- sr-users - kamailio.org

Anybody using 2 instances with topos communicating with each other?
by Daniel Tryba 01 Jun '18

01 Jun '18

Scenario: endpoint (bob) <-> kamailio proxy (5.1.3) <-> kamailio backend (5.1.2) Both kamailio instances are running the topos module (mysql backend). Call setup is fine, but forwarding BYEs fail. With a call from endpoint to kamailio, topos on the proxy scrambles Contact to kamailio backend. But in responses from backend contact remains untouched. In the ACK to the 200, topos on the proxy does nothing. If the caller hangsup, the BYE will loop internally till max hops is exceeded (483 which will be send to caller eventually). When the backend initiates the call to the endpoint, topos appears to work, but the BYE from the endpoint on hangup get a 404 Not here from the proxy (route[WITHINDLG]). Disabling topos on the proxy will result in a functioning proxy (config attached). Is there something wrong with my config? Or is this an unforseen scenario that just doesn't work with topos? * Endpoint to backend scenario endpoint:5060 -> proxy:5060 INVITE sip:callee@sip.pocos.nl;transport=tcp SIP/2.0 Via: SIP/2.0/UDP endpoint:5060;rport;branch=... Max-Forwards: 70 Route: <sip:proxy:5060;lr> Contact: <sip:bob@endpoint:5060;ob> proxy:5060 -> backend:5060 INVITE sip:callee@sip.itco.nl;transport=tcp SIP/2.0 Via: SIP/2.0/UDP proxy;branch=... CSeq: 29848 INVITE Route: <sip:proxy:5060;lr> Contact: <sip:btpsh-5af5972d-bf5e-2@proxy> backend:5060 -> proxy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy;branch=... Contact: <sip:atpsh-5af19bf0-9f94-7@backend> proxy:5060 -> endpoint:5060 SIP/2.0 200 OK Contact: <sip:atpsh-5af19bf0-9f94-7@backend> Via: SIP/2.0/UDP endpoint:5060;received=endpoint;rport=5060;branch=... Record-Route: <sip:proxy;lr> endpoint:5060 -> proxy:5060 ACK sip:atpsh-5af19bf0-9f94-7@backend SIP/2.0 Via: SIP/2.0/UDP endpoint:5060;rport;branch=... Route: <sip:proxy;lr> proxy:5060 -> backend:5060 ACK sip:atpsh-5af19bf0-9f94-7@backend SIP/2.0 Via: SIP/2.0/UDP proxy;branch=... Via: SIP/2.0/UDP endpoint:5060;received=endpoint;rport=5060;branch=... endpoint:5060 -> proxy:5060 BYE sip:atpsh-5af19bf0-9f94-7@backend SIP/2.0 Via: SIP/2.0/UDP endpoint:5060;rport;branch=... Max-Forwards: 70 Route: <sip:proxy;lr> proxy:5060 -> proxy:5060 BYE sip:atpsh-5af19bf0-9f94-7@backend SIP/2.0 Via: SIP/2.0/UDP proxy;branch=... Via: SIP/2.0/UDP endpoint:5060;received=endpoint;rport=5060;branch=... Max-Forwards: 69 Route: <sip:proxy;lr> proxy:5060 -> proxy:5060 BYE sip:atpsh-5af19bf0-9f94-7@backend SIP/2.0 Via: SIP/2.0/UDP proxy;branch=... Via: SIP/2.0/UDP proxy;rport=5060;branch=... Via: SIP/2.0/UDP endpoint:5060;received=endpoint;rport=5060;branch=... Max-Forwards: 68 Route: <sip:proxy;lr> [not going to mention the 50+ BYEs adding Vias and decrementing Max-Forwards. * Backend to endpoint scenario backend:5060 -> proxy:5060 INVITE sip:bob@endpoint:5060;ob SIP/2.0 Via: SIP/2.0/UDP backend;branch=... Route: <sip:loadbalancer@proxy;lr;received=sip:endpoint:5060> Contact: <sip:btpsh-5af07a27-72b-502c1@backend> proxy:5060 -> endpoint:5060 INVITE sip:bob@endpoint:5060;ob SIP/2.0 Via: SIP/2.0/UDP proxy;branch=... Contact: <sip:btpsh-5af5972d-bf5d-5@proxy> endpoint:5060 -> proxy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy;received=proxy;branch=... Contact: <sip:bob@endpoint:5060;ob> proxy:5060 -> backend:5060 SIP/2.0 200 OK Contact: <sip:bob@endpoint:5060;ob;alias=endpoint~5060~1> Via: SIP/2.0/UDP backend;rport=5060;branch=... Record-Route: <sip:proxy;lr> endpoint:5060 -> proxy:5060 BYE sip:btpsh-5af5972d-bf5d-5@proxy SIP/2.0 Via: SIP/2.0/UDP endpoint:5060;rport;branch=... proxy:5060 -> endpoint:5060 SIP/2.0 404 Not here Via: SIP/2.0/UDP endpoint:5060;rport=5060;branch=...

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Dispatcher module ERROR
by Aleksandar Sosic 01 Jun '18

01 Jun '18

Hi everyone, We have an error using the dispatcher module. In the logs: ```0(107) ERROR: dispatcher [dispatcher.c:815]: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled``` We are running 5.2.0~dev5+0~20180523010152.1118+xenial The configuration file parts regarding dispatcher are as follows: ``` loadmodule "dispatcher.so" ... # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/tmp/dispatcher.list") # Dispatcher: Enable Failover-Support modparam("dispatcher", "flags", 2) # Dispatcher: Overwrite Destination address, if required. modparam("dispatcher", "force_dst", 1) # AVP's required for Fail-Over-Support: modparam("dispatcher", "dst_avp", "$avp(DISPATCHER_DST_AVP)") modparam("dispatcher", "grp_avp", "$avp(DISPATCHER_GRP_AVP)") modparam("dispatcher", "cnt_avp", "$avp(DISPATCHER_CNT_AVP)") # Try to recover disabled destinations every 15 seconds. modparam("dispatcher", "ds_ping_interval", 15) # Actively query the gateways: modparam("dispatcher", "ds_probing_mode",0)# ----- dispatcher params ----- modparam("dispatcher", "list_file", "/tmp/dispatcher.list") # Dispatcher: Enable Failover-Support modparam("dispatcher", "flags", 2) # Dispatcher: Overwrite Destination address, if required. modparam("dispatcher", "force_dst", 1) # AVP's required for Fail-Over-Support: modparam("dispatcher", "dst_avp", "$avp(DISPATCHER_DST_AVP)") modparam("dispatcher", "grp_avp", "$avp(DISPATCHER_GRP_AVP)") modparam("dispatcher", "cnt_avp", "$avp(DISPATCHER_CNT_AVP)") # Try to recover disabled destinations every 15 seconds. modparam("dispatcher", "ds_ping_interval", 15) # Actively query the gateways: modparam("dispatcher", "ds_probing_mode",0)``` What are we doing wrong? What are we missing? Thanks, -- Aleksandar Sosic linkedin.com/in/alexsosic/ mail: alex.sosic(a)evosip.cloud

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Authenticate against outbound carrier
by Krzysztof Grobelak 01 Jun '18

01 Jun '18

Hello List, I have a working setup with a kamailio + freeswitch where i can register extension and make succesfull calls between them. I added now a "external" path for outbound call to which i can successfully route using the dialplan + dispatcher combo. My problem is that my outbound path requires authentication and i'm struggling to find a way to successfully authenticate with the remote end. I configured the UAC module to register against the remote end with a set of credentials but in the SIP trace i can see that the kamailio simply uses the local user credentials instead of the external ones. I believe that this is because the 401 Unauthorized from the carrier gets proxied to the extension which responds with its own credentials. Can anybody suggest how can i stop forwarding them in this case to extension and instead use the UAC to reply with different set of credentials? Regards, Krzysztof The config: request_route { ... if($rU=~"[0-9]{5,}"){ xlog("L_INFO","Looking up route for $rU\n"); dp_match("1", "$rU"); $var(dsid) = $(var(attrs){s.int}); if(!ds_select_dst("$var(dsid)", "4")) { send_reply("404", "No destination"); exit; } if(!ds_select_domain("$var(dsid)", "4")) { send_reply("404", "No destination"); exit; } xlog("L_INFO", "Looking up $fn in database\n"); $avp(s:auth_realm_avp) = "realm.com"; $avp(s:auth_username_avp) = "username"; $avp(s:auth_password_avp) = "password"; if(uac_reg_request_to("$fn", 1)) { xlog("L_INFO", "Found remote user [$rU] on [$rd] via [$du]"); t_on_failure("REMOTE_AUTH"); xlog("L_INFO", "going to <$ru> via <$du>\n"); route(RELAY); } exit; } failure_route[REMOTE_AUTH] { if ($T_reply_code == 401 or $T_reply_code == 407) { xlog("L_INFO", "Remote asked for authentication"); uac_auth(); } } -- PGP: 9E5D E635 5A03 485C 2E00 4530 EB83 920A 9E21 FBA8

1 0

Acc module and PostgreSQL
by Javohir Maxmarajabov 01 Jun '18

01 Jun '18

Is it possible to enable accounting module with PostgreSQL? All I found is documentation and examples that showing accounting module is used with MySQL module. -- Javokhir M.M. ________________________________________________ Disclaimer: The information in this e-mail is confidential. If you are not addressed recipient then please return and delete this e-mail from your system. Unauthorised use of or disclose the contents of this e-mail may be unlawful.

2 1

Strange behavior with $shtinc
by Brooks Bridges 01 Jun '18

01 Jun '18

Let me preface this with the statement that as best I can tell from docs and code, this function is supposed to be 100% atomic and this shouldn't be able to happen. I also have not found anything in the module's github commit logs that would indicate this behavior has changed from previous versions. That being said, I appear to have run into an oddity where intermittently I'm getting 2 slots being put in the same entry, which results in trying to parse the output for graphing and reporting not working quite right. The mechanism I'm using is a very simple counter mechanism to record statistics about calls. As an example, I am using the following mechanism: $var(z) = $shtinc(callstats=>$avp(realm),cps_exceeded); or $var(z) = $shtinc(callstats=>$avp(realm),sessions_exceeded); This typically results in something like this: { entry: 4335 size: 1 slot: { { name: realm1,total_requests value: 2365034 type: int } } } { entry: 4532 size: 1 slot: { { name: realm2,cps_exceeded value: 30 type: int } } } And so on... However, one or more of them sometimes end up like this: { entry: 4646 size: 2 slot: { { name: realm1,total_requests value: 15958026 type: int } { name: realm2,cps_exceeded value: 6432103 type: int } } } Thanks in advance for your help and/or suggestions! Brooks Bridges | Engineering Manager O1 Communications 4359 Town Center Boulevard, Suite 217 El Dorado Hills, California 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

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Authentication against outbound carrier trunk
by Krzysztof Grobelak 31 May '18

31 May '18

Hello List, I have a working setup with a kamailio + freeswitch where i can register extension and make succesfull calls between them. I added now a "external" path for outbound call to which i can successfully route using the dialplan + dispatcher combo. My problem is that my outbound path requires authentication and i'm struggling to find a way to successfully authenticate with the remote end. I configured the UAC module to register against the remote end with a set of credentials but in the SIP trace i can see that the kamailio simply uses the local user credentials instead of the external ones. I believe that this is because the 401 Unauthorized from the carrier gets proxied to the extension which responds with its own credentials. Can anybody suggest how can i stop forwarding them in this case to extension and instead use the UAC to reply with different set of credentials? Regards, Krzysztof The config: request_route { ... if($rU=~"[0-9]{5,}"){ xlog("L_INFO","Looking up route for $rU\n"); dp_match("1", "$rU"); $var(dsid) = $(var(attrs){s.int}); if(!ds_select_dst("$var(dsid)", "4")) { send_reply("404", "No destination"); exit; } if(!ds_select_domain("$var(dsid)", "4")) { send_reply("404", "No destination"); exit; } xlog("L_INFO", "Looking up $fn in database\n"); $avp(s:auth_realm_avp) = "realm.com"; $avp(s:auth_username_avp) = "username"; $avp(s:auth_password_avp) = "password"; if(uac_reg_request_to("$fn", 1)) { xlog("L_INFO", "Found remote user [$rU] on [$rd] via [$du]"); t_on_failure("REMOTE_AUTH"); xlog("L_INFO", "going to <$ru> via <$du>\n"); route(RELAY); } exit; } failure_route[REMOTE_AUTH] { if ($T_reply_code == 401 or $T_reply_code == 407) { xlog("L_INFO", "Remote asked for authentication"); uac_auth(); } } -- PGP: 9E5D E635 5A03 485C 2E00 4530 EB83 920A 9E21 FBA8

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error in tls configuration in kamailio 5.1.3
by Narasimhaiah, Vinod Maralenahalli 31 May '18

31 May '18

Hi, I upgraded kamailio 4.4.2 to kamailio5.1.3 and i am getting this error *"Unable to load certificate file '/etc/kamailio/kamailio-selfsigned.pem",* kamilio 4.4.2 does not need the ssl file, but kamailio 5.1.3 need ssl file, what changes required in kamailio 5.1.3 to run with tls. Please help me in resolving this issue. Thank you vinod.M.N -- To download the app, click: iOS <http://goo.gl/ZJx3y1> | Android <http://goo.gl/G1ehjZ> -- ***This transmission contains information that is confidential and proprietary of Kirusa Inc.,its affiliates and/or its partners and customers. This information is intended to be shared only with the recipient of this e-mail who is bound by confidentiality. If you received this e-mail in error, please delete the e-mail and notify the sender by return e-mail.***

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Upgrade issue with old Kamailio 3.0.3 in table rls_presentity
by Kristijan Vrban 31 May '18

31 May '18

Hi, i try a upgrade from old 3.0.3 Kamailio mysql db. Got stuck with update the structure : ALTER TABLE rls_presentity MODIFY callid VARCHAR(255) NOT NULL;ALTER TABLE rls_presentity MODIFY contact VARCHAR(128) NOT NULL; found on: https://www.kamailio.org/wiki/install/upgrade/3.1.x-to-3.2.0 But this two table does not exist? See also table definition at: https://kamailio.org/docs/db-tables/kamailio-db-3.1.x.html#GEN-DB-RLS-PRESE… I reckon it is safe to skip this to tables? Kristijan

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Register on kamailio.org wiki page not working
by Kristijan Vrban 31 May '18

31 May '18

Hi, https://www.kamailio.org/wiki/start?do=register Is not working. You get a empty page here. Kristijan

2 2

Sharing static variables in external library across child proceses
by User 721 31 May '18

31 May '18

Hi, I am adding a kamailio module that references an external C library that maintains a hashtable in global scope. I am wondering what practice I should follow so that: - This hashtable (which I don't control directly) is shared by all child processes - The hashtable is changed by one child process at a time Thanks in advance

2 1

PIKE module
by Robert Barrera 31 May '18

31 May '18

Good morning! We operate a VoIP platform and are interested in integrating the PIKE module. Is it possible for someone to assist us to discuss how this can be done and our options? Thank you logo_nop Robert Barrera 305-663-8989 305-600-4786 fax info(a)myrealmobile.com Complete Business PBX phone systems and Nationwide wireless! REAL Mobile does it!

2 1

sip_msg structure and memory free
by Konstantin Polyakov 31 May '18

31 May '18

Hello All, There is a structure sip_msg which is passed into a module and represents a received sip message. It contains number of pointers to hdr_field members. In its turn hdr_field contains member void* parsed In the sanity module we assign to parsed member a pointer to newly allocated memory and we don't free that memory at the end. Do we need to free explicitly or this memory will be deallocated later together with sip_msg ? In other words is it memory leak or not? Best regards, Konstantin

2 1

General Kamailio with Asterisk (or another PBX) Opinion
by Wilkins, Steve 30 May '18

30 May '18

HI All, I am curious, for those of you who use Kamailio with Asterisk or another PBX. Who forwards Registrations to Asterisk or PBX, and who lets Kamailio maintain Registrations? Thanks All!

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send CANCEL to caller after hitting fr_inv_timeout
by Sergiu Pojoga 29 May '18

29 May '18

TM's noisy_ctimer=1 seems to take care of sending a CANCEL to the callee, but it eludes me which tm param makes a CANCEL to be sent to caller as well? Caller gets a '408 Request Timeout' only, which in my case doesn't effectively terminate a call for the calling party. Merci.

1 0

INFO: stun_parse_header: incomplete header of STUN message
by Pan Christensen 29 May '18

29 May '18

Hello! After proxying a REGISTER from a WebRTC client to our SIP backend, the backend starts sending keepalives. Every 30 seconds I get a log entry in Kamailio saying: "INFO: stun [kam_stun.c:169]: stun_parse_header(): INFO: stun_parse_header: incomplete header of STUN message". Here's one such packet: No. Time Source Destination Protocol Length Info 1 0.000000 xxx.xxx.xxx.xxx yyy.yyy.yyy.yyy UDP 60 5060 ? 15060 Len=4 Frame 1: 60 bytes on wire (480 bits), 60 bytes captured (480 bits) Ethernet II, Src: JuniperN_4f:3f:f0 (40:a6:77:4f:3f:f0), Dst: 02:81:64:58:cf:d1 (02:81:64:58:cf:d1) Internet Protocol Version 4, Src: xxx.xxx.xxx.xxx, Dst: yyy.yyy.yyy.yyy 0100 .... = Version: 4 .... 0101 = Header Length: 20 bytes (5) Differentiated Services Field: 0x00 (DSCP: CS0, ECN: Not-ECT) Total Length: 32 Identification: 0x5bf5 (23541) Flags: 0x4000, Don't fragment Time to live: 56 Protocol: UDP (17) Header checksum: 0x2fc3 [validation disabled] [Header checksum status: Unverified] Source: xxx.xxx.xxx.xxx Destination: yyy.yyy.yyy.yyy User Datagram Protocol, Src Port: 5060, Dst Port: 15060 Source Port: 5060 Destination Port: 15060 Length: 12 Checksum: 0xfa28 [unverified] [Checksum Status: Unverified] [Stream index: 0] Data (4 bytes) 0000 00 00 00 00 .... Data: 00000000 [Length: 4] I found this in kam_stun.c: static int stun_parse_header(struct stun_msg* req, USHORT_T* error_code) { if (sizeof(req->hdr) > req->msg.buf.len) { /* the received message does not contain whole header */ LOG(L_INFO, "INFO: stun_parse_header: incomplete header of STUN message\n"); /* Any better solution? IMHO it's not possible to send error response * because the transaction ID is not available. */ return FATAL_ERROR; } ... Could someone please explain what's wrong? Med vennlig hilsen Pan B. Christensen Utvikler Phonect AS Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway E-post: pan.christensen(a)phonect.no<mailto:pan.christensen@phonect.no> Mobil: 41 88 88 00 [cid:image007.png@01D3A0E8.376921D0] <http://www.phonect.no/> [facebook_2]<https://www.facebook.com/phonectno> [LinkedIn_logo_initials (1)] <https://www.linkedin.com/company/44983?trk=tyah&trkInfo=clickedVertical%3Ac…>

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Dialplan module for load balancing
by igor.potjevlesch＠gmail.com 29 May '18

29 May '18

Hello, I'm maybe confused on how to use dialplan module. I have a dialplan setup for a long time now and I'd like to add a new rule with the same match_exp than an existing one but with a different repl_exp. If I set the same priority, the latest entry is used systematically. If I set priority +1, the new rule is never used. The goal is having two rules and alternatively reply with one or the other repl_exp. Is it possible? Regards, Igor.

1 0

Pkg memory stats
by Konstantin Polyakov 29 May '18

29 May '18

Hello, Could somebody explain what do numbers mean for command "kamcmd mod.stats all pkg"? Is there somewhere documentation about that? I can see the following information. " Module: xlog { mod_init(213): 4104 xlog_fixup_helper(533): 3176 xdbg_fixup_helper(499): 632 Total: 7912 } " I guess that numbers after colon are bytes. But what do numbers in (213) mean? When I use command "kamcmd pkg.stats", I got absolutely different numbers which don't match previous statistics? What is wrong? What is the best way to analyze pkg memory usage? Thank you Best regards Konstantin

1 0

Realtime Extensions SQLITE
by Benjamin Marty 29 May '18

29 May '18

I compiled a fresh Asterisk 13.18 certified with res_config_sqlite. I configured following: extconfig.conf [settings] sippeers => sqlite,asterisk,ast_config res_config_sqlite.conf [general] dbfile => /home/myuser/database.db On Bootup it gives the following error: [May 29 17:07:14] WARNING[32444]: res_config_sqlite.c:1206 realtime_multi_handler: no such table: ast_config Is the sqlite module deprecated or is my configuration wrong? Couldn't really find any documentation on that topic. The table ast_config is definitely present in the sqlite file.

2 1

Serial forking
by igor.potjevlesch＠gmail.com 29 May '18

29 May '18

Hello, I know that the default behaviour of two aliases in kamailio.aliases result in parallel forking. I'm wondering if its possible to do serial forking instead? For example, in case that the first INVITE has no 180/183, or neither 100 Trying, then send the INVITE to the second entry? Regards, Igor.

2 2

Switching form UDP to TCP causes authentication errors
by Wilkins, Steve 29 May '18

29 May '18

Hello All, When I switched from UDP to TCP I started getting Authentication Errors, Asterisk responds to an INVITE via Kamailio with a '401 Unauthorized', but Kamailio does nothing with it. Processing just stops near WITH_BLOCK401407. Shouldn't the 401 be relayed so a new INVITE can be sent? Thank you.

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ims_isc: 3rd party Registration (S-CSCF) and P-Associated-Uri header
by Valentin Christoph 29 May '18

29 May '18

Hi, The function build_p_associated_uri() in module ims_isc filters for TEL URIs. Why are not ALL unbarred IMPUs added, but only TEL URIs? Is this specified in some 3GPP spec? Thanks, Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

1 0

jsonrpc http interface cant recognize method
by Yuriy Gorlichenko 29 May '18

29 May '18

Hi i trying to get htable.dump mytable via jsonrpc request {jsonrpc: "2.0", method: "htable.dump mytable"} { "jsonrpc": "2.0", "error": { "code": 500, "message": "Method Not Found" } } But via kamcmd it wroks perfect Aslo i see that if i will make request withont param (core.version for example) IT works fime How to make request with param?

1 1

Carrierroute Wholesale termination issue
by riaz hasan 29 May '18

29 May '18

Hello, I am facing an issue with kamailio carrierroute.I have a setup where multiple asterisk is behind a kamailio server.I am able to dispatch the load by using carrierroute. Its working perfectly for retail users.But problem is its not working for wholesale users.I am attaching the dump for wholesale client. here is a little descrption for the shark:- 192.151.149.102 is the wholesale client ip. 46.166.139.60:4848 is kamailio server. 46.166.139.60:3628 is asterisk server. when 192.151.149.102 sends invite to 46.166.139.60:4848 [Kamailio] we are getting 401. [dump2.pcap] when 192.151.149.102 sends invite to 46.166.139.60:3628 [Direct Asterisk ] we are getting 200. [dump.pcap] here is the code i am using for load balancing:- route[Carrer] { #!ifdef WITH_CARRIERROUTE if (!cr_route("default", "default", "$rU", "$rU", "call_id")) { xlog("L_ERR", "cr_route failed\n"); exit; } #!endif $avp(s:host)= $rd+":"+$rp; t_on_failure("1"); if (!t_relay()) { sl_reply_error(); }; } failure_route[1] { revert_uri(); #!ifdef WITH_CARRIERROUTE if (!cr_next_domain("default", "default", "$rU", "$avp(s:host)", "$T_reply_code", "$avp(s:domain)")) { xlog("L_ERR", "cr_next_domain failed\n"); exit; } if (!cr_route("default", "default", "$rU", "$rU", "call_id")) { xlog("L_ERR", "cr_route failed\n"); } t_on_failure("1"); route(NASIR); exit; #!endif $avp(s:host)= $rd+":"+$rp; if (!t_relay()) { xlog("L_ERR", "t_relay failed-custom\n"); exit; }; }

1 0

S-SCSF not forwarding SIP packets to AS
by John M 29 May '18

29 May '18

Hi All, My setup consists of Kamailio P-CSCF,I-SCSF , S-CSCF and OpenIMS HSS nodes and I have configured application server details and trigger point on HSS. I am using Boghe IMS client SIP validation. Whenever the client sends a REGISTER, messages are not reaching AS. In S-SCSF log files, I am seeing " isc_match_filter_reg(): No REGISTER match found in filter criteria" and " isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group" error message. Kmailio version: version: kamailio 5.1.3 (x86_64/linux) HSS XML The UserData XML document which is sent to the S-CSCF: <?xml version="1.0" encoding="UTF-8"?><IMSSubscription><PrivateID> alice(a)ims.mnc001.mcc001.3gppnetwork.org </PrivateID><ServiceProfile><PublicIdentity><Identity> sip:alice@ims.mnc001.mcc001.3gppnetwork.org </Identity><Extension><IdentityType>0</IdentityType></Extension></PublicIdentity><InitialFilterCriteria><Priority>0</Priority><TriggerPoint><ConditionTypeCNF>0</ConditionTypeCNF><SPT><ConditionNegated>0</ConditionNegated><Group>0</Group><Method>REGISTER</Method><Extension><RegistrationType>0</RegistrationType></Extension></SPT></TriggerPoint><ApplicationServer><ServerName>sip:192.168.20.10:5060 ;transport=TCP</ServerName><DefaultHandling>0</DefaultHandling></ApplicationServer></InitialFilterCriteria></ServiceProfile></IMSSubscription> SCSF Log-1 0(1541) DEBUG: ims_isc [ims_isc_mod.c:428]: isc_match_filter_reg(): Checking triggers 0(1541) DEBUG: ims_isc [ims_isc_mod.c:432]: isc_match_filter_reg(): Starting triggering 0(1541) DEBUG: ims_isc [ims_isc_mod.c:446]: isc_match_filter_reg(): Orig User before clean: <sip:alice@ims.mnc001.mcc001.3gppnetwork.org> [1] 0(1541) DEBUG: ims_isc [ims_isc_mod.c:448]: isc_match_filter_reg(): Orig User after clean: <sip:alice@ims.mnc001.mcc001.3gppnetwork.org> [1] 0(1541) DEBUG: ims_isc [checker.c:416]: isc_checker_find(): isc_checker_find: starting search 0(1541) DEBUG: ims_isc [checker.c:438]: isc_checker_find(): isc_checker_find(): got a r_public for the user sip:alice@ims.mnc001.mcc001.3gppnetwork.org 0(1541) DEBUG: ims_isc [checker.c:454]: isc_checker_find(): About to try p->s->service_profiles_cnt!! #profiles is 1 0(1541) DEBUG: ims_isc [checker.c:456]: isc_checker_find(): About to try p->s->service_profiles[a].filter_criterai_cnt 0(1541) DEBUG: ims_isc [checker.c:467]: isc_checker_find(): DEBUG ISC: SECOND TIME About to try p->s->service_profiles_cnt!! 0(1541) DEBUG: ims_isc [checker.c:471]: isc_checker_find(): DEBUG ISC : About to try p->s->service_profiles 0(1541) DEBUG: ims_isc [checker.c:474]: isc_checker_find(): DEBUG ISC : About to try public identities 0(1541) DEBUG: ims_isc [checker.c:477]: isc_checker_find(): DEBUG ISC : About to try WPSI 0(1541) DEBUG: ims_isc [checker.c:319]: isc_check_filter_criteria(): ifc_checker_trigger: Starting expression check: 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == PUBLISH ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method REGISTER 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == PUBLISH ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method REGISTER 0(1541) DEBUG: ims_isc [checker.c:333]: isc_check_filter_criteria(): ifc_checker_trigger: - group 1 => 0. 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == SUBSCRIBE ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method REGISTER 0(1541) DEBUG: ims_isc [checker.c:333]: isc_check_filter_criteria(): ifc_checker_trigger: - group 2 => 0. 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == SUBSCRIBE ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method REGISTER 0(1541) DEBUG: ims_isc [checker.c:333]: isc_check_filter_criteria(): ifc_checker_trigger: - group 3 => 0. 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:348]: isc_check_filter_criteria(): ifc_checker_trigger: Check finished => 0 0(1541) DEBUG: ims_isc [ims_isc_mod.c:460]: isc_match_filter_reg(): No REGISTER match found in filter criteria SCSF Log-2 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == PUBLISH ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method SUBSCRIBE 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == PUBLISH ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method SUBSCRIBE 0(1541) DEBUG: ims_isc [checker.c:333]: isc_check_filter_criteria(): ifc_checker_trigger: - group 1 => 0. 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == SUBSCRIBE ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method SUBSCRIBE 0(1541) DEBUG: ims_isc [checker.c:333]: isc_check_filter_criteria(): ifc_checker_trigger: - group 2 => 1. 0(1541) DEBUG: ims_isc [checker.c:252]: isc_check_spt(): ifc_check_spt: SPT type 3 -> Header[Event] %= .*presence.* ? 0(1541) DEBUG: ims_isc [checker.c:88]: isc_check_headers(): isc_check_headers: Looking for Header[Event(22)] .*presence.* 0(1541) DEBUG: ims_isc [checker.c:99]: isc_check_headers(): isc_check_headers: Found Header[Event(22)] message-summary 0(1541) DEBUG: ims_isc [checker.c:99]: isc_check_headers(): isc_check_headers: Found Header[Allow-Events(39)] refer, presence, presence.winfo, xcap-diff, conference 0(1541) DEBUG: ims_isc [checker.c:261]: isc_check_spt(): ifc_check_spt: SPT type 4 -> Session Case == 1 ? 0(1541) DEBUG: ims_isc [checker.c:263]: isc_check_spt(): ifc_check_spt: Found session_case 0 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == SUBSCRIBE ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method SUBSCRIBE 0(1541) DEBUG: ims_isc [checker.c:333]: isc_check_filter_criteria(): ifc_checker_trigger: - group 3 => 1. 0(1541) DEBUG: ims_isc [checker.c:252]: isc_check_spt(): ifc_check_spt: SPT type 3 -> Header[Event] %= .*presence.* ? 0(1541) DEBUG: ims_isc [checker.c:88]: isc_check_headers(): isc_check_headers: Looking for Header[Event(22)] .*presence.* 0(1541) DEBUG: ims_isc [checker.c:99]: isc_check_headers(): isc_check_headers: Found Header[Event(22)] message-summary 0(1541) DEBUG: ims_isc [checker.c:99]: isc_check_headers(): isc_check_headers: Found Header[Allow-Events(39)] refer, presence, presence.winfo, xcap-diff, conference 0(1541) DEBUG: ims_isc [checker.c:261]: isc_check_spt(): ifc_check_spt: SPT type 4 -> Session Case == 2 ? 0(1541) DEBUG: ims_isc [checker.c:263]: isc_check_spt(): ifc_check_spt: Found session_case 0 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:348]: isc_check_filter_criteria(): ifc_checker_trigger: Check finished => 0 Regards, Uts

1 0

Kamailio 5.1.x - Dialog event route not fired
by Loic Chabert 28 May '18

28 May '18

Hi list, I'm facing a strange behaviour with dialog module and stop event route. When a BYE request has been receved by Kamailio, no stop event route has been fired but dialog state go to 5. My config is following this guide: http://kb.asipto.com/kamailio:presence:k43-blf Using dialog_manage on RELAY route, as describe on this guide. If i set more debug information, i can see dialog has been found and dialog status incremented : 1(7) DEBUG: <core> [core/udp_server.c:491]: udp_rcv_loop(): received on udp socket: (106/100/818) [[BYE sip:172.16.74.201:5060;prxroute=1 SIP/2.0 0D 0A Record-Route: <sip:172.16.74.24;r2=on;lr=on;ftag=1;ng]] 1(7) DEBUG: <core> [core/parser/msg_parser.c:604]: parse_msg(): SIP Request: 1(7) DEBUG: <core> [core/parser/msg_parser.c:606]: parse_msg(): method: <BYE> 1(7) DEBUG: <core> [core/parser/msg_parser.c:608]: parse_msg(): uri: <sip:172.16.74.201:5060;prxroute=1> 1(7) DEBUG: <core> [core/parser/msg_parser.c:610]: parse_msg(): version: <SIP/2.0> 1(7) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKc2c.939fa953ea648059586a7f684fd56b5a.0>; state=16 1(7) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 1(7) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 1(7) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 1(7) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 235, <rport> = <5060>; state=6 1(7) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-69-1-10>; state=16 1(7) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 1(7) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=78 1(7) DEBUG: <core> [core/parser/msg_parser.c:505]: parse_headers(): this is the second via 1(7) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=f651e13a-0aac-40d5-b877-1c24497ff87c 1(7) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 1(7) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [87]; uri=[sip:+33975181257@sip.testinfratrunk.vox.fr] 1(7) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body [<sip:+33975181257@sip.testinfratrunk.vox.fr>] 1(7) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <3> <BYE> 1(7) DEBUG: <core> [core/receive.c:232]: receive_msg(): --- received sip message - request - call-id: [1-69(a)60.0.0.30] - cseq: [3 BYE] 1(7) DEBUG: <core> [core/receive.c:287]: receive_msg(): preparing to run routing scripts... 1(7) DEBUG: maxfwd [mf_funcs.c:74]: is_maxfwd_present(): value = 69 1(7) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=1 1(7) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 1(7) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=0 1(7) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 1(7) DEBUG: sanity [sanity_mod.c:256]: w_sanity_check(): sanity checks result: 1 1(7) NOTICE: <script>: {1 3 BYE 1-69(a)60.0.0.30} Initialisation des variables. 1(7) DEBUG: tm [t_lookup.c:1018]: t_check_msg(): msg (0x7ffa420727d8) id=2 global id=1 T start=(nil) 1(7) DEBUG: tm [t_lookup.c:476]: t_lookup_request(): start searching: hash=3116, isACK=0 1(7) DEBUG: tm [t_lookup.c:435]: matching_3261(): RFC3261 transaction matching failed 1(7) DEBUG: tm [t_lookup.c:659]: t_lookup_request(): no transaction found 1(7) DEBUG: tm [t_lookup.c:1087]: t_check_msg(): msg (0x7ffa420727d8) id=2 global id=2 T end=(nil) 1(7) DEBUG: siputils [checks.c:123]: has_totag(): totag found 1(7) DEBUG: rr [loose.c:113]: find_first_route(): No Route headers found 1(7) DEBUG: rr [loose.c:960]: loose_route(): There is no Route HF 1(7) DEBUG: tm [t_lookup.c:1018]: t_check_msg(): msg (0x7ffa420727d8) id=2 global id=2 T start=(nil) 1(7) DEBUG: tm [t_lookup.c:1094]: t_check_msg(): T previously sought and not found for msg (0x7ffa420727d8) 1(7) DEBUG: dialog [dlg_hash.c:768]: internal_get_dlg(): ref dlg 0x7ffa327dd588 with 1 -> 3 1(7) DEBUG: dialog [dlg_hash.c:770]: internal_get_dlg(): dialog callid=' 1-69(a)60.0.0.30' found on entry 3063, dir=1 1(7) DEBUG: dialog [dlg_profile.c:536]: set_current_dialog(): setting current dialog [3063:9173] 1(7) DEBUG: dialog [dlg_hash.c:1151]: next_state_dlg(): dialog 0x7ffa327dd588 changed from state 4 to state 5, due event 7 (ref 3) 1(7) DEBUG: dialog [dlg_hash.c:932]: dlg_ref_helper(): ref op on 0x7ffa327dd588 with 1 from dlg_handlers.c:1389 1(7) DEBUG: dialog [dlg_hash.c:936]: dlg_ref_helper(): ref dlg 0x7ffa327dd588 with 1 -> 4 1(7) DEBUG: dialog [dlg_handlers.c:1400]: dlg_onroute(): BYE successfully processed Do you have any hints concerning this issue ? Is it a bug or a config part is missing ? Thanks in advance, Best regards.

2 1

Loop Detect - 482 Merged Request
by Amar Tinawi 28 May '18

28 May '18

Hello All i'm facing strange behavior with Zoiper Client only in an IMS Kamailio based system. in IMS call scenario (ims to ims) a lot of Loop Detect Msgs Appeared in the sip log. while other clients like Acrobits did not occurred i'm using TCP protocol and Wireshark shows a lot of (TCP Re-transmission and TCP ACKed unseen Segment) Any Suggestion Thanks !

1 0

Link problem between kamailio and MariaDB
by Ginhoux, Patrick 28 May '18

28 May '18

Hi, At the moment, I'm trying to replace MySQL by MariaDB on my Kamailio server (5.0.1) . In previous post I have been told that Kamailio supports MariaDB and there is no special module to load. This work is done on a new server running RHEL 7.4 and MariaDB 10.2.14 that is installed under /opt/mysql/na/10.2.14 folder. I encounter different problem: - Without changing anything in the kamailio script, kamailio failed to find the libmysqlclient.so.18 library. - So I thought that I could get rid of the above error by changing the MODULEPATH in the Kamailio-local.cfg as follows : MODULEPATH="/usr/lib64/kamailio/modules/:/opt/mysql/na/10.2.14/lib64/" But with this, I get 2 problem: The kamailio.log that was written in a /opt/log/Kamailio folder is no longer updated. I can only find the errors in the /var/log/messages file: May 23 11:19:19 vm-siprouter74 systemd: Starting SYSV: Kamailio is a fast, reliable and flexible SIP Server.... May 23 11:19:19 vm-siprouter74 kamailio: Not starting : invalid configuration file! May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.lex:1822]: pp_define_get(): ### returning define ID [DEFINE_LOG_FACILITY] value [LOG_LOCAL0] May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.lex:1822]: pp_define_get(): ### returning define ID [LOCAL_LISTEN] value [udp:129.227.82.109:5060] May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.lex:1822]: pp_define_get(): ### returning define ID [LOCAL_LISTEN_2] value [udp:127.0.0.1:5060] May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.lex:1822]: pp_define_get(): ### returning define ID [MODULEPATH] value ["/usr/lib64/kamailio/modules/:/opt/mysql/na/10.2.14/lib64/"] May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.y:1632]: yyparse(): loading modules under /usr/lib64/kamailio/modules/:/opt/mysql/na/10.2.14/lib64/ May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.y:1613]: yyparse(): loading module db_mysql.so May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/sr_module.c:575]: load_module(): trying to load </usr/lib64/kamailio/modules/db_mysql.so> May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) ERROR: <core> [core/sr_module.c:582]: load_module(): could not open module </usr/lib64/kamailio/modules/db_mysql.so>: libmysqlclient.so.18: cannot open shared object file: No such file or directory May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) CRITICAL: <core> [core/cfg.y:3401]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 191, column 12-24: failed to load module May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.y:1613]: yyparse(): loading module jsonrpcs.so May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/sr_module.c:575]: load_module(): trying to load </usr/lib64/kamailio/modules/jsonrpcs.so> To work around the problem we have created 2 symbol links: <<< ln -s /opt/mysql/na/10.2.14/lib64/libmysqlclient.so.18.0.0 /usr/lib64/libmysqlclient.so.18 >>> the libmysqlclient error <<< ln -s /myqdata/mysiprdb/data/my.cnf /etc/my.cnf >>> to fix the issue to open the db It seems that kamailio is always looking for the my.cnf in the /etc/ directory. My concern are to share this job with people who have much more experience with Kamailio. I would like to know - If the different workaround (ln) makes sense to you or is there better change to do - How to explain the logging issue when changing the modulepath variable - Is it true that kamailio is always looking for the my.cnf in the /etc/ directory? Any thoughts are welcome. Thanks in advance. Cordialement. Patrick GINHOUX

2 2

Developer Guide
by Igor Olhovskiy 27 May '18

27 May '18

Hi! Is http://www.asipto.com/pub/kamailio-devel-guide/ still holds actual info? Cause it’s mostly for 3.3 version, and now is 5.1 And where to read, except source code of course, about how to develop modules for Kamailio? Regards, Igor

2 1

Compact Headers
by Duarte Rocha 26 May '18

26 May '18

Greetings, Is there any method on Kamailio that automatically rewrites the headers in its compact form ? I know how i can do it by re-writing each of the eligible headers, but i was wondering if there is any way to do it automatically. Thanks

2 1

UAC Module Registration Table
by Greg Adams 25 May '18

25 May '18

Hey there, I am currently using the UAC module to maintain remote registrations. I am just curious if it is possible to register new users immediately rather than waiting for reg_timer_interval? I am not seeing any functionality like that but just wondering if I am missing something or if it would be possible to add in. For example an RPC command that takes a key and instantly registers that user. Thanks, Greg ________________________________

1 0

kamailio.org server maintenance work
by Oriol Capsada 25 May '18

25 May '18

Hello, A kamailio.org server maintenance work will be done today at 18:00 CEST. There will be some minutes of downtime (should be less than 5') of the following main services: website (including the wiki) and mailing lists. Regards, Oriol.

1 0

Kamailio does not receive anything after exit route
by Andrey Deykunov 25 May '18

25 May '18

Hi there, I've added a simple black list that rejects incoming requests (REGISTER and INVITE) containing certain text in the message body. Source IP addresses of the rejected requests are going to be banned and added to the ban list: route > { > ... > > > $var(ban_list_key) = $si + ":" + $sp; > if ($sht(ip_ban_list=>$var(ban_list_key)) != $null) { > xlog("L_INFO", "$var(log_fmt_tag) The incoming $rm request > from: $fU, si: $si, sp: $sp was rejected. The source host is temporarily > banned."); > sl_send_reply("400", "Bad request"); > exit; > } > > ... > > if (method=="REGISTER" || method=="INVITE") { > sht_iterator_start("i1", "black_list"); > while(sht_iterator_next("i1")) { > $var(term_is_in_wl) = 0; > $var(collision_idx) = 0; > while($sht(white_list=>$fU[$var(collision_idx)]) != $null > ) { > if($shtitval(i1) == > $sht(white_list=>$fU[$var(collision_idx)])) { > $var(term_is_in_wl) = 1; > break; > } > $var(collision_idx) = $var(collision_idx) + 1; > } > if ($var(term_is_in_wl) == 0) { > if ( $mb =~ $shtitval(i1) ) { > # found a term from the black list. access denied. > xlog("L_INFO", "$var(log_fmt_tag) The incoming $rm > request from: $fU, si: $si, sp: $sp was rejected. SIP message body contains > a term from the black list - $shtitval(i1)"); > # add device to banlist > $var(ban_info) = "" + $Ts + "_" + $shtitval(i1); > $sht(ip_ban_list=>$var(ban_list_key)) = > $var(ban_info); > xlog("L_INFO", "$var(log_fmt_tag) The source host > $si:$sp was banned for 1 hour."); > sht_iterator_end("i1"); > sl_send_reply("400", "Bad request"); > exit; > } > } > } > sht_iterator_end("i1"); > } > ... > } All works fine with UPD, but using TCP, kamailio ceases to accept any requests. Please take a look, what do I wrong. Thanks, Andrey

2 1

Performance Testing Kamailio 5.1 as SIP300 LCR on AWS - Recommendation on Tools and Methods
by KamDev Essa 24 May '18

24 May '18

I am currently setup with Kamailio 5.1 on an AWS t2.micro as a SIP300 LCR solution with a http_query call to an external resource (c# .NetCore app for LRN dip and MySQL Route choices) to get the contacts string, tag it to the Contact header an send 300. Its been a while since I have performance tested on SIP and It was SIPp. - Is http_query a bad choice for performance. Is KEMI better? - Are there any better free apps than SIPp? - Has anyone got cps stats for a SIP300 only Kamailio instance on AWS - Is AWS a bad choice for prime time LCR. Any gory stories? KD

2 3

SIP Server Hardware or VM ??
by Pinter, Gerd. 24 May '18

24 May '18

Hallo there, we are a Broadcasting Network with about 50 Stations and try to establish our own SIP Network for high quality Audio over IP with Bitrates of 192-256 kbit for transmitting an accurate stereo signal. Since we encountered problems with connections via mobile Networks we set up our SIP Server as RTP Proxy which resolved all those multiple NAT issues. But on the other hand we pay for that with a higher jitter rate. The question is, if anyone has experience if this issue maybe can be solved by running the Sip Server on a standalone hardware plugged into network Port on our Hardware Firewall, instead of a virtual Machine running on VM with virtual switches and so on. The current SIP Server has a Public IP Address, but all the Traffic runs thru Firewall, VPN, Hardware Switches and Virtual Switches. What you think? Thanks a Lot Nice regards Gerhard Pinter

4 5

count of registered users or count of tls connections
by Karthik Srinivasan 24 May '18

24 May '18

Hello, Via the routing file i am trying to find access two items:

2 5

Core Value : myself
by Duarte Rocha 23 May '18

23 May '18

Greetings, I'm doing some work with the AUTH routes and modules, and sometimes i get a "403 Not relaying" answer when the INVITE has a parameter in the R-URI. That parameter was not included in the REGISTER. I think it's this part of the code that throws this reply : # 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; } Does a different or new parameter in the R-URI trigger the "uri!=myself" ? if so, can I set it up to check R-URI but not check the parameters? Thanks

2 1

Any modules with LRN dipping capacity?
by KamDev Essa 22 May '18

22 May '18

Are there any modules with LRN dipping capacity?. If not, whats the best way we can implement it as a bolt on. KD

3 14

Best way to implement custom LCR
by KamDev Essa 22 May '18

22 May '18

I apologize for asking unit questions today and decided to just present the big picture and address what I am trying to get @ holistically. I want to implement a custom LCR only solution on kamailio. INVITE in 300 out. Nothing else. I expect awesome cps stats that makes we lean towards kamailio rather than throwing up a home grown SIP server and worrying about performance. I have the guts of the LCR logic down pat, but todays its manual. In my process I do the following : - Check LRN of the requested user. I use dig doing a ENUM call. - Use the requesting user and LRN of requested user to mediate call type. Local or Interstate using LRN check via ENUM call (above) and LCAD data lookup. I use sql query. - Mediate call type Intrastate or International if not Local or Interstate using LERG lookup. SQL query too. - dip into database with requesting user and call type and get N lists by costs ascending. SQL query. - Construct Contact info in db or kamailio. - Send 300 reply will contact having the various routes. I can automate 2 through 6 using Kamailio. Basically xquery with supplied data and deal with the Contact header with data received. 404 or default route if no data received. I cannot handle the ENUM in kamailio or mysql. The provider does not provide SIP redirect. What are my options, besides changing LRN providers? KD

3 11

300 Multiple Choices response engine
by KamDev Essa 22 May '18

22 May '18

Whats the best route to implement a basic 300 Multiple Choices response engine. INVITE in 300 Multiple Choices out with choices in the response. I have a logic to generate my list choices, so a basic process would be better. H

2 10

2 pull requests: issue #1168 and another small issue
by Valentin Christoph 22 May '18

22 May '18

Hi all, Issued two pull requests. Any comment welcome, in particular, whether it will be taken for master branch...... https://github.com/kamailio/kamailio/pull/1538 https://github.com/kamailio/kamailio/pull/1539 Kr Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

1 0

I think I found issue to Kamailio not forwarding ACK sent to it by Asterisk
by Wilkins, Steve 22 May '18

22 May '18

Hello All, When Kamailio sends a '200 OK' to Asterisk, it is putting its Public IP into the Record-Routes. In Asterisk 14.6, it would send the 'ACK' back to the Private IP Address of Kamailio, but Asterisk 15.x is using the Public IP Address that Kamailio placed in the Record-Routes so... Is there a way to force Kamailio to set the Record-Route to its Private IP address of the first Record-Route in messages forwarded to Asterisk? Thank you?

2 8

rtpp_test() in kamailio log
by Giovanni Tommasini - evosip 22 May '18

22 May '18

Hi guys, I've a question on using rtpengine with kamailio *scenario*: kamailio istance with rtpengine enabled and configured in localhost (but with a remote rtpengine is the same) modparam("rtpengine", "rtpengine_sock","udp:localhost:22222") kamailio version 4.4.6 + rtpengine version 6.3.0.0 running on Ubuntu 16.04 when I start kamailio I see a lot of INFO log rows like that May 22 12:13:12 my-dev /usr/sbin/kamailio[3388]: INFO: rtpengine [rtpengine.c:2546]: rtpp_test(): rtp proxy <udp:localhost:22222> found, support for it enabled May 22 12:13:12 my-dev /usr/sbin/kamailio[3397]: INFO: rtpengine [rtpengine.c:2546]: rtpp_test(): rtp proxy <udp:localhost:22222> found, support for it enabled ... more to the point. I see always 24 rows for each rtpengine_sock configured i n rtpengine.c I find the rtpp_tes function and that's fine, but why 24 times? thanks a lot -- Giovanni Tommasini | evosip web: evosip.cloud

2 2

Kamailio sends SIP messages to old websocket after reconnection during call
by Idris AVCI 22 May '18

22 May '18

Hi, We use kamailio as a backend for JSSIP based WebRTC UAs. We use SIP over Weboscket as transport and everythings work fine generally. Some websocket connections to kamailio drop during an active call and JSSIP reconnects immediately. New websocket connection has a different port than the dropped one but kamailio tries to send to old websocket port which is not active anymore. We use GRUU and configured all the related parameters in registrar, dialog, tm, websocket, tls, xhttp and userloc modules both for versions 4.4 and 5.1 but couldn't find a solution yet. For a typical call below are the ordered events: 1. UA register to kamailio over wss (Code: WS1) 2. Kamailio send an INVITE to the UA (Code: Call1) 3. UA answer and media starts 4. UA drops WS1 to kamailio and reconnect (Code: WS2) 5. UA has an active websocket (WS2) to kamailio 6. Kamailio tries to send BYE message of Call1 to WS1 which is no longer active and complains that websocket could not be found 7. Kamailio send an INVTE to the UA (Code: Call2) over WS2 8. UA answer and media starts 9. Kamailio send BYE message of Call2 to UA over WS2 and everything work as expected (No websocket drop) Error log: WARNING: <core> [msg_translator.c:2761]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found I think we need a mechanism that will update the dialog information of Call1 to use WS2 after reconnection. Any help greatly appreciated. Thanks.

2 2

htable - unable to update entry when event_route[htable:expired:<table>] is executed
by David Heath 22 May '18

22 May '18

Hi, Looking for some help/ advice on updating the expiration timer/ values in a hash table when when [htable:expired:<table>] occurs. On startup my script successfully reads in a set of entries from a DB; I also know from testing/ tracing that I can add new entries to the hash table in request_route. My issue occurs when the expiration timer kicks in - my goal was to re-read from the DB to ensure either the entry still exists or hasn’t changed and either delete, update or refresh the value until the next time the expiration event occurs. This way entries in the DB could be modified overtime. At the moment all that’s occurring is that in the event_route[htable:expired:cli] block the entries are being drained one-by-one. I re-read the row from the from the DB and I reset the value using $sht(cli=>$shtrecord(key)) = $xavp(cli=>MAPPED_CLI); I’ve also tried updating the expiration timer against the key. I’ve tried both $shtex(cli=>$shtrecord(key)) = 60 and $shtex(cli=>$shtrecord(key)) = $TS+60 [the latter based on the values shown from sht_print()] but the entries still get deleted. I’ve tried this against both 5.1.0 & 5.1.2. Any help appreciated. Thanks David Heath

1 0

Is it possible to change a positive (2xx) response to a negative one (4xx) during kamailio processing of a reply (g711 fallback drama)?
by George Diamantopoulos 21 May '18

21 May '18

Hello all, I've been hitting a wall in getting T.38 fallback to G.711 to work in a specific case, and I was wondering if I could use kamailio to the rescue, since all my other endeavours have failed miserably. In short, this is this the scenario where I need to have this manipulation happen: - During an established dialog between a SIP endpoint and asterisk (with kamailio in the middle acting as proxy), the latter (asterisk) sends a re-INVITE with a single offer for m=image with T.38 in the SDP. - The other (eccentric) endpoint (yes, it's a Cisco) replies with 200 OK, but will reject the T.38 offer in the SDP body instead by including a port number of '0' - Instead of falling back to G.711 as configured, asterisk in that case will do nothing, and the dialog will eventually end with a BYE (don't remember which of the two endpoints will send that, but it doesn't matter much) The thing here is that asterisk *WILL* fallback to G711 with a second re-INVITE, if the other endpoint replies to the first T.38 re-INVITE with a negative response (488) instead of a positive one (200) declining the media in the SDP body. I've looked at using freeswitch for a fax gateway instead, but apparently FS doesn't support fallback at all even in the 488 reply case, much less in the inactive media stream in 200 OK corner case I'm investigating. So, I thought I'd do that in kamailio and convert those 200 OKs to 488s with change_reply_status() from the textopsx module. Of course, the universe hates me, and this will fail spectacularly: > ERROR: textopsx [textopsx.c:301]: ki_change_reply_status(): the class of provisional or positive final replies cannot be changed So you can't change a reply's "class" with change_reply_status(), only the response code within that class can be manipulated. I guess my question is now whether there is any way to do this sort of transformation with kamailio, perhaps with some other function or method? If anyone else has had to deal with this problem and come up with a different solution I'd be glad to hear it, as neither my head nor the surrounding walls can take any more mutual banging at this point. Thanks! BR, George. P.S. I hate fax

2 4

Charging support/protocol information in Kamailio
by amit Kumar 21 May '18

21 May '18

Hi All, Could you please provide the charging related Information in Kamailio. ie list of Charging protocol/standard Kamailio Support ??. As per the kamailio modules(ims modules) , Kamailio support Diameter.Is there any other standard/interface Kamailio support.?? Thanks, amit

2 1

Cannot get rid of following errors, and wondering if this is causing some of my issues
by Wilkins, Steve 21 May '18

21 May '18

ERROR: <core> [core/pvapi.c:1452]: pv_printf(): no more space for spec value ERROR: <core> [core/pvapi.c:1461]: pv_printf(): buffer overflow -- increase the buffer size... I have pv_buffer_size=16384; tcp_rd_buf_size=16384; Any ideas? Thank you,

1 0

Supper confused on Kamailio with Asterisk Call
by Wilkins, Steve 21 May '18

21 May '18

Hi All, I am using Kamailio 5.2 and Asterisk (14.6 & 15.x) and I am having a very strange issue. If I use Asterisk 14.6, the call (WebRTC<=>WebRTC) works perfectly. However, if I use Asterisk 15.x, the call drops in 30 seconds. In comparing tcpdump files, the first place I see a difference is in the '200 OK' response to Kamailio' s 'INVITE to Asterisk'; in Asterisk 15.3 the first record Record-Route is Kamailio Public IP Address, however, in the Asterisk 14.6 (the working call) it is Kamailio' s Private IP Address. Asterisk Developers told me it is Kamailio doing this...and I am confused. Is this true, and if so why would it be different based on the Asterisk version? Thank you!

1 0

Sudden database drop
by Amar Tinawi 20 May '18

20 May '18

Hello list recently i faced strange behaviour when dealing with ng-voice version of kamailio ims specially in P-CSCF node, the database "pcscf" dropped suddenly (deleted) without any command or action from user this happened many times and unfortunately there is nothing in the syslog as error massages Any advice please ? Thanks Cheers, Ammar

2 3

WebRTC to WebRTC call Kamailio ACK Problem
by Wilkins, Steve 20 May '18

20 May '18

Hello, I am using Kamailio 5.2 and Asterisk 15.3 and while doing a WebRTC (client1) to WebRTC (client2) call, Asterisks sends a re-invite and Kamailio responds with a '200 OK', but when Asterisk sends back the "ACK" to Kamailio, it cannot be relayed. Errors => WARNING: <core> [core/msg_translator.c:2767]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found ERROR: <core> [core/msg_translator.c:1982]: build_req_buf_from_sip_req(): could not create Via header ERROR: <core> [core/forward.c:549]: forward_request(): building failed ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) As a side note, I still occasionally see the Errors => ERROR: <core> [core/pvapi.c:1452]: pv_printf(): no more space for spec value ERROR: <core> [core/pvapi.c:1461]: pv_printf(): buffer overflow -- increase the buffer size... No matter how large I make pv_buffer_size, and tcp_rd_buf_size; pv_buffer_size=16384, and tcp_rd_buf_size=16384; Thank you all!

1 0

deb.kamailio.org bionic
by Kelvin Chua 19 May '18

19 May '18

any plans of adding support for bionic? Kelvin Chua

3 3

Transport issue thought
by Wilkins, Steve 18 May '18

18 May '18

It appears that the bottom line of my TCP transport not working is that Kamailio is randomly assigning large port numbers to send the TCP traffic out on. I am not able to randomly open high ports for this purpose. Is there a way to tell Kamailio to only use specific ports for this. I have tried using force_send_socket() with Kamailio' s IP, and the port I want to send out on, but this did not work either. Does this sound like I am on the right track? Thank you, -

3 11

Transport question
by Wilkins, Steve 18 May '18

18 May '18

Hi All, I was attempting to switch from UDP to TCP for relaying. I used t_relay_to_tcp() instead of t_relay() => if (!t_relay_to_tcp()) { sl_reply_error(); } The problem is that I am still seeing the SIP messages as UDP on the Server that the messages are relayed to. Any Ideas? Thank you, Steve

6 17

Hunting - Dynamic logistics
by KamDev Essa 18 May '18

18 May '18

So I finally got to the docs on hunting under the TM Modules and there was an awesome write up and it is making a lot more sense now. Basic logic is to append_branch with your extensions and and manage serial and/or parallel forking by setting the q values. The script is static. Is there a module that automatically loads registered contacts from a grouped list of extensions or is the best way to dip into the db and DIY? I would like to know how some of you have implemented hunting, Also is there a way to set the fr_inv_timer of each branch. Thanks in advance. KD

2 1

Alcatel-Lucent Enterprise acquires majority stake in Sipwise
by Andreas Granig 18 May '18

18 May '18

Dear community, I'd like to inform you about Alcatel-Lucent Enterprise (ALE) acquiring a majority stake in Sipwise. Over the last 10 years, we at Sipwise were committed to contributing significant efforts to the open source telecom environment, resulting amongst others in the free and open source Sipwise CE soft-switch appliance, the rtpengine media relay, as well as major contributions to the Kamailio SIP proxy. Under the ALE roof, Sipwise is continuing to operate as its own brand, and our commitment to open source and our community components will be as strong as ever. Please find the official announcement here: https://www.sipwise.com/ale-sipwise/ Best regards, Andreas

3 2

Re: [SR-Users] rtpproxy not forwarding rtp packets.
by Giovanni Tommasini - evosip 18 May '18

18 May '18

Hi Younas, when you have a call if you make a trace in your sip server with sngrep or tshark can you see the RTP packet ? you says "I have kamailio server behind nat with rptproxy.", so is it possible that the router in front your SIP server blocks the traffic? could you have a trace there? I mean the packets exit the rtpproxy but don't arrive to callee or just don't come out of the rtpproxy? Giovanni Tommasini | *evosip* <http://evosip.cloud> 2018-05-18 1:36 GMT+02:00 Aqs Younas <aqsyounas(a)gmail.com>: > Thanks for replying. Callee is actually a sip provider. I see no firewall > or iptables rules on server to prevent rtp following towards callee. > > Is there anything i can do to see which thing is blocking rtps. > > Best > > On Fri, 18 May 2018 at 3:59 AM, Mack Hendricks <ap(a)goflyball.com> wrote: > >> Is there something blocking RTP traffic from reaching the callee? Is the >> callee a carrier or the actually endpoint endpoint (aka SIP Phone)? >> >> *Mack Hendricks / Head of Support / dOpenSource* >> web: http://dopensource.com >> support: +888-907-2085 >> dSIPRouter <http://dsiprouter.org> - GUI focused on implementing >> Kamailio to provide SIP Trunking and PBX Hosting Services >> >> On May 18, 2018, at 12:50 AM, Aqs Younas <aqsyounas(a)gmail.com> wrote: >> >> Greetings list, >> >> >> I have kamailio server behind nat with rptproxy. But i am getting no >> voice on the call. After taking trace i could see that rtpproxy was getting >> rtp packets but not packets was being forwarded towards callee side. >> >> Though i see in rtpproxy logs, packets being relayed from caller side. >> >> Does rtpproxy wait to receive a single rtp frame from callee before >> fowarding that to callee? >> >> Since, i am getting no rtp from callee. So, might be rptrpoxy is waiting >> to learn source address of callee side rtp, that is why it is not >> forwarding rtp packets from caller towards callee. >> >> >> Any pointer/suggestion is much appreciated. >> >> Best Regards, >> >> Aqs Younas >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users(a)lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >

4 4

Help on how to debug cfg_get variable having unexpected value?
by Malcolm O'Hare 18 May '18

18 May '18

I'm trying to debug an issue I've encountered where a variable that I've set using kamcmd cfg.set has an unexpected value when used in kamailio. The variable should be 0 or 1, but it looks like its getting set accidently somehow. 2018-05-11T22:02:58.344260+00:00 ip-172-31-129-45 kamailio[5253]: DEBUG: <core> [select.c:263]: resolve_select(): 'with_dynamodb_user_data' 2018-05-11T22:02:58.344264+00:00 ip-172-31-129-45 kamailio[5253]: DEBUG: <core> [cfg/cfg_select.c:174]: select_cfg_var(): DEBUG: select_cfg_var(): select fixup is postponed: features.with_dynamodb_user_data 2018-05-11T22:05:03.467110+00:00 ip-172-31-129-45 kamailio[5551]: INFO: <core> [cfg/cfg_ctx.c:608]: cfg_set_now(): INFO: cfg_set_now(): features.with_dynamodb_user_data has been changed to 1 2018-05-11T22:15:51.695118+00:00 ip-172-31-129-45 kamailio[5574]: [2c03c333-1705-41b1-aa26-6f10492a5a6f] DEBUG: with_dynamodb_user_data is 1157627905 I've tried to debug using gdb using breaks and set a watch on the memory address of the var, but it looks like it keeps using different memory locations. The value in the address pointed at by group.handle seems to change for each invocation of set_cfg_now, so when I try and put a watch on the address of *group.handle + var.offset it never gets hit. (gdb) print *group $36 = {num = 25, mapping = 0x7f56ee4474f0, vars = 0x0, add_var = 0x0, size = 164, meta_offset = 968, var_offset = 984, handle = 0x7f56edf92fd0, orig_handle = 0x0, dynamic = 1 '\001', next = 0x0, name_len = 8, name = "f"} (gdb) print *var $62 = {def = 0x7f56ee471668, name_len = 23, pos = 24, offset = 164, flag = 0} Any ideas? Malcolm

2 2

Auth module with shared "secret"
by Asgaroth 18 May '18

18 May '18

Hi All, I am testing a scenario where we have 3 proxies and I'm trying to ensure that if a messages comes in to 1 proxy, which challenges the ua, is able to be auth'd successfully on one of the remaining 2 proxies if, for some reason the ua decides to send the response to one of the remaining two proxies. I am testing this using a seperate kamailio proxy which loadbalances messages across the 3 proxies that I am testing with. I have 1 ua (zoiper) which I tell to subscribe for mwi, the initial subscribe is sent to proxy 1, which sends a 407 back, the new subscribe with auth, from the ua, is sent to proxy 2, which deems the nonce as expired and rechallenges, this continues in a loop across the 3 proxies and never authenticates. I think I'm missing something in my settings that I just cannot put my finger on, and/or, I'm mis-understanding the documentation and how these settings interact with each other. My auth module settings are as follows: # # ----- auth params ----- # modparam("auth", "qop", "auth") modparam("auth", "nonce_count", 1) modparam("auth", "one_time_nonce", 1) modparam("auth", "nid_pool_no", 4) modparam("auth", "nc_array_size", 4194304) modparam("auth", "otn_in_flight_no", 8388608) modparam("auth", "auth_checks_no_dlg", 9) modparam("auth", "auth_checks_in_dlg", 15) modparam("auth", "auth_checks_register", 11) modparam("auth", "secret", "secret_32_char_str") I tried with one_time_nonce enabled and disabled but the result is the same, continous rechallenge. The documentation mentions that if nonce_count and one_time_nonce are enable, and a ua sends a response with nonce and qop in the auth header then one_time_nonce will not be used, one_time nonce will be used as a fallback in the case the ua does not support qop. My auth block in the routing logic looks like so: route[AUTH_CHECK]{ xlog("L_DBG", "route[AUTH_CHECK] : $rm : Performing authentication checks for '$var(creds_key)'"); $var(challenge_flags) = 1; if ( ! pv_auth_check("$fd", "$sht(creds=>$var(creds_key)::Password)", "20", "1") ) { switch($retcode) { case -1: xlog("L_DBG", "route[AUTH_CHECK] : $rm : Generic error occurred, no reply sent out."); break; case -2: xlog("L_DBG", "route[AUTH_CHECK] : $rm : Invalid password supplied, re-challenging client and removing from hash table."); sht_rm_name_re("creds=>$var(creds_key)"); break; case -4: xlog("L_DBG", "route[AUTH_CHECK] : $rm : Nonce has expired, re-challenging client."); $var(challenge_flags) = $var(challenge_flags) + 16; break; case -5: xlog("L_DBG", "route[AUTH_CHECK] : $rm : No credentials supplied, challenging client."); break; case -6: xlog("L_DBG", "route[AUTH_CHECK] : $rm : Nonce has already been used to auth a previous request, challenging client."); break; case -8: xlog("L_DBG", "route[AUTH_CHECK] : $rm : Auth user is different to From/To user, challenging client."); break; } xlog("L_INFO", "route[AUTH_CHECK] : $rm : auth_challenge will be called with '$fd' and '$var(challenge_flags)'"); auth_challenge("$fd", "$var(challenge_flags)"); exit; } consume_credentials(); } When I look at the logs I always see the proxies seeing the nonce as expired: oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' I have checked: [1] that the proxies are all syncing time, and they are not out of sync, [2] The source address of the client doesnt change (in case auth_checks_no_dlg is failing), [3] The r-uri of the subscription dialog is the same (in case auth_checks_no_dlg is failing) I am using kamailio version 5.0.6: kamailio -V version: kamailio 5.0.6 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, 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: unknown compiled on 15:57:47 Mar 5 2018 with gcc 4.8.5 Any thoughts on what I can look at or if I'm missing something fundamental here? All suggestions/criticisms are appreciated. Thanks for reading.

3 6

rtpproxy not forwarding rtp packets.
by Aqs Younas 17 May '18

17 May '18

Greetings list, I have kamailio server behind nat with rptproxy. But i am getting no voice on the call. After taking trace i could see that rtpproxy was getting rtp packets but not packets was being forwarded towards callee side. Though i see in rtpproxy logs, packets being relayed from caller side. Does rtpproxy wait to receive a single rtp frame from callee before fowarding that to callee? Since, i am getting no rtp from callee. So, might be rptrpoxy is waiting to learn source address of callee side rtp, that is why it is not forwarding rtp packets from caller towards callee. Any pointer/suggestion is much appreciated. Best Regards, Aqs Younas

2 2

Further question on transport issue.
by Wilkins, Steve 17 May '18

17 May '18

Hello, I am still struggling with getting TCP relay to work. Kamailio is now trying to send TCP but it is using the incorrect port. Kamailio is relaying to Asterisk on xx.xx.xx.xx:some-large-port but Asterisk is listening on xx.xx.xx.xx:5060. I'm not sure why this is happening. I would appreciate any help. Thank you,

3 4

cfg script 101
by KamDev Essa 17 May '18

17 May '18

I have decided to do Hunting DIY. I needed some help with the cfg script part and using a database dip. Say I do the db dip sql_xquery("ca", "call usp_GetHuntBranches( '$rU', '$rd');", "ra");that will return the uri/q pairs and I will add using append_branch dynamically. How do I do the looping in the script? I have 2 ways to deliver the list via the sp. - Rows with uri and q values. easiest - Single row with variable paired fields like say ur1,q1,ur2,q2 etc etc Questions : - Can the script handle sql_xquery with multiple rows. If so, can I get a code snippet or example that show me how to iterate through. Is there a "while" loop capability in the script? - Is there a "substring" or "split" function in the script that could allow me to get the data as a single text with uri and q separated by a \ or |. - Or is there a document that explains the various generic functions I can use the cfg script area. KD

2 2

NatHelper's Keep-alive timeout behind proxy
by Asgaroth 17 May '18

17 May '18

Hi All, We are looking to enable nathelpers keepalive timeout parameter where it will remove an aor's contact if it doesn't receive a reply from the ua. The problem we have is that our registrar's are seperated from our proxies, so this means nathelper is sending options messages via the proxy to the ua. In this case, if the us has lost connectivity, the kamailio proxy sends a 408 Request Timeout back to the registrar and the nathelper module sees that as a response to the initial options message it sends, so it never actually removes the contact from the location table. Is there a way I can stop the proxy from sending any responses for these specific nat ping/options messages, or, is there a way I can tell the nathelper modules that a 408 response should be treated as the ua being down and then after the timeout interval remove the contact from the location table? I've had a read of the nathelper, tm and sl modules but I didnt see anything that jumps out at me that I could achieve one or the other possible solutions I mentioned above. Does anyone have any suggestions or pointers on how we can achieve this with the registrars being seperate from the edge proxies. Any pointrs/tips etc would be greatly appreciated. Thanks

2 2

Updated NAT traversal article
by Alex Balashov 17 May '18

17 May '18

Hi, I have updated this article with some other topics: http://blog.csrpswitch.com/server-side-nat-traversal-with-kamailio-the-defi… "The definitive guide" is of course an ambitious and moving target, so there may be more evolution in the future. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

5 7

t_load_contacts() and t_next_contacts() guts
by KamDev Essa 16 May '18

16 May '18

I am tackling Hunting and see that there are two functions recommended to accomplish this. Before I jump into using them I wanted to know a bit about each regard to the call flow. >From where does t_load_contacts() load contacts and how does it order then in case of a serial fork flow, Is there a generic way of doing it say with a loop in the script. Docs on these two functions are a bit cryptic for me H

2 7

Contact Header
by KamDev Essa 16 May '18

16 May '18

My scenario is UA <====> Kamailio <====> Freeswitch UA calls a number. Kam delivers to FS and FS plays a wav file. My problem is that Kam relays the 200 OK from FS back to UA without change and UA ACKs back to FS because the FS IP is in the Contact Header. - Why does Kam not change the contact header since the ACK should come back to Kam - If this is normal what do I need to do to change the contact header for the 200K to the UA to make this work. KD

3 4

Whats is wrong in this call
by Sergey Safarov 15 May '18

15 May '18

Hello I have a call with not passed ACK by Kamailio. http://prntscr.com/jhtobo Same behaviour i looked before when SIP endpoint contact string is wrong In this call look as contact string correct is correct (packets 7 and 8) and i not understand why ACK (packet 9) not passed to callee. Could you help me understand this. PCAP file is create using homer tracer and TCP transport packets is stored as UDP. Sergey

3 4

Last negative 5XX reply not getting relayed
by Sergiu Pojoga 15 May '18

15 May '18

Not sure if it's a default Kamailio behavior or something I'm doing wrong... When receiving a negative reply for an INVITE (more like in the 5xx range), proxy doesn't relay the message to the originator but instead issues it's own "500 Service Unavailable". That doesn't happen, from observations so far, to "603 Declined" or "486 Busy Here" or "404 Not Found" or "200 OK" replies. *For example, here's the negative reply proxy receives from the callee:* 2018/05/14 10:52:37.716627 65.XX.XX.166:5060 -> 65.XX.XX.167:5060 SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 65.xx.xx.167;branch=z9hG4bK9223.00b4f3143d245a895c4bb6a107add258.0;received=65.xx.xx.167;rport=5060 Via: SIP/2.0/UDP 205.xx.xx.221;received=205.xx.xx.221;rport=5060;branch=z9hG4bKQ4XpXga0vgZ8B From: "514XXXXXXX" <sip: 514XXXXXXX @205.xx.xx.221>;tag=tXj0NHrje9jNS To: <sip:450XXXXXXX@65.xx.xx.167>;tag=as18e452b7 Call-ID: 48eac296-d229-1236-a685-005056a149a4 CSeq: 122818714 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas X-Asterisk-HangupCause: Subscriber absent X-Asterisk-HangupCauseCode: 20 Content-Length: 0 *Here's what the originator gets in the final message:* 2018/05/14 10:52:37.718143 65.xx.xx.167:5060 -> 205.xx.xx.221:5060 SIP/2.0 500 Service Unavailable Via: SIP/2.0/UDP 205.xx.xx.221;rport=5060;branch=z9hG4bKQ4XpXga0vgZ8B;received=205.xx.xx.221 From: "514XXXXXXX" <sip:514XXXXXXX@205.xx.xx.221>;tag=tXj0NHrje9jNS To: <sip:450XXXXXXX@65.xx.xx.167>;tag=31a2c69b5d434ef4a60796e26f568bf0-9343 Call-ID: 48eac296-d229-1236-a685-005056a149a4 CSeq: 122818714 INVITE Server: Kamailio 5.1 Content-Length: 0 I'd like any reply to be relayed back as is, unless I do some t_check_status() => t_reply() for specific reasons (which I have none at this point). Thanks Sergiu

3 5

ACC_CDRS table with src and dst data are inverted
by Emanuele Gambaro 15 May '18

15 May '18

Hi to all, I have some troubles with acc_db module and acc_cdrs table. First scenario: Alice calls Bob and after a while, when a call is established, Alice hangs up, sending a BYE. The acc_cdrs table is filled up with the right data: * src_user, src_domain, src_ip are from Alice account (the caller) * dst_user etc etc are from Bob account (the callee) In this scenario all works as expected: the src data are from who starts the call, and the dst data are from who receives the call. In the following scenario src and dst data are inverted. Alice calls Bob and after a while, when a call is established, Bob hangs up, sending a BYE. The acc_cdrs table now is: * src_user, src_domain, src_ip are from Bob account (the callee that sends the BYE message) * dst_user etc etc are from Alice account (the caller that receives the BYE) This is the intended behavior or I’m missing something? How I can record the CDR data with the right CALLER and CALLEE data ? Emanuele

2 1

UAC Module Registration Table
by Greg Adams 15 May '18

15 May '18

Hey there, I am currently using the UAC module to maintain remote registrations. I am just curious if it is possible to register new users immediately rather than waiting for reg_timer_interval? I am not seeing any functionality like that but just wondering if I am missing something or if it would be possible to add in. For example an RPC command that takes a key and instantly registers that user. Thanks, Greg

2 1

Parallel Forking - 1 Ring on extension to notify that Call Forward is in place
by KamDev Essa 14 May '18

14 May '18

I am trying to simulate a Broadsoft feature that blips the extension with a single ring that basically heralds the user that their extension has CF turned on. I know parallel forking can do that. But how do I set the fr timer on the call to the extension @ just 1 ring (5 seconds) and let the other one the default timeout set in fr_inv_timer. KD

2 1

Tagging a call with custom pseudo variables - cfg 101
by KamDev Essa 14 May '18

14 May '18

Being spoiled by Freeswitch profiles and dialplans, the stock 1 piece Kamailio cfg file is cool @ first to get certain call flows knocked out but quite a maze when you get in deeper. Right now, using the 1 piece stock cfg file, I am faced with the fact that my carrier outbound calls started failing after I enabled VM. Basically WITH_VOICEMAIL in route LOCATION started sending the carrier calls to TOVOICEMAIL and it basically dies there. I can see the route(TOVOICEMAIL) line and can condition it out with an if but I really don't now how to differentiate between a ext to ext call or a outbound call in the cfg file. However given SQL I could. Is there a way for me to place custom pseudo variables in the call right @ the start and use them in the script as needed. For example I could mark the call as ext to ext or carrier bound right @ the start and dodge VM in the LOCATION route. KD

2 5

Question about using Kamailio and Asterisk and flow of an "INVITE"
by Wilkins, Steve 13 May '18

13 May '18

Hello All, I am trying to resolve, in my mind, the flow of a WebRTC<=>WebRTC call using Kamailio and Asterisk. Each WebRTC client is registered in Kamailio and when I call WebTRC Client1 from WebRTC Client2 what I see is -> The Invite is sent from Kamailio to Asterisk and then Asterisk is sending the Invite back to Kamailio. Also depending on The version of Asterisk, the INVITE will then get forwarded to the AOR that is registered in Kamailio for the called number. Does this seem correct? It seems like there is an extra hop in there. The reason I am now very curious now is because everything works fine if using Kamailio 5.0 and Asterisk 14.6, but I switch to Asterisk 15.3 I get the extra hop and call is dropped after 30 seconds. I would appreciate any thoughts on this. Thank you in advance.

3 6

"Sleeping Giant - No Snore" Kam?
by KamDev Essa 11 May '18

11 May '18

We have noticed pesky SIP hackers already knocking on our POC servers. Kam is holding the fort by sending out the proverbial 401/407 (snore). I can SQLOPS to do a dip to check destination domain or Source IP and nix the snore for alien requests, but wanted to know if there was a native way to keep silent when we have a request from foes. KD

3 6

Understanding port changing
by Social Boh 11 May '18

11 May '18

Hello list, I'm looking for to understand a problem have with Kamailio. The configuration is: user (behind NAT) -> Kamilio (Public IP) -> user (behind NAT) When I make a call from user A to user B (both registered) , User B receive the INVITE correctly but in the ringing SIP message change the port present on the Contact Header and same think on 200OK so when Kamailio try to send it ACK SIP message to this port fail because the Softphone listen on the port where has received the INVITE from Kamailio. This happens with 3 different Softphone: BRIA, Xlite, Linphone on the same computer. Thank you Regards -- --- I'm SoCIaL, MayBe

2 1

WebRTC to SIP gateway
by Pan Christensen 11 May '18

11 May '18

Hello! It's been several years since I've used Kamailio. My current employer wants to implement WebRTC, which is currently not supported in our SIP backend, and asked if I could set up a Kamailio server as a gateway. I've been able to make calls in all directions between SIP and WebRTC clients registered locally on Kamailio. When I tried to connect the server to the SIP backend, I ran into an issue. I'm able to register SIP clients in the backend via the gateway and make calls everywhere. However, the WebRTC client fails to register. Here are the messages between the Kamailio gateway and the SIP backend: U 2018/05/09 10:12:58.316643 GATEWAY:15060 -> DOMAIN:5060 REGISTER sip:DOMAIN SIP/2.0. Via: SIP/2.0/UDP GATEWAY:15060;branch=z9hG4bK4fc6.04d1730be5b78d595c69a3aa137987c1.0. Via: SIP/2.0/AUTO lr2l9s72ehhc.invalid;rport=61353;received=CLIENT;branch=z9hG4bK5927151. Max-Forwards: 68. To: <sip:4777519304@DOMAIN>. From: <sip:4777519304@DOMAIN>;tag=9qhqhrnj3s. Call-ID: j5h7830ivr5dfc2mn5sov1. CSeq: 4 REGISTER. Contact: <sip:c3qkm4fv@lr2l9s72ehhc.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:0ef2deac-2d56-465a-840b-543b9fd01af8>";expires=600. Expires: 600. Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO. Supported: path,gruu,outbound. User-Agent: JsSIP 3.2.9. Content-Length: 0. Path: <sip:GATEWAY:15060;lr>. . U 2018/05/09 10:12:58.368409 DOMAIN:5060 -> GATEWAY:15060 SIP/2.0 400 Wrong transport. Provided transport either invalid or not supported.. Via: SIP/2.0/UDP GATEWAY:15060;branch=z9hG4bK4fc6.04d1730be5b78d595c69a3aa137987c1.0. Via: SIP/2.0/AUTO lr2l9s72ehhc.invalid;rport=61353;received=CLIENT;branch=z9hG4bK5927151. To: <sip:4777519304@DOMAIN>;tag=91334f57. From: <sip:4777519304@DOMAIN>;tag=9qhqhrnj3s. Call-ID: j5h7830ivr5dfc2mn5sov1. CSeq: 4 REGISTER. Content-Length: 0. I believe that this error message is caused by ';transport=ws' in the Contact header. I'm not allowed to modify this header. In the backend database, I found that some other clients have ';transport=UDP' in their path headers, so I tried to add that. (Why can I not add parameters in path module without adding username?) I still got the same error. How do I best proceed? For your information: We have outsourced the development of the WebRTC client, so we are able to change it. We also have the option of paying the supplier of the backend for development there. With kind regards Pan B. Christensen Developer Phonect AS Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway E-mail: pan.christensen(a)phonect.no<mailto:pan.christensen@phonect.no> Mobile: +47 41 88 88 00 [cid:image007.png@01D3A0E8.376921D0] <http://www.phonect.no/> [facebook_2]<https://www.facebook.com/phonectno> [LinkedIn_logo_initials (1)] <https://www.linkedin.com/company/44983?trk=tyah&trkInfo=clickedVertical%3Ac…>

2 2

"No Answer" time out after number of rings
by KamDev Essa 11 May '18

11 May '18

Does Kamailio have any way to control how long it tries to connect per call? KD

2 5

Is Media Release (signaling only) possible in Kamailio?
by KamDev Essa 10 May '18

10 May '18

What cfg files changes do I need to make to get Kamailio to be a signally only server, yet manipulate the SDP part of the INVITE message to allow remote parties to send media to each other? In Freeswitch terms "bypassmedia". KD

2 14

Build Kamailio with MariaDB failed.
by Ye, Charlie (NSB - CN/Qingdao) 10 May '18

10 May '18

Hi all, Does everyone use MariaDB in Kamailio? When I built Kamailio with MariaDB, the below error happened. Would anyone give a help? Thanks in advance! [root@clcmos /usr/local/src/kamailio-devel/kamailio]$ # make all make -C src/ all make[1]: Entering directory `/usr/local/src/kamailio-devel/kamailio/src' make[3]: `libsrdb2.so.1.0' is up to date. make[3]: `libsrdb1.so.1.0' is up to date. LD (gcc) [M db_mysql.so] db_mysql.so /usr/bin/ld: cannot find -lmariadb collect2: error: ld returned 1 exit status make[2]: *** [db_mysql.so] Error 1 make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/local/src/kamailio-devel/kamailio/src' make: *** [all] Error 2 Thanks and Best Regards, Charlie

4 6

Parse presece xml and put stats into asterisk database
by Aqs Younas 09 May '18

09 May '18

Greetings list, I need to put presence info of a device into asterisk database through sqlops module. Currently, i opt to use a lua script to parse xml body of PUBLISH packet issued by pua_dialoginfo and insert trying/ringing stats into asterisk database. Is there any better way to accomplish this? Best Regards, Aqs Younas

1 0

P-CSCF error: pv_parse_spec2(): error searching pvar "uac_req"
by John M 09 May '18

09 May '18

Hello, I upgraded Kamailio 4.4 to 5.2.0-dev5 and when I tried to start the service(p-cscf), I am getting below error. However I am able to start the service with 4.4 p-cscf example cfg. Also after upgrading I-CSCF and S-CSCF nodes are running without any issue. Any help would be greatly appreciated! [root@8bd3221bd31e kamailio]# kamailio -f /usr/local/src/kamailio-5.1/kamailio/misc/examples/ims/pcscf/kamailio.cfg loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(2464) ERROR: <core> [core/pvapi.c:903]: pv_parse_spec2(): error searching pvar "uac_req" 0(2464) ERROR: <core> [core/pvapi.c:1107]: pv_parse_spec2(): wrong char [m/109] in [$uac_req(method)] at [9 (5)] 0(2464) CRITICAL: <core> [core/cfg.y:3489]: yyerror_at(): parse error in config file /usr/local/src/kamailio-5.1/kamailio/misc/examples/ims/pcscf/kamailio.cfg, line 831, column 3-18: Can't get from cache: $uac_req(method) ERROR: bad config file (1 errors) 0(2464) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Regards, Uts

2 1

Use GStreamer instead of FreeSwitch or Asterisk?
by Chandramouli P 09 May '18

09 May '18

Hello, I am using Kamailio as registration server and FreeSwitch for signalling (RTP packet handling). Can I use GStreamer instead of Freeswitch or Asterisk? Thank you. Regards, CM

2 1

Connecting UAs behind Firewall/CgNat with Kamailio without using a Stun Server
by Pinter, Gerd. 09 May '18

09 May '18

Hi there, unfortunately I am personally not a complete IT geek, I am more in to Pro Audio. Anyway, I have to maintain a SIP Server for high quality Audio transmission with special UA's. Also we use Software UA's running as Apps (Luci Live) on mobile devices such as I Phones and I pads which are by nature situated in mobile networks. Due to "lack of IP address issues" most providers use private ranges separated by CGNAT from the real Internet. This causes a lot of trouble. In some cases stun server entries on the clients helps, in other cases don't! But these days I read in announcement of a german company (Mayah Communications), which is in the ProAudio biz since decades, that they offer a SIP Service where no stun is recommended whether the clients are behind NAT or not. After I acknowledged that IT business is not witchcraft (almost :-) ) , I thought that there might exist a solution for my Kamailio SIP server to make it work like the "Mayah" Sip Server. What settings have to be done that Kamailio uses IP Header Information for SIP signalling instead of the content of SIP packages themselves. Without stun you find the Local Address in the SIP packages, and with Stun the external Address but mostly with the wrong portnumber, but in the IP header you'll find the correct IP:port. I already read a lot of the Kamailio documentation but due to the fact that I am not too deep in IT I am not sure if I got it all. If someone could give me a useful hint on that, that would make my day. Hope this description is unterstandable. Nice Regards Gerhard Pinter

4 8

Kamailio + RTPEngine + RTPEngine Recording daemon forward-to option
by Balázs Radák 08 May '18

08 May '18

Dear all, I would need some help setting up Kamailio with RTP engine. I'm trying to make use of RTPEngine recording daemon's forward-to option, to have the calls streams sent to 3rd party application. Have someone managed to have this forward-to option work? If I understood well, when RTPEngine's forward-to option is turned on, the call metainfo + streams can be sent to a socket server? Thanks in advance! Regards, Balazs

2 1

Copy $xavp
by Duarte Rocha 08 May '18

08 May '18

Greetings, I'm getting a query result in a $xavp. I'm trying to copy the entire structure to a new $xavp like this : $xavp(Valid_Acd_Results) = $xavp(ACD_Query_Result[$var(iterator)]); However, i can't retrieve the value with : xnotice(" Mode = $xavp(Valid_Acd_Results=>TBK_AcdMode)"); since it returns NULL. Am i doing something wrong? Thanks

1 0

Record-route and proxy_challenge
by Евгений Голей 08 May '18

08 May '18

Hi! Tell me why when using the save ('') function in the response generated by the server, there are the Record-route headers that were in the request. And when using the following functions: - proxy_challenge (), - www_challenge (), - send_reply (), - sl_send_reply there is no Record-route header. How can you add them? -- Evgeniy

2 2

Garbage in to field
by Kjeld Flarup 08 May '18

08 May '18

I'm having problems with my To field. I'm doing an async_route, and afterwards sends the call with auth to a SIP server. My problem is that in the first invite I see the expected To field sip:004520202020@isp.com But when the Auto is send the To field is changed to the, To field for the PBX sip:1234@pbx.local I tried to fix this with uac_replace_to("$ru"), but that ends up in this To field: sip:004520202020@isp.comsip:004520202020@isp.com It seems like uac_replace_to sometimes appends rather that replaces. If I run it twice, then it for sure appends. Is this expected behaviour. -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

2 1

How to change Calling Line ID in cr_route
by KamDev Essa 08 May '18

08 May '18

I have cr_route working well. Only problem is that the username is sent off to the carrier and the 3 or 4 digit username appears as the callerid. Is there a module or scripts that can replace the username with a db table defined callerid. Say 202 maps to a 10 digit TN. KD

1 0

STUN
by Wilkins, Steve 08 May '18

08 May '18

Hello all, Does anyone know the current status of how to enable STUN. I am using Kamailio Ver 5.x. Thank you, -Steve

5 6

Kamailio v5.1.3 Released
by Daniel-Constantin Mierla 07 May '18

07 May '18

Hello, Kamailio SIP Server v5.1.3 stable release is out. This is a maintenance release of the latest stable branch, 5.1, that includes fixes since the release of v5.1.2. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.3. For more details about version 5.1.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/04/kamailio-v5-1-3-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

3 3

Determining next hop for any SIP message
by George Diamantopoulos 07 May '18

07 May '18

Hello all, I've been trying to figure out a cleaner way to determine the next hop for a SIP message, mainly for use within the NATMANAGE route for multi-homed kamailio instances (with three or more network interfaces on the kamailio host). So far I have achieved this with a series of nested ifs, depending on whether the message is a request or a response and by calculating the next hop based on the various headers (R-URI, Route) and variables ($T_req($Ri), $dd, ) involved in SIP routing. A simpler way to do it, of course, would be to use the onsend_route, but that would most likely introduce an unnecessary overhead for all routed messages. I recently noticed there a pseudovariable called $nh(key), and I believe I can use $nh(d) to the same effect. I understand however, that this works for requests only. Also, the description of this PV in the documentation reads as follows: Return attributes of next hop for the SIP request. Address is taken from > dst_uri, if set, if not from new r-uri or original r-uri. > What is not clear to me is if this covers in-dialog requests with the Route header set as well. Does the inclusion of a route header set the dst_uri PV? And if yes, is it safe to rely on dst_uri during request processing or is it set only after completion of script processing? Lastly, is there an analogue for SIP responses? If not, is it safe to rely on first Via header to determine next hop for responses, or are there any other corner cases I need to heed? Thanks in advance, BR, George

3 6

Server-side NAT traversal with Kamailio
by Alex Balashov 07 May '18

07 May '18

Recent repeated questions on this topic have inspired a blog post: http://blog.csrpswitch.com/server-side-nat-traversal-with-kamailio-the-defi… This is a rough cut, and may be edited or expanded later. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

3 5

Route header in local reply
by Евгений Голей 07 May '18

07 May '18

Hi. Colleagues write how to configure the local responses to requests so that the response added the header of the Route in accordance with the record-route headers that were in the request ? Example SBC --> Kamailio REGISTER sip:domain.com SIP/2.0 Record-Route: <sip:AAA.AAA.AAA.AAA:6060;lr> Via: SIP/2.0/UDP CCC.CCC.CCC.CCC;branch=z9hG4bKae7f.474bd538e793b8bc912ce7d0c8e0bae8.0 Via: SIP/2.0/UDP S.s.s.s;branch=z9hG4bKae7f.d81efe54000000000000000000000000.0 To: <sip:000143949@domain.com> From: <sip:000143949@domain.com>;tag=d55866b28e60059575f1fd1facc53fdd-516b CSeq: 10 REGISTER Call-ID: f74322050fd19c19f5d0ce763cc7dbdb Max-Forwards: 70 Content-Length: 0 User-Agent: SBC Expires: 0 Kamailio --> SBC SIP/2.0 200 OK Record-Route: <sip:AAA.AAA.AAA.AAA:6060;lr> Via: SIP/2.0/UDP CCC.CCC.CCC.CCC;branch=z9hG4bKae7f.474bd538e793b8bc912ce7d0c8e0bae8.0 Via: SIP/2.0/UDP S.s.s.s;branch=z9hG4bKae7f.d81efe54000000000000000000000000.0 To: <sip:000143949@domain.com>;tag=550bc1454fc5070bdadb084c686d99a2.ac18 From: <sip:000143949@domain.com>;tag=d55866b28e60059575f1fd1facc53fdd-516b CSeq: 10 REGISTER Call-ID: f74322050fd19c19f5d0ce763cc7dbdb Server: kamailio (5.1.0 (x86_64/linux)) Content-Length: 0 -- Evgeniy

1 1

How long time does a push take?
by Kjeld Flarup 07 May '18

07 May '18

Hello I have a question to those of You who uses push via Google or Apple to initiate calls. How long time does it usually take, before the phone gets the push and wakes up? Our experience is, that it can take from 2 to 10 seconds, and that not impressing when we are talking about telephony. -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

4 5

Pleasing all kinds of UACs for RTP/SRTP with rtpengine
by Sergiu Pojoga 07 May '18

07 May '18

In continuation of previous <https://marc.info/?l=sr-users&m=150730146302404&w=2>thread about none/optional/compulsory SRTP handling... I used the suggested approach of *t_reuse_branch()* in a *tm:branch-failure* detected by t_check_status("(415)|(488)") replies, with the only difference of me doing it vice-versa from the suggested mantra of offer SRTP first then fallback to RTP. It works... well, for some UACs, not so with others. The 'good' UACs happily accept an updated branch, like for e.g. Yealink, will reply with "*488 Not Acceptable Here*" at first followed by a *200 OK*. The 'bad' UACs, like for e.g. Bria || Zoiper, will issue a "*415 Unsupported Media Type*" at first, and then a "*482 Merged Request*" after receiving and updated branch with a different RTP profile. Rightfully so, they decided that the updated branch is a retransmission, since both branches have the same CSeq, Call-ID and From tag. The branch ID however differs, so does the RTP profile obviously. *Any suggestion on how to convince all UACs that the 2nd attempt is different than the 1st one?* According to RFC 3261, a request is a merged request if ... "8.2.2.2 Merged Requests If the request has no tag in the To header field, the UAS core MUST check the request against ongoing transactions. If the From tag, Call-ID, and CSeq exactly match those associated with an ongoing transaction, but the request does not match that transaction (based on the matching rules in Section 17.2.3), the UAS core SHOULD generate a 482 (Loop Detected) response and pass it to the server transaction."

2 2

TOPOS (5.1.[23]) breaks PRACK
by Daniel Tryba 07 May '18

07 May '18

Enabling topos breaks PRACK in my simple (near default) proxy setup (attached kamailio.conf). asterisk 13 (10.0.3.147) <-> kamailio (10.0.3.87) <-> asterisk 13 (10.0.3.154) With the event route topos:msg-outgoing dropping all. The flow of messages is as expected. INVITE -> <- Ringing PRACK -> <- 200 OK With topos enabled, the PRACK doesn't get past route[WITHINGDLG] and is answered with the sl_send_reply("404","Not here"); INVITE -> <- Ringing PRACK -> <- 404 Not here These specific tests were done on kamailio 5.1.2, but with 5.1.3 I see the exact same messages.

2 1

Need help with NAT Traversal - Changing INVITE OK contact to IP port to Contact in Location
by KamDev Essa 06 May '18

06 May '18

The stock cfg states definng WITH_NAT and alls good. I did exactly that. However boing with stock cfg and WITH_NAT, the Registration info in location table is perfect with the UAs external IP address and port. This is not the case with the OK that in response to an INVITE. In the Contact of the OK the private IP shows up and Kamailio just sends it off to the Carrier who obviously cannot send a ack to a private IP. Is there a step by step guide for NAT Traversal? H

1 0

log CDRs to text file
by eyas barhouk 06 May '18

06 May '18

Hello Dears I'm trying to get Kamailio to log its CDRs (collected By ACC module ) to text file instead of a database so i used modparam("acc", "db_url", "flatstore:/var/log/acc") and in each time kamailio trying to write on the text file i get the error : " db_flatstore flat_new_connection() : not a directory" so could any one guide me about how i have to create the database text file "acc"? and what is the cause of the error i got ?

2 2

Carrier IP on softphone call history !!!
by KamDev Essa 04 May '18

04 May '18

I have a simple call flow for Inbound call going to an extendion carrier call ===> WITH_ALIASDB ===> extension (on Xlite) Works perfectly getting the call. Issue : - On Xlite I see the call as <TN>@<carrierIP>. I dont want my customers to know what carrier I use and also I want to be in the loop for CDR. - Hanging on either phone will not send a BYE to my server. How do I get Kamailio to be in the SIP ladder for this call? KD

2 7

Private IP in Contact Header of 200 OK back to Carrier, so no ACK - Call drop
by KamDev Essa 04 May '18

04 May '18

Scenario Carrier ===> Kamailio ===> Ext on domainINVITE ===> ====> 200 OK <== <=== Repeated because carrier pcap shows private 192.168 IP in Contact header. cfg has WITH_NAT setup. I assumed that all I need to do. What more is required to get Kamilio IP and port in 200K to Carrier? K

2 2

Where do I get rpc documentation for Kamailio
by KamDev Essa 04 May '18

04 May '18

I would like to perform subscriber and domain adds and reloads programmatically. Where can I find info on kamailio APIs I could use? KD

2 2

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