- sr-users - kamailio.org

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!

4 3

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…>

2 3

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.

3 4

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

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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!

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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

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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!

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

19 May '18

any plans of adding support for bionic? Kelvin Chua

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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, -

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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

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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

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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

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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 > >

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