn_clean(): tls shm_available after shm_free: 63545192 For second TLS connection tls [tls_server.c:189]: tls_complete_init(): tls: enough memory for safe operation: shm=63545192 threshold1=12058624 tls [tls_server.c:649]: tls_h_tcpconn_clean(): tls shm_available after shm_free: 63489352 *Memory Leak:* for tls connection connection 63604096 - 63545192 = 58904 bytes not freed. for tls connection connection 63545192 - 63489352 = 55840 bytes not freed. I tested same scenario on kamailio - 4.2.4, 4.3.4 versions result is same. Could you please provide the fix? Thanks & Regards, Raja Chegondi.

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register_timer cb function does not have access to shared memoery var
by Andy 22 Dec '16

22 Dec '16

in mod_init I have something like list = (struct list_t*) shm_malloc ..... this list is available from any functions in the module, now there is a timer registered, in the same module register_timer(callback_f, 0 , 5) the callback_f function when tries to use list, always sees as empty -- View this message in context: http://sip-router.1086192.n5.nabble.com/register-timer-cb-function-does-not… Sent from the Users mailing list archive at Nabble.com.

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new URI [] shorter than old URI - error
by Nelson Migliaro 21 Dec '16

21 Dec '16

Hello, My SIP vendor request me to replace the from in order to authenticate calls. I use: uac_replace_from("111111111", "sip:111111111@vendor-domain.com"); And when finishing calls I have this error: ERROR: uac [replace.c:607]: restore_uri(): new URI [] shorter than old URI [sip:34912345678@vendor-public-ip] Information ------------------------------------------------------------------ 34912345678 -> original CLI 111111111 -> my vendor contract vendor-domain.com -> domain of the vendor ------------------------------------------------------------------ I use the function uac_replace_from in the request_route My Kamailio version is: 4.2.8-1.1 Thank you Nelson.-

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kamailio 4.4 segmentation fault when dp_translate match dialplan with ftmatch operator and empty match expression
by Julia Boudniatsky 21 Dec '16

21 Dec '16

version: kamailio 4.4.4 id:852c47 ENTRY: { PRIO: 16765956 MATCHOP: 2 *MATCHEXP: * MATCHLEN: 0 SUBSTEXP: REPLEXP: ATTRS: 1 } Please see generated core file in extension. Thank you, Julia.

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nathelper - variable natping_interval per registration
by Phil Lavin 20 Dec '16

20 Dec '16

Hi all, Is it possible to set the natping_interval to a different value, per registration, at the time the user registers? For example, user A has 10 seconds and user B has 60 seconds. Thanks

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auth result -3
by Jon Bonilla (Manwe) 19 Dec '16

19 Dec '16

Hi all I'm getting an auth challenge result of -3 and I don't know what it means as it's not listed in the auth module doc. Here's syslog' output: db_postgres [km_dbase.c:230]: db_postgres_submit_query(): sending query ok: 0x7f36d1f75f00 (2) - [select secret from sip_buddies where username='48481628780'] db_postgres [km_res.c:108]: db_postgres_get_columns(): 1 columns returned from the query db_postgres [km_res.c:133]: db_postgres_get_columns(): RES_NAMES(0x7f36d1f92238)[0]=[secret] db_postgres [km_res.c:166]: db_postgres_get_columns(): use DB1_STRING result type db_postgres [km_res.c:211]: db_postgres_convert_rows(): no rows returned from the query auth_db [authorize.c:199]: get_ha1(): no result for user '48481628780@' INFO: <script>: Ha fallado la auth con resultado -3 cheers Jon

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INFO: jsonrpcs used instead of mi_fifo in the default kamailio.cfg and kamctl
by Daniel-Constantin Mierla 19 Dec '16

19 Dec '16

Hello, with MI code planned to be removed, I updated the kamctl and default kamailio.cfg to use jsonrpcs (jsonrpc server) module instead of mi_fifo to interact with kamailio at runtime. The default fifo file for rpc command is /var/run/kamailio/kamailio_rpc.fifo and expects JSONRPC commands as described at: - http://kamailio.org/docs/modules/devel/modules/jsonrpcs.html#idp31525708 Switching back to mi_fifo is still possible by loading the module and editing kamctlrc, but the module might not make it to 5.0, so for new installations and testing is better not to use it. Few rpc commands are missing in carrierroute, otherwise the kamctl commands should work just as before, but be aware that the output is now in json format. Testing of the master branch with the latest kamctl and providing feedback will be really appreciated -- I could do tests only for the modules I typically use. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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Forking with 404 replies
by José Seabra 19 Dec '16

19 Dec '16

Hello there, I have a Kamailio server acting as registrar server and when the subscriber has more than 1 AOR, kamailio sends an INVITES to each AOR, if one of them replies 404 what should be the kamailio behavior: 1. should Kamailio reply with an ACK and not pass the 404 to UAC? 2. should Kamailio forward the 404 to UAC? I'm asking this because i'm facing the second behavior, that is, kamailio is forwarding the 404 to UAC and the call is terminated. -- Regards José Seabra

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RFC: renaming the modules with - (dash) in name
by Daniel-Constantin Mierla 19 Dec '16

19 Dec '16

Hello, given the work for 5.0 which will bring major changes on source tree structure, I am considering to remove the - (dash) from the modules that have it in the name. There are 4 of them now: cpl-c janssonrpc-c jsonrpc-c jsonrpc-s One of the reasons is that this kind of name doesn't work for the feature of defining MOD_modname for each loaded module, because - is a separator for ids in config language grammar, so for example, MOD_jsonrpc-c cannot be used in an #!ifdef. More over, for embedded interpreters, most of exported functions there use kamailio module name as package name, and again this will break. Of course, a different package name can be defined, but won't be coherent anymore. Anyone against this renaming? Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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Re: [SR-Users] Question on multiple 200OK received after forking
by Daniel-Constantin Mierla 19 Dec '16

19 Dec '16

Hello, well, the specs say that all 200ok must be forwarded to the caller, because a middle point cannot decide for the user making the call. Even sending an ACK for the 200ok from a middle point is not enough because the callee of that branch will think it is in an active call. The caller can take different actions, the usual is to ACK the 2nd 200ok and then immediately send BYE. But it can also decide to end previous branch and switch to this one (e.g., uses better codecs, or has video, etc...). Cheers, Daniel On 14/12/2016 15:05, Marino Mileti wrote: > > Hi guys, > > > > I've a Kamailio instance listening on both eth0 and wlan0. > > Kamailio forks call to two branches, so both clients rings together. > > > > In case (very very very rare J) of simultaneous answers is there any > way to kill/hangup/destroy/cancel the "slowest" leg? I've tried to > arrange a semaphore with lock/unlock capabilities of cfgutils, I've > used the drop command but client still sending the 200OK because it's > waiting for an ACK L > > > > Any help will be very appreciated J > > > > /Marino Maria Mileti/// > > /marino.mileti(a)alice.it/ <mailto:marino.mileti@alice.it>// > > / / > > /cid:006a01cb6b0e$67eecdae$_CDOSYS2.0//Reduce your energy consumption > and keep polar bears on ice!/ > > > > > > ------------------------------------------------------------------------ > Avast logo > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > > > > _______________________________________________ > 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 www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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(Correction) unusual characters added to reINVITE message
by AL S 19 Dec '16

19 Dec '16

Hi, An INVITE with following FROM and TO headers received by Kamailio: From: <sip:+11111111111@10.10.10.10;user=phone>;tag=gK006cb08b To: <sip:+12222222222@20.20.20.20;user=phone> Kamailio forward it to next node with following FROM and TO headers: From: <sip:+11111111111@10.10.10.10>;tag=gK006cb08b To: <sip:+12222222222@20.20.20.20;user=phone> After 15 min, reINVITE for the same call received by Kamailio with following FROM and TO headers: From: <sip:12222222222@20.20.20.20;user=phone>;tag=gK04c77151 To: <sip:11111111111@10.10.10.10>;tag=gK006cb08b Kamailio forward reINVITE to next node with following unusual characters in TO headers From: <sip:12222222222@20.20.20.20;user=phone>;tag=gK04c77151 To: <sip:11111111111@10.10.10.10\000\000;user=phone>;tag=gK006cb08b I was wondering if there is a bug in Kamailio to insert these unusual characters. Thanks, AS

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Limiting calls based on CLI and dst number
by Cibin Paul 19 Dec '16

19 Dec '16

Hi, Is there a way to limit the number of active calls per cli or dst number basis? Yesterday my system had a ddos attack in which 123456 (CLI) was hitting 2345 (DID) at a rate of 100-200 calls per minute. I would like to restrict to 2-4 simultaneous calls per CLI/DST. Any help would be appreciated. Thanks, Cibin

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http_async_client - queries
by Ramachandran, Agalya (Contractor) 17 Dec '16

17 Dec '16

Hi team, We are trying to use kamailio for our project purpose, which will be used as a proxy server. Am trying to use "http_async_client" module to send my rest queries asynchronously. I have few queries regarding the usage of http_async_query() and using PUT method. http_append_header("Authorization: Bearer $var(auth)"); http_append_header("Content-Type: application/json"); http_set_method("PUT"); http_async_query("http://sdk-evm-wcdc-c-001.rtc.sys.comcast.net/events/createxmpprooteventfor…", "{'from_tn':'$fU','to_tn':'$tU','callid':'ci'}", "HTTP_REPLY"); Here am passing from, to, callid in post data field. Passing as 'key':'value' is mandatory or I can pass only value- ? In the file, async_http.c - in async_send_query(), if I modify the post data to the format I need, I see the value is truncated up to three fields only. How can I achieve that my payload format whatever I form in async_send_query(),doesn't get truncated and will be passed to the server-? Can you please explain how this works-? Also, I observed that kamailio is restricting the content-length of HTTP request to 256. Is this something modifiable-? If so please guide me how this can be achieved. Regards, Agalya

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sipp dead calls
by Diego Nadares 16 Dec '16

16 Dec '16

Hi guys, We are testing kamailio with sipp. We are running it with 20cps and some calls do the following. When Kamailio is processing the 'ringing' a '200ok' arrives in the middle. First, kamailio forwards the 200 ok and then the ringing. 'ACK' arrives and, I suppose, the call is established. The thing is than when the 'BYE' arrives kamailio responds with a 404. This is the summary of the call Id Time Source Destination Protocol Len Info 6483 2016-12-14 11:26:06.264697 SIPP-A KAMAILIO SIP/SDP 692 Request: INVITE sip:11111111@172.16.213.38:5060 | 6484 2016-12-14 11:26:06.264937 KAMAILIO SIPP-A SIP 367 Status: 100 trying -- your call is important to us | 6485 2016-12-14 11:26:06.266327 KAMAILIO SIPP-B SIP/SDP 1179 Request: INVITE sip:22222222@172.16.213.31:5060 | *6486 2016-12-14 11:26:06.267217 SIPP-B KAMAILIO SIP 566 Status: 180 Ringing | * 6487 2016-12-14 11:26:06.267268 SIPP-B KAMAILIO SIP/SDP 733 Status: 200 OK | 6488 2016-12-14 11:26:06.267758 KAMAILIO SIPP-A SIP/SDP 788 Status: 200 OK | *6489 2016-12-14 11:26:06.267833 *KAMAILIO SIPP-A* SIP 442 Status: 180 Ringing | * 6490 2016-12-14 11:26:06.268868 SIPP-A KAMAILIO SIP 493 Request: ACK sip:127.0.0.8;line=sr-N6IAzBFwMJZfWJZLM.M7MlF-W.y6Mx14NEt7Nw05NhPQKjaP | 6491 2016-12-14 11:26:06.269162 KAMAILIO SIPP-B SIP 609 Request: ACK sip:172.16.213.31:5060;transport=UDP | 6492 2016-12-14 11:26:06.269614 SIPP-A KAMAILIO SIP 404 Request: BYE sip:11111111@172.16.213.38:5060 | 6493 2016-12-14 11:26:06.269782 KAMAILIO SIPP-A SIP 348 Status: *404 Not here* | We are using modules rtjson, evapi, uac, topoh, rtpproxy for all calls. My debug level is -1. With higher levels this behavior increase. Kamailio is running in a virtual machine with centos7 with 8 cores and 8gb of ram. Do you need any further information? I can send you a pcap or ngrep file. Best regards, Diego.

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unusual characters added to reINVITE message by Kamailio
by AL S 16 Dec '16

16 Dec '16

Hi, An INVITE with following FROM and TO headers received by Kamailio: From: <sip:+11111111111@10.10.10.10;user=phone>;tag=gK006cb08b To: <sip:+12222222222@20.20.20.20;user=phone> Kamailio forward it to next node with following FROM and TO headers: From: <sip:+11111111111@10.10.10.10>;tag=gK006cb08b To: <sip:+12222222222@20.20.20.20;user=phone> After 15 min, reINVITE for the same call received by Kamailio with following FROM and TO headers: From: <sip:7272697443@71.44.13.84;user=phone>;tag=gK04c77151 To: <sip:2057030997@72.31.12.161>;tag=gK006cb08b Kamailio forward reINVITE to next node with following unusual characters in TO headers From: <sip:7272697443@71.44.13.84;user=phone>;tag=gK04c77151 To: <sip:2057030997@72.31.12.161\000\000;user=phone>;tag=gK006cb08b I was wondering if there is a bug in Kamailio to insert these unusual characters. Thanks, AS

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ACK not routed to destination
by Wim Van Cauwenbergh 16 Dec '16

16 Dec '16

Hi, I have setup kamailio with the dispatcher module and have a problem with the ACK in response to 200OK not being forwarded by kamailio: Scenario is very simple: sipp(.219) --> kamailio (.220) --> sipp(.239) SIP trace: root@DNS-enum-8Gb-3:/var/log# ngrep -W byline -d eth0 port 5060 interface: eth0 (10.57.26.0/255.255.255.0) filter: (ip or ip6) and ( port 5060 ) # U 10.57.26.219:5060 -> 10.57.26.220:5060 INVITE sip:service@10.57.26.220:5060 SIP/2.0. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: sip:sipp@10.57.26.219:5060. Max-Forwards: 70. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.219. s=-. c=IN IP4 10.57.26.219. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Server: kamailio (4.2.0 (x86_64/linux)). Content-Length: 0. . # U 10.57.26.220:5060 -> 10.57.26.239:5060 INVITE sip:service@10.57.26.220:5060 SIP/2.0. Record-Route: <sip:10.57.26.220;lr>. Via: SIP/2.0/UDP 10.57.26.220;branch=z9hG4bKf92d.33550a94aa52aef3978cb507902d8aaa.0. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: sip:sipp@10.57.26.219:5060. Max-Forwards: 69. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.219. s=-. c=IN IP4 10.57.26.219. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.239:5060 -> 10.57.26.220:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.57.26.220;branch=z9hG4bKf92d.33550a94aa52aef3978cb507902d8aaa.0, SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Length: 0. . # U 10.57.26.239:5060 -> 10.57.26.220:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.220;branch=z9hG4bKf92d.33550a94aa52aef3978cb507902d8aaa.0, SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Length: 0. . # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.219:5060 -> 10.57.26.220:5060 ACK sip:10.57.26.239:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-5. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 ACK. Contact: sip:sipp@10.57.26.219:5060. Max-Forwards: 70. Subject: Performance Test. XContactinOK: <sip:10.57.26.239:5060;transport=UDP>. Content-Length: 0. . # U 10.57.26.239:5060 -> 10.57.26.220:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.220;branch=z9hG4bKf92d.33550a94aa52aef3978cb507902d8aaa.0, SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # *U 10.57.26.219:5060 <http://10.57.26.219:5060> -> 10.57.26.220:5060 <http://10.57.26.220:5060>* *ACK sip:10.57.26.239:5060;transport=UDP SIP/2.0.* *Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-5.* *From: sipp <sip:sipp@10.57.26.219:5060 <http://sip:sipp@10.57.26.219:5060>>;tag=15698SIPpTag021.* *To: sut <sip:service@10.57.26.220:5060 <http://sip:service@10.57.26.220:5060>>;tag=14783SIPpTag018.* *Call-ID: 1-15698(a)10.57.26.219 <1-15698(a)10.57.26.219>.* *CSeq: 1 ACK.* *Contact: sip:sipp@10.57.26.219:5060 <http://sip:sipp@10.57.26.219:5060>.* *Max-Forwards: 70.* *Subject: Performance Test.* *XContactinOK: <sip:10.57.26.239:5060;transport=UDP>.* *Content-Length: 0.* . # U 10.57.26.219:5060 -> 10.57.26.220:5060 BYE sip:10.57.26.239:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-7. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 2 BYE. Contact: sip:sipp@10.57.26.219:5060. Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-7. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 2 BYE. Server: kamailio (4.2.0 (x86_64/linux)). Content-Length: 0. . # U 10.57.26.219:5060 -> 10.57.26.220:5060 BYE sip:service@10.57.26.220:5060 SIP/2.0. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1--1. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 3 BYE. Max-Forwards: 70. Contact: <sip:sipp@10.57.26.219:5060;transport=UDP>. Content-Length: 0. . # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 404 Not here. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1--1. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 3 BYE. Server: kamailio (4.2.0 (x86_64/linux)). Content-Length: 0. . # U 10.57.26.239:5060 -> 10.57.26.220:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.220;branch=z9hG4bKf92d.33550a94aa52aef3978cb507902d8aaa.0, SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 10.57.26.220:5060 -> 10.57.26.219:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 10.57.26.219:5060;branch=z9hG4bK-15698-1-0. From: sipp <sip:sipp@10.57.26.219:5060>;tag=15698SIPpTag021. To: sut <sip:service@10.57.26.220:5060>;tag=14783SIPpTag018. Call-ID: 1-15698(a)10.57.26.219. CSeq: 1 INVITE. Contact: <sip:10.57.26.239:5060;transport=UDP>. Content-Type: application/sdp. Content-Length: 135. . v=0. o=user1 53655765 2353687637 IN IP4 10.57.26.239. s=-. c=IN IP4 10.57.26.239. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. *kamailio.conf:* # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo") modparam("ctl", "binrpc", "unix:/var/run/kamailio/kamailio_ctl") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 0) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "log_flag", 1) modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- tm params ----- modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- #modparam("dispatcher", "db_url", # "mysql://openser:openserro@localhost/openser") modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list") modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") ####### Routing Logic ######## # main request routing logic route { # per request initial checks route(REQINIT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); # account only INVITEs if (is_method("INVITE")) { setflag(1); # do accounting } # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); route(RELAY); } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { if (!mf_process_maxfwd_header("15")) { sl_send_reply("483","Too Many Hops"); exit; } # 1= only check the header RURI, 1126 checks some other general stuff if(!sanity_check("1126", "1")) { xlog("Malformed SIP message from $si:$sp\n"); exit; } } # Handle requests within SIP dialogs route[WITHINDLG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing xlog("#######WITHINDLG:hastotag $si:$sp\n"); if (loose_route()) { xlog("#######WITHINDLG:looseroute $si:$sp\n"); if (is_method("BYE")) { setflag(1); # do accounting ... setflag(3); # ... even if the transaction fails } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { xlog("#######WITHINDLG:ACK transaction found $si:$sp\n"); # non loose-route, but stateful ACK; # must be ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard. xlog("#######WITHINDLG:ACK no transaction found so discard $si:$sp\n"); exit; } } sl_send_reply("404","Not here"); } exit; } } Debug: Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:623]: parse_msg(): SIP Request: Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:625]: parse_msg(): method: <ACK> Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:627]: parse_msg(): uri: <sip:10.57.26.239:5060 ;transport=UDP> Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:629]: parse_msg(): version: <SIP/2.0> Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/parse_via.c:1284]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-15698-1-5>; state=16 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/parse_via.c:2672]: parse_via(): end of header reached, state=5 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:513]: parse_headers(): parse_headers: Via found, flags=2 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:515]: parse_headers(): parse_headers: this is the first via Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [receive.c:154]: receive_msg(): After parse_msg... Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [receive.c:197]: receive_msg(): preparing to run routing scripts... Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: sl [sl_funcs.c:412]: sl_filter_ACK(): DEBUG : sl_filter_ACK: to late to be a local ACK! Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/parse_addr_spec.c:176]: parse_to_param(): DEBUG: add_param: tag=14783SIPpTag018 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/parse_addr_spec.c:898]: parse_addr_spec(): end of header reached, state=29 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:190]: get_hdr_field(): DEBUG: get_hdr_field: <To> [57]; uri=[sip:service@10.57.26.220:5060] Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:192]: get_hdr_field(): DEBUG: to body [sut < sip:service@10.57.26.220:5060>] Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:170]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <1> <ACK> Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: sanity [mod_sanity.c:255]: w_sanity_check(): sanity checks result: 1 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: siputils [checks.c:106]: has_totag(): totag found Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) ERROR: <script>: #######WITHINDLG:hastotag 10.57.26.219:5060 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:204]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [parser/msg_parser.c:106]: get_hdr_field(): found end of header Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: rr [loose.c:113]: find_first_route(): No Route headers found Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: rr [loose.c:929]: loose_route(): There is no Route HF Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: tm [t_lookup.c:1072]: t_check_msg(): DEBUG: t_check_msg: msg id=9 global id=8 T start=(nil) Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: tm [t_lookup.c:527]: t_lookup_request(): t_lookup_request: start searching: hash=53919, isACK=1 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: tm [t_lookup.c:485]: matching_3261(): DEBUG: RFC3261 transaction matching failed Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: tm [t_lookup.c:709]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: tm [t_lookup.c:1141]: t_check_msg(): DEBUG: t_check_msg: msg id=9 global id=9 T end=(nil) Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) ERROR: <script>: #######WITHINDLG:ACK no transaction found so discard 10.57.26.219:5060 Dec 15 15:12:02 DNS-enum-8Gb-3 kamailio[6864]: 1(6883) DEBUG: <core> [usr_avp.c:643]: destroy_avp_list(): DEBUG:destroy_avp_list: destroying list (nil)

3 3

Siremis installation with postgresql.
by José Seabra 16 Dec '16

16 Dec '16

Hi there, I'm facing some issues trying to install siremis with postgresql. The file Config.xml has the correct configuration for postgresql driver but It seems that during installation phase siremis can't create database structure for postgresql. Anyone have installed siremis with postgresql, if yes can you share the steps needed for this installation. -- Cumprimentos José Seabra

4 6

kamcli - transferring the github repository to kamailio organization
by Daniel-Constantin Mierla 16 Dec '16

16 Dec '16

Hello, kamcli (kamailio command line interface tool written in python) got a consistent set of features trying to become in pair with kamctl. It is currently hosted by asipto organization on github, but I am considering to transfer (move) it to kamailio organization in order to manage easier the access for kamailio devs as well as increase its exposure to community. It interacts with kamailio using rpc commands via jsonrpcs module. You can read more about what it can do at: - https://github.com/asipto/kamcli The commands are implemented in a plugin-like fashion -- see the ones done so far: - https://github.com/asipto/kamcli/tree/master/kamcli/commands Given that it can handle better the jsonrpc responses, provides more options to format the output and the help is embedded in each command, I think it is a good candidate to replace the old shell(bash)-based kamctl. Probably we can adjust kamctl for rpc when mi code will be removed, but the options to build and extend in long term a good cli for kamailio are quite limited in shell. For the sake of an example, here is the implementation of 'ps' command, handling the json rpc to display a very compact list of kamailio processes: - https://github.com/asipto/kamcli/blob/master/kamcli/commands/cmd_ps.py Dealing with database records is also rather trivial: - https://github.com/asipto/kamcli/blob/master/kamcli/commands/cmd_group.py Anyone with comments/suggestions/alternatives? Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

1 1

Forwarding Emergency calls based on caller IP address
by João Vitor Torres 15 Dec '16

15 Dec '16

Hello all, I am looking for a solution to forward emergency calls inside an enterprise network. Is there any way to forward calls based on the IP address of caller device? The requirement is that calls to a unique emergency number must be forwarded to specific local numbers in each building, and the IP address of the caller is the criteria for choosing the destination. A table maps IP nets to specific local numbers. The SIP contact header has the caller IP address, but I am new to Kamailio and I am not sure where to start. Any suggestions? Thanks in advance. João Vitor.

2 1

Re: [SR-Users] htable lock - unlock
by Daniel-Constantin Mierla 15 Dec '16

15 Dec '16

Hello, the log message you refer to is inside the if block: if ($sht(pht=>$ci)!=$null) { ... } but you don't set the item in memory, no assignment like: $sht(pht=>$ci) = ...; The sht_lock(pht=>$ci) is just locking the slot where the value of $sht(pht=>$ci) will be stored, but don't store any valye for that variable. Cheers, Daniel On 15/12/2016 11:46, Marino Mileti wrote: > > Hi guys, > > > > I'm going out of mind J > > I've this piece of script inside an "on_reply_route" that captures > only 200OK of INVITE (i would like to save in ht the contact of who > has answered a call) > > > > sht_lock("pht=>$ci"); > > if ($sht(pht=>$ci)!=$null){ > > xlog("L_INFO","Before assignment.."); > > $sht(pht=>$ci)=$var(myvar); > > xlog("L_INFO","Assigned done..."); > > sht_unlock("pht=>$ci"); > > } > > else{ > > xlog("L_INFO","Skipping.."); > > } > > sht_unlock("pht=>$(hdr(Call-ID))"); > > > > The situation is that I never see the "Assigned done" message in log > file... > > > > Is this the right use of sht_lock/sht_unlock method? > > > > /Marino Maria Mileti/// > > /marino.mileti(a)alice.it <mailto:marino.mileti@alice.it>/ > > / / > > /cid:006a01cb6b0e$67eecdae$_CDOSYS2.0//Reduce your energy consumption > and keep polar bears on ice!/ > > > > > > ------------------------------------------------------------------------ > Avast logo > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > Questa e-mail è stata controllata per individuare virus con Avast > antivirus. > www.avast.com > <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…> > > > > > > _______________________________________________ > 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 www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com

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Updating to SIREMIS 4.3
by Kevin Greene 15 Dec '16

15 Dec '16

Hello fellow SIP people: I upgraded from SIREMIS 4.2 to 4.3 for my Kamailio system and I now get the error message shown below. Any ideas on how to solve it? I am assuming there was a change on how SIREMIS did a date query and my version of Linux (6.8) or my PHP does not support it as expected. Any help would be useful. Thanks. Kevin [{"target":"ERROR","content":"\n[2016-12-14 22:59:55 (GMT)] An exception occurred while executing this script: \nError message: #2, date(): It is not safe to rely on the system's timezone settings. You are *required* to use the date.timezone setting or the date_default_timezone_set() function. In case you used any of those methods and you are still getting this warning, you most likely misspelled the timezone identifier. We selected 'UTC' for 'UTC\/0.0\/no DST' instead<\/font> \nScript name and line number of error: \/var\/www\/siremis-4.3.0\/openbiz\/bin\/service\/logService.php:160<\/font> \n *function:<\/b> errorHandler ( 2, \"date(): It is not safe to rely on the system's timezone settings...\", \"\/var\/www\/siremis-4.3.0\/openbiz\/bin\/service\/logService.php\", 160, Array(8) ) @ \/var\/www\/siremis-4.3.0\/openbiz\/bin\/sysheader.inc 117\n *function:<\/b> userErrorHandler ( 2, \"date(): It is not safe to rely on the system's timezone settings...\", \"\/var\/www\/siremis-4.3.0\/openbiz\/bin\/service\/logService.php\", 160, Array(8) ) @ \n *function:<\/b> date ( \"m\/d\/Y\" ) @ \/var\/www\/siremis-4.3.0\/openbiz\/bin\/service\/logService.php 160\n The "date" command returns: $ date Thu Dec 15 01:13:41 UTC 2016 The /etc/localtime entry is: $ more /etc/localtime TZif2 UTC $ which php /usr/bin/php $ php -version PHP 5.3.3 (cli) (built: Aug 11 2016 13:38:15) Copyright (c) 1997-2010 The PHP Group Zend Engine v2.3.0, Copyright (c) 1998-2010 Zend Technologies PHP I am using is the latest patched version I have. ***

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