- sr-users - kamailio.org

Re: [SR-Users] TSILO question
by Federico Cabiddu 01 Dec '16

01 Dec '16

Hi, about your questions: Q1: yes it's feasible as long as the two Registrars (Kamailio) have the location information. Please see my reply on a thread on this topic: http://lists.sip-router.org/pipermail/sr-users/2016-February/091612.html Q2: given that TSILO depends on the users' location you have to put it in the Registrar. Best regards, Federico On Wed, Nov 30, 2016 at 5:32 AM, C. J Hsu (徐熙杰) <CJ.Hsu(a)quantatw.com> wrote: > hi Federico > > i'm stuck at TSILO and wondering if you could give me some advises. > > Q1: > > in lab1, I'm having two Kamailios with proxy and registrar together, UA1 > and UA2. > UAs go thru hardware load-balancing device (F5) to register themselves in > Kamailio, > > I wonder, UA1 calling to UA2, invite suspended in Kamailio 1, waiting for > UA2; > and when UA2 come to register in Kamailio 2, TSILO in Kamailio 1 resumes > the invite and sending it to Kamailio 2, is this a no problem scenario? > > Q2: > > if we decide to separate Proxy and Registrar in different Kamailio, when > should I put TSILO in? > > > RGDS > > C J >

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Real error?
by Olvera, Victor 01 Dec '16

01 Dec '16

Is this a real error? 9(1899) ERROR: cdp [peerstatemachine.c:635]: I_Snd_CER(): I_Snd_CER(): Error on finding local host address > Socket operation on non-socket Great regards, Victor Olvera victor.olvera(a)verizonwireless.com<mailto:victor.olvera@verizonwireless.com> mobile # 001-214-923-9405

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In dialog messages cannot be routed after re-registration with outbound and gruu
by Robert 01 Dec '16

01 Dec '16

Hi, I'm using outbound with gruu enabled and with edge proxy, registrar and auth proxy all running as separate entities. My problem is once a dialog is formed between two UAs, if one side re-registers and gets assigned a new flow token, all subsequent in-dialog messages cannot be routed because the flow token in the Route set is no longer valid. What is the proper way to update the binding? I read in this thread <http://sip-router.1086192.n5.nabble.com/Outbound-and-Registrar-No-binding-u…> that "using GRUU means that Outbound is also used, so the Contact has a +sip.instance and reg-id params which are inspected by the registrar to create/update/delete a binding". How is the binding updated exactly? Any help would be appreciated. -- View this message in context: http://sip-router.1086192.n5.nabble.com/In-dialog-messages-cannot-be-routed… Sent from the Users mailing list archive at Nabble.com.

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Kamailio Processes Stop Responding
by Peter DiSanto 30 Nov '16

30 Nov '16

We have a Kamailio 4.2 machine with a custom config file we've been using for the better part of two years without issue. Suddenly we've run into an issue where Kamailio just stops responding to any INVITEs, and stops any logging. The processes keep running but stop doing anything. It is a full production machine which naturally causes problems. I have the memlog from the restart after a hang if that helps, but I presume it is too large for this list. We have increased shared memory to 1536 and Pkg memory to 32 but that didn't help, and in any case the shmem stats look OK to us. It is a full production machine doing about 200 CPS, the load is barely measurable but turning logging to debug will surely kill it. We use TOPOH and do get lots of log entries like : /usr/sbin/kamailio[17865]: ERROR: <core> [parser/parse_via.c:2048]: parse_via(): ERROR: parse_via: bad char <▒> on state 100 /usr/sbin/kamailio[17865]: ERROR: <core> [parser/parse_via.c:2738]: parse_via(): ERROR: parse_via on: <▒#010d▒▒_▒#0117Jt▒#015#012X> /usr/sbin/kamailio[17865]: ERROR: <core> [parser/parse_via.c:2744]: parse_via(): ERROR: parse_via: via parse error /usr/sbin/kamailio[17865]: ERROR: topoh [th_msg.c:438]: th_unmask_via(): cannot find cookie in via2 /usr/sbin/kamailio[17865]: ERROR: topoh [th_mask.c:165]: th_mask_decode(): invalid input string"642525435_130081688(a)xxx.xxx.xxx.xxx" /usr/sbin/kamailio[17865]: ERROR: topoh [th_msg.c:484]: th_unmask_callid(): cannot decode callid and CRITICAL: sl [../../forward.h:279]: msg_send(): unknown proto 0 However, I believe these are not new and do not seem to affect overall call processing. Anyone have any idea what to look for or how to figure this out? shmem:fragments = 4019 shmem:free_size = 1538943936 shmem:max_used_size = 84349120 shmem:real_used_size = 71668800 shmem:total_size = 1610612736 shmem:used_size = 54855384

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Offload SSL from backends with Kamailio
by Vladyslav Zakhozhai 30 Nov '16

30 Nov '16

Hi, I'm trying to understand the best (or reasonable) approach of offloading SSL encryption from backend to Kamailio. Let me simplify a little bit: UAC == SIP/TLS ==> Kamailio == SIP/UDP ==> FreeSWITCH My main problem is in Contact header of SIP packet which passes through Kamailio SIP proxy and remains unmodified. For example, REGISTER request. There is FreeSWITCH backend which is registrar server as well. UAC send REGISTER request to it through Kamailio SIP proxy via SIP/TLS. This request dispatches to backend(s) by Kamailio with dispatcher module. Backend does not configured to support TLS. In this case everything works fine: I see REGISTER requests on FreeSWITCH. But Contact header of SIP message which is passing Kamailio remains unmodified. And as result I see on FreeSWITCH something like the following: Call-ID: Jpmjp4ruHI User: user_name@domain_name Contact: "" <sip:user_name@uac_ip :27026;transport=tls;fs_path=sip%3Akamailio_ip%3A5060> Agent: Linphone/3.10.2 (belle-sip/1.5.0) Status: Registered(TLS)(unknown) EXP(2016-11-28 11:48:28) EXPSECS(110) Ping-Status: Reachable Ping-Time: 0.00 Host: kamailio_host IP: kamailio_ip Port: 5060 Auth-User: unknown Auth-Realm: domain_name MWI-Account: user_name@domain_name As a result FreeSWITCH tries to originate call over SIP/TLS and it fails because FreeSWITCH does not configured to work with TLS. I want to understand what is correct workaraound of this issue. Do I need to modify Contact header manually on kamailio host and this is right approach? Or kamailio in case of correct config rewrites this header itself? If parts of my kamailio config would be useful I will post it later. Thanks in advance. -- С уважением, Владислав Захожай

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kazoo module
by Ján Füri 30 Nov '16

30 Nov '16

Hello, I am trying to install kazoo module. According to documentation I have installed libraries : librabbitmq, libjson, libuuid, but It always fails on missing json.h : CC (gcc) [M kazoo.so] kz_trans.o In file included from kz_trans.c:53:0: kz_json.h:12:18: fatal error: json.h: No such file or directory #include <json.h> ^ compilation terminated. ../../Makefile.rules:97: recipe for target 'kz_trans.o' failed make[1]: *** [kz_trans.o] Error 1 Makefile:511: recipe for target 'modules' failed make: *** [modules] Error 1 Can anybody help me ? BR Jan

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Re: [SR-Users] Kamailio IMS virtual machine
by Franz Edler 29 Nov '16

29 Nov '16

Hi Rafael, There should not be a problem with the VMware mage. Regarding your question: you have to check kamailio-pcscf.cfg to recognize when and it how forward the SIP message towards the I-CSCF. I guess it is the statement "t_relay();" in line 896. You may provide a core tracefile (generated with tcpdump), then I may have aquick look ate your ise. BR Franz Von: Rafi Aizenstain [mailto:rafi_a@cellowireless.com] Gesendet: Montag, 28. November 2016 16:12 An: franz.edler(a)technikum-wien.at Betreff: Kamailio IMS virtual machine Hello Mr. Franz I'm trying to build a testbed using the VMware image you kindly have shared here: https://www.kamailio.org/w/2016/02/kamailio-ims-getting-started-box/ But when running the machine as-is I keep getting errors from the P-CSCF module and it doesn't forward the SIP message towards the I-CSCF. Please see the following log: root@server:~# kamailio -f /usr/local/etc/kamailio/kamailio-pcscf.cfg loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(2096) INFO: <core> [sctp_core.c:75]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Listening on udp: server.net1.test [10.0.0.9]:4060 Aliases: *: pcscf.net1.test:* WARNING: no fork mode 0(2096) INFO: rr [../outbound/api.h:54]: ob_load_api(): Failed to import bind_ob 0(2096) INFO: rr [rr_mod.c:174]: mod_init(): outbound module not available 0(2096) INFO: ims_usrloc_pcscf [hslot.c:62]: ul_init_locks(): locks array size 512 0(2096) INFO: <core> [udp_server.c:150]: probe_max_receive_buffer(): SO_RCVBUF is initially 212992 0(2096) INFO: <core> [udp_server.c:200]: probe_max_receive_buffer(): SO_RCVBUF is finally 425984 0(2096) WARNING: ims_usrloc_pcscf [usrloc_db.c:59]: connect_db(): DB connection already open... continuing 0(2096) INFO: <script>: [REGISTER] from [sip:bob@net1.test] to [sip:bob@net1.test] 0(2096) ERROR: <script>: REGISTER (sip:bob@net1.test (10.0.0.200:59721) to sip:bob@net1.test, 7013b8fc-11a0-2b70-3654-9ce68424e9a7) Looking into the configuration files I couldn't understand how does the P-CSCF determine that it has to forward the SIP message towards the I-CSCF module. Are you aware of any issue with the VMware machine you have shared. Please advise. Best Regards, Rafael A.

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LCR modules troubles
by Laura 29 Nov '16

29 Nov '16

Dear list, i have a strange situation with module LCR on kamailio 4.3.6. A customer send a call to my first kamailio and i forward it with some tech prefix to my supplier. Here the 2 correct invites.. customer is sending to me with prefix 9001 14:34:29.155719 CUSTOMER_IP.101:5060 -> EU_KMAILIO_IP.40:5060 INVITE sip:900133139864069@EU_KMAILIO_IP.40 SIP/2.0. Record-Route: <sip:CUSTOMER_IP.101;lr;ftag=33F431FC-2103;vst=AAAAAAwFBQQFAwEACgkFDQgADXADCG4DDQEABQUWHwYGLjEwMQ-->. Via: SIP/2.0/UDP CUSTOMER_IP.101;branch=z9hG4bK2d4.4853821c2be0bf7d75377fce7683f593.0. Via: SIP/2.0/UDP CUSTOMER_IP.108:5060;branch=z9hG4bK504DC634. Remote-Party-ID: <sip:441753825186@CUSTOMER_IP.108>;party=calling;screen=yes;privacy=off. From: <sip:441753825186@CUSTOMER_IP.108>;tag=33F431FC-2103. To: <sip:900133139864069@EU_KMAILIO_IP.40>. Date: Mon, 28 Nov 2016 14:34:29 GMT. Call-ID: 9A8A4545-B4AE11E6-9238E2D3-6BBD43E(a)CUSTOMER_IP.108. Supported: 100rel,timer,resource-priority,replaces,sdp-anat. Min-SE: 1800. Cisco-Guid: 2592672861-3031306726-2677997592-0419390928. User-Agent: Cisco-SIPGateway/IOS-12.x. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER. CSeq: 101 INVITE. Max-Forwards: 69. ..... and i forward to my supplier U 2016/11/28 14:34:29.157516 EU_KMAILIO_IP.40:5060 -> EU_SUPPLIER_IP.198:5060 INVITE sip:00233139864069@EU_SUPPLIER_IP.198 SIP/2.0. Record-Route: <sip:EU_KMAILIO_IP.40;lr;ftag=33F431FC-2103;did=ff5.6f01;vst=AAAAAAkAAgIAAgIKAQ4CBAYPeXIECQAABAcAAwQIAAUJOA--;nat=yes>. Record-Route: <sip:CUSTOMER_IP.101;lr;ftag=33F431FC-2103;vst=AAAAAAwFBQQFAwEACgkFDQgADXADCG4DDQEABQUWHwYGLjEwMQ-->. Via: SIP/2.0/UDP EU_KMAILIO_IP.40:5060;branch=z9hG4bK2d4.782830bfdde5e873d3dc6a71bcbba70f.0. Via: SIP/2.0/UDP CUSTOMER_IP.101;rport=5060;branch=z9hG4bK2d4.4853821c2be0bf7d75377fce7683f593.0. Via: SIP/2.0/UDP CUSTOMER_IP.108:5060;branch=z9hG4bK504DC634. Remote-Party-ID: <sip:441753825186@CUSTOMER_IP.108>;party=calling;screen=yes;privacy=off. From: <sip:441753825186@CUSTOMER_IP.108>;tag=33F431FC-2103. To: <sip:00233139864069@EU_SUPPLIER_IP.198>. Date: Mon, 28 Nov 2016 14:34:29 GMT. Call-ID: 9A8A4545-B4AE11E6-9238E2D3-6BBD43E(a)CUSTOMER_IP.108. Supported: 100rel,timer,resource-priority,replaces,sdp-anat. Min-SE: 1800. Cisco-Guid: 2592672861-3031306726-2677997592-0419390928. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER. CSeq: 101 INVITE. Max-Forwards: 68. Timestamp: 1480343669. Contact: <sip:441753825186@CUSTOMER_IP.108:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 310. On both invites the To field respect the correct information from INVITE so that the kamailio CDR is working correct. After that the supplier reply me 503 and call is come back to my kamailio. LCR modules send the call to second supplier that on this case is my second box kamailio on another countrie.. Here there is this problem that i signed on RED U 2016/11/28 14:34:43.191618 EU_KMAILIO_IP.40:5060 -> EU_SUPPLIER2_IP.242:5060 *INVITE sip:905233139864069@EU_SUPPLIER2_IP.242 SIP/2.0.* Record-Route: <sip:EU_KMAILIO_IP.40;lr;ftag=33F431FC-2103;did=ff5.6f01>. Record-Route: <sip:CUSTOMER_IP.101;lr;ftag=33F431FC-2103;vst=AAAAAAwFBQQFAwEACgkFDQgADXADCG4DDQEABQUWHwYGLjEwMQ-->. Via: SIP/2.0/UDP EU_KMAILIO_IP.40:5060;branch=z9hG4bK2d4.782830bfdde5e873d3dc6a71bcbba70f.1. Via: SIP/2.0/UDP CUSTOMER_IP.101;rport=5060;branch=z9hG4bK2d4.4853821c2be0bf7d75377fce7683f593.0. Via: SIP/2.0/UDP CUSTOMER_IP.108:5060;branch=z9hG4bK504DC634. Remote-Party-ID: <sip:441753825186@CUSTOMER_IP.108>;party=calling;screen=yes;privacy=off. From: <sip:441753825186@CUSTOMER_IP.108>;tag=33F431FC-2103. *To: <sip:900133139864069@EU_KMAILIO_IP.40>.* Date: Mon, 28 Nov 2016 14:34:29 GMT. Call-ID: 9A8A4545-B4AE11E6-9238E2D3-6BBD43E(a)CUSTOMER_IP.108. Supported: 100rel,timer,resource-priority,replaces,sdp-anat. Min-SE: 1800. Cisco-Guid: 2592672861-3031306726-2677997592-0419390928. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER. CSeq: 101 INVITE. Max-Forwards: 68. Timestamp: 1480343669. Contact: <sip:441753825186@CUSTOMER_IP.108:5060>. Expires: 180. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. .... The LCR modules make the new INVITE using the original To field and don't rewrite it like on the first request.. How is that possibile ? Any idea ? Best regards Laura

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XAVP Null
by Linux Vince 29 Nov '16

29 Nov '16

Hi, I am sure i am doing something very basic wrongly. Here is my config file snippet $xavp(caller_peer[0]=>accountname) = "AAA"; xlog("L_WARN", "XAVP ACCOUNT NAME $xavp(caller_peer[0]=>accountname) \n"); No matter what i set this XAVP at, it always prints null in log file as below WARNING: <script>: XAVP ACCOUNT NAME <null> Please help how i can set XAVP and use it further in config file.

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MSRP auth does not work with blink
by Pranathi Venkatayogi 29 Nov '16

29 Nov '16

Hi, I am trying to use MSRP feature of blink client. In the server config when auth is challenged, I see the following error in the client. Can you please advise if I am configuring the server right? Note: the return value of pv_www_authenticate is "-5". 13 2016-11-07 17:57:26.494124 [blink.exe 24976]: Closed connection to 10.0.16.51:2855 ('NotificationProxyLogger' object has no attribute 'received_illegal_data') Find attached full set of client logs. Here is the relevant portion of kamailio.cfg: if (!pv_www_authenticate("MY_DOMAIN", "$var(passwd)", "0", "$msrp(method)")) { www_challenge("MY_DOMAIN", "1"); xlog("L_ERR", "MSRP auth failed with: $var(retValue)\n"); exit; } Thanks, Pranathi Venkatayogi System Developer II (520) 745-9447 x4466 www.cyracom.com<http://www.cyracom.com/> Join us: [cid:image001.gif@01D1C9C9.C4E8B6D0] <https://www.facebook.com/pages/CyraCom-LLC/134704783312720> [cid:image002.gif@01D1C9C9.C4E8B6D0] <https://twitter.com/cyracom> [cid:image003.gif@01D1C9C9.C4E8B6D0] <http://www.linkedin.com/company/cyracom> [cid:image004.gif@01D1C9C9.C4E8B6D0] <http://www.cyracom.com/blog/> [cid:image005.png@01D1C9C9.C4E8B6D0] DISCLAIMER: This e-mail and any attached content may contain confidential or privileged material delivered for the sole use of the intended recipient(s). Any review, use, distribution or disclosure by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please contact the sender immediately by reply e-mail and delete all copies of this message. It is the recipient's responsibility to scan this e-mail and any attachments for viruses. The content of this e-mail message, including any attachments, does not comprise a contract or a portion of a contract, and so does not bind CyraCom International, Inc. or any of its agents or subsidiaries. CyraCom, LLC and Voiance Language Services, LLC are wholly owned subsidiaries of CyraCom International, Inc.

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Problems with Siremis and Kamailio - PHP7
by Diogenes Aviles 29 Nov '16

29 Nov '16

I have Kamailio 4.4 installed in ubuntu 16.04.03, with Siremis 4.2.0, PHP7 and apache2. All is OK, but when I try to install the Siremis Wizard, the step 2 (Data Base Configuration) doesn't work, and there is the message "Something Went Wrong". I tried all the option on Step 2, but the problem continuous. Then I can't go to step 3 Application Configuration. The Siremis DataBase is created and the user also. I appreciate any help. -- Diogenes

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Siremis v4.3.0 Released
by Daniel-Constantin Mierla 29 Nov '16

29 Nov '16

Hello, a new release of Siremis, the web management interface for Kamailio, has been released recently. The main work was towards making it compatible with PHP 7 and the new constraints from MySQL 5.7+. The announcement is available at: - http://siremis.asipto.com/2016/11/26/siremis-v4-3-0-released/ Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio World 2017 - May 8-10, in Berlin - www.kamailioworld.com

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custom version database/table
by David Villasmil 28 Nov '16

28 Nov '16

Hello all, I have a databse where i created a "version" table only with a version, a dispatcher records. I only want to use dispatcher on a db. I don't know if kamailio is looking for the version tables somewhere else, or it does find it in my database (not named kamailio) but the format is wrong. I used: CREATE TABLE `version` ( `table_name` VARCHAR(32) NOT NULL, `table_version` INT UNSIGNED DEFAULT 0 NOT NULL, CONSTRAINT table_name_idx UNIQUE (`table_name`) ); INSERT INTO version (table_name, table_version) values ('version','1'); INSERT INTO version (table_name, table_version) values ('dispatcher','4'); To create an populate the table, but kamailio returns: mod_init(): could not initiate a connect to the database but it does connect properly... any thoughts? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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Recent problem with fix_nated_sdp and kamailio 4.4.4
by Anthony Messina 28 Nov '16

28 Nov '16

Prior to the upgrade to Kamailio 4.4.4, I was using 4.4.3 with updates through 81df84b from the 4.4 branch and the following fix_nated_sdp call worked through the initial INVITE and when a voicemail branch was appended from the failure_route. route[NATMANAGE] { ... # RTPEngine is not needed but if STUN addresses are being used on our # internal network, rewrite the addresses to the internal source address if(!isbflagset(FLB_RTPENGINE) && has_body("application/sdp")) { if(dst_ip==10.1.1.2 && compare_pure_ips($sel(contact.uri.host), "<EXTERNAL_IP>")) { fix_nated_sdp("10"); } } ... } Since the upgrade to 4.4.4, when the original INVITE fails over to voicemail (branch appended), fix_nated_sdp creates the SDP below, doubling the replaced IP addresses. Asterisk replies with 488, as the media address is completely invalid. I need something like the above (which worked for the request and the reply) but can't figure out what the problem is. v=0 o=Zoiper 0 0 IN IP4 10.1.1.18210.1.1.182 s=Zoiper c=IN IP4 10.1.1.18210.1.1.182 t=0 0 m=audio 63732 RTP/SAVP 9 3 0 97 101 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=crypto:5 AES_256_CM_HMAC_SHA1_80 inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDIGKI2LPOLdNDOcpFdtqu5DQ== a=crypto:6 AES_256_CM_HMAC_SHA1_32 inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDIGKI2LPOLdNDOcpFdtqu5DQ== a=crypto:3 AES_192_CM_HMAC_SHA1_80 inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDIGKI2LPOLdNA= a=crypto:4 AES_192_CM_HMAC_SHA1_32 inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDIGKI2LPOLdNA= a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDI a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:n2C0BcdXOr32BPTEE8pfHII9mamAK566xMNvQIDI a=oldmediaip:<EXTERNAL_IP> a=oldmediaip:<EXTERNAL_IP> a=oldmediaip:<EXTERNAL_IP> a=oldmediaip:<EXTERNAL_IP> -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery F9B6 560E 68EA 037D 8C3D D1C9 FF31 3BDB D9D8 99B6

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kamailio core dumps once every few days
by davy van de moere 28 Nov '16

28 Nov '16

Gents, On a not really loaded kamailio (less than 50 users at the moment), I'm getting a core dump every few days. I've tried multiple verisons of Kamailio, 4.4.x , 4.3.x and also on 5.0.0-dev7 . I can't reproduce it adhoc, but every few days it happens... Is that something someone wants to jump in? grtz, davy Version: version: kamailio 5.0.0-dev7 (x86_64/linux) 7360b0 [New LWP 29728] [Thread debugging using libthread_db enabled] Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1". Core was generated by `/usr/local/sbin/kamailio -P /var/run/kamailio.pid -f /usr/local/etc/kamailio//k'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f92c5a0be98 in __dialog_sendpublish (dlg=0x7f92c0e3a6c0, type=64, _params=0x7f92c0423500 <params>) at pua_dialoginfo.c:264 264 uri = dlginfo->to_uri; (gdb) backtrace #0 0x00007f92c5a0be98 in __dialog_sendpublish (dlg=0x7f92c0e3a6c0, type=64, _params=0x7f92c0423500 <params>) at pua_dialoginfo.c:264 #1 0x00007f92c01e2563 in run_dlg_callbacks (type=64, dlg=0x7f92c0e3a6c0, req=0x7f92cc6d5db8, rpl=0x0, dir=1, dlg_data=0x0) at dlg_cb.c:256 #2 0x00007f92c01a92b1 in dlg_terminated (req=0x7f92cc6d5db8, dlg=0x7f92c0e3a6c0, dir=1) at dlg_handlers.c:366 #3 0x00007f92c01b610f in dlg_onroute (req=0x7f92cc6d5db8, route_params=0x7ffc7de27ba0, param=0x0) at dlg_handlers.c:1409 #4 0x00007f92c9292171 in run_rr_callbacks (req=0x7f92cc6d5db8, rr_param=0x7ffc7de27c20) at rr_cb.c:96 #5 0x00007f92c9287ac2 in after_loose (_m=0x7f92cc6d5db8, preloaded=0) at loose.c:928 #6 0x00007f92c9288168 in loose_route (_m=0x7f92cc6d5db8) at loose.c:962 #7 0x00007f92c928b7ee in w_loose_route (msg=0x7f92cc6d5db8, p1=0x0, p2=0x0) at rr_mod.c:266 #8 0x000000000041f5d9 in do_action (h=0x7ffc7de285b0, a=0x7f92cc5f7a70, msg=0x7f92cc6d5db8) at action.c:1054 #9 0x000000000042bcde in run_actions (h=0x7ffc7de285b0, a=0x7f92cc5f7a70, msg=0x7f92cc6d5db8) at action.c:1552 #10 0x000000000042c383 in run_actions_safe (h=0x7ffc7de29c60, a=0x7f92cc5f7a70, msg=0x7f92cc6d5db8) at action.c:1620 #11 0x000000000061fe86 in rval_get_int (h=0x7ffc7de29c60, msg=0x7f92cc6d5db8, i=0x7ffc7de28cc0, rv=0x7f92cc5f7bc8, cache=0x0) at rvalue.c:912 #12 0x00000000006243c3 in rval_expr_eval_int (h=0x7ffc7de29c60, msg=0x7f92cc6d5db8, res=0x7ffc7de28cc0, rve=0x7f92cc5f7bc0) at rvalue.c:1910 #13 0x000000000041f0d1 in do_action (h=0x7ffc7de29c60, a=0x7f92cc600430, msg=0x7f92cc6d5db8) at action.c:1030 #14 0x000000000042bcde in run_actions (h=0x7ffc7de29c60, a=0x7f92cc600430, msg=0x7f92cc6d5db8) at action.c:1552 #15 0x000000000041f548 in do_action (h=0x7ffc7de29c60, a=0x7f92cc6006d0, msg=0x7f92cc6d5db8) at action.c:1045 #16 0x000000000042bcde in run_actions (h=0x7ffc7de29c60, a=0x7f92cc6006d0, msg=0x7f92cc6d5db8) at action.c:1552 #17 0x000000000041c0fb in do_action (h=0x7ffc7de29c60, a=0x7f92cc545730, msg=0x7f92cc6d5db8) at action.c:678 #18 0x000000000042bcde in run_actions (h=0x7ffc7de29c60, a=0x7f92cc537760, msg=0x7f92cc6d5db8) at action.c:1552 #19 0x000000000042c44b in run_top_route (a=0x7f92cc537760, msg=0x7f92cc6d5db8, c=0x0) at action.c:1641 #20 0x00000000005726a2 in receive_msg ( buf=0xa1c200 <buf> "BYE sip:echo@123.48.11.215:5070;transport=udp SIP/2.0\r\nVia: SIP/2.0/UDP 0.0.0.0:5060;rport;branch=z9hG4bKPj26b505ba-7ede-4198-a917-bde966b6d8b6\r\nMax-Forwards: 69\r\nFrom: sip:192.168.60.1;tag=9a10d8ab-"..., len=462, rcv_info=0x7ffc7de29fb0) at receive.c:264 #21 0x000000000047bcdc in udp_rcv_loop () at udp_server.c:507 #22 0x0000000000509de1 in main_loop () at main.c:1618 #23 0x00000000005112de in main (argc=5, argv=0x7ffc7de2a478) at main.c:2639 (gdb)

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Balancer with Dispacther Module
by Rodrigo Moreira 28 Nov '16

28 Nov '16

I need help. I'm building a load balancer using the dispatcher. It does NAT with too many SIP Servers. But when the balancer receives an ACK from UAC, it immediately returns the message "404 Not Here" to UAC. Help me solve this, since this message (ACK) has to be forwarded to SIP Servers and finally to UAS to start an RTP session. I thank you. Att., -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

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Re: [SR-Users] msilo
by Slava Bendersky 27 Nov '16

27 Nov '16

Hello Daniel, I am trying more simple example of $dbr in REGISTER first. Insert into location temporary data. The problem is not looping though all entries in database only last one. Which compare same entry 100 times. if(is_method("REGISTER")) { add_path_received(); xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); sql_query("cb", "SELECT * FROM location", "result"); xlog("rows: $dbr(result=>rows) cols: $dbr(result=>cols)\n"); $var(result) = 0; while($var(i)<$dbr(result=>rows)) { sql_pvquery("cb", "SELECT ruid FROM location WHERE callid = '$ci'", "$var(dbruid)"); if(fnmatch("$var(dbruid)", "$var(ruid)")) { xlog("L_INFO", "AOR: Found match --> [$var(dbruid)] = [$var(ruid)]\n"); } else { sql_query_async("cb","DELETE FROM location WHERE ruid = '$var(dbruid)'"); sql_query_async("cb","INSERT INTO location(id, username, expires, contact, user_agent, domain, callid, ruid) VALUES(0, '$au', NOW(), '$ct', '$ua', '$fd', '$ci', '$var(ruid)')"); } } } From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, 27 November, 2016 00:58:29 Subject: Re: [SR-Users] msilo Ok lua works right now. I was need remove "" in the value. Slava From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, 27 November, 2016 00:29:39 Subject: Re: [SR-Users] msilo Hello Everyone, I tried use sr.pv.sets, but not sure how to pass value to it. #!/usr/bin/lua math.randomseed(os.time()) ruid = string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf)) print(ruid) ---> Works sr.pv.sets("$var(ruid)", "ruid") ---> In kamailio just static ruid Output AOR: New user Ruid -->[ruid] Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, 26 November, 2016 23:41:46 Subject: Re: [SR-Users] msilo Hello Everyone, Can't find how actually return actual value from lua script not 1. Because my setup is pass through REGISTER to B2BUA, I can't use registrar module. I though insert manually, but location table require ruid entry. I installed small lua script to generate it. Not sure if this good approach for it. When user send first REGISTER in request do like this if(is_method("REGISTER")) { add_path_received(); xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); sql_query_async("cb","INSERT INTO location(id, username, expires, contact, user_agent, domain, callid, ruid) VALUES(0, '$au', $TS, '$ct', '$ua', '$fd', '$ci', '$var(ruid)')"); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 25 November, 2016 02:14:59 Subject: Re: [SR-Users] msilo Hello Everyone, I am trying run small lua script to generate random formatted string, but always get 1. Is script need be set in special way ? That related to MSILO setup which I am trying to do. xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); Output Nov 25 01:06:54 cavprx00 /usr/sbin/kamailio[3345]: INFO: <script>: AOR: New user Ruid -->[1] Actual output [root@cavprx00 kamailio]# /usr/bin/ruid.lua uloc-8b7dae10-c17-2 [root@cavprx00 kamailio]# cat /usr/bin/ruid.lua #!/usr/bin/lua math.randomseed(os.time()) print(string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf))) Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 12:15:45 Subject: Re: [SR-Users] msilo Hello Everyone, For dump I though do something like this. No sure if possible optimise it. Also how if possible convert inc_time to human readable ? xlog("L_INFO", "[$rm], Got Register request for <$tU> dumping OFFLINE SMS\n"); while(sql_pvquery("cb","SELECT id FROM silo ORDER BY id ASC","$avp(id)")) { sql_pvquery("cb", "SELECT dst_addr, src_addr, body, inc_time FROM silo WHERE id = '$avp(id)'","$avp(dst), $avp(src), $avp(body), $avp(time)"); crypto_aes_decrypt("$var(body)", "decryption key", "$avp(decrypted)"); $uac_req(method)="MESSAGE"; $uac_req(furi)=$avp(src); $uac_req(turi)=$avp(dst); $uac_req(body)= $avp(decrypted) ; uac_req_send(); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 09:34:18 Subject: Re: [SR-Users] msilo Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] msilo
by Slava Bendersky 27 Nov '16

27 Nov '16

Ok lua works right now. I was need remove "" in the value. Slava From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Sunday, 27 November, 2016 00:29:39 Subject: Re: [SR-Users] msilo Hello Everyone, I tried use sr.pv.sets, but not sure how to pass value to it. #!/usr/bin/lua math.randomseed(os.time()) ruid = string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf)) print(ruid) ---> Works sr.pv.sets("$var(ruid)", "ruid") ---> In kamailio just static ruid Output AOR: New user Ruid -->[ruid] Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, 26 November, 2016 23:41:46 Subject: Re: [SR-Users] msilo Hello Everyone, Can't find how actually return actual value from lua script not 1. Because my setup is pass through REGISTER to B2BUA, I can't use registrar module. I though insert manually, but location table require ruid entry. I installed small lua script to generate it. Not sure if this good approach for it. When user send first REGISTER in request do like this if(is_method("REGISTER")) { add_path_received(); xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); sql_query_async("cb","INSERT INTO location(id, username, expires, contact, user_agent, domain, callid, ruid) VALUES(0, '$au', $TS, '$ct', '$ua', '$fd', '$ci', '$var(ruid)')"); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 25 November, 2016 02:14:59 Subject: Re: [SR-Users] msilo Hello Everyone, I am trying run small lua script to generate random formatted string, but always get 1. Is script need be set in special way ? That related to MSILO setup which I am trying to do. xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); Output Nov 25 01:06:54 cavprx00 /usr/sbin/kamailio[3345]: INFO: <script>: AOR: New user Ruid -->[1] Actual output [root@cavprx00 kamailio]# /usr/bin/ruid.lua uloc-8b7dae10-c17-2 [root@cavprx00 kamailio]# cat /usr/bin/ruid.lua #!/usr/bin/lua math.randomseed(os.time()) print(string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf))) Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 12:15:45 Subject: Re: [SR-Users] msilo Hello Everyone, For dump I though do something like this. No sure if possible optimise it. Also how if possible convert inc_time to human readable ? xlog("L_INFO", "[$rm], Got Register request for <$tU> dumping OFFLINE SMS\n"); while(sql_pvquery("cb","SELECT id FROM silo ORDER BY id ASC","$avp(id)")) { sql_pvquery("cb", "SELECT dst_addr, src_addr, body, inc_time FROM silo WHERE id = '$avp(id)'","$avp(dst), $avp(src), $avp(body), $avp(time)"); crypto_aes_decrypt("$var(body)", "decryption key", "$avp(decrypted)"); $uac_req(method)="MESSAGE"; $uac_req(furi)=$avp(src); $uac_req(turi)=$avp(dst); $uac_req(body)= $avp(decrypted) ; uac_req_send(); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 09:34:18 Subject: Re: [SR-Users] msilo Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] msilo
by Slava Bendersky 27 Nov '16

27 Nov '16

Hello Everyone, I tried use sr.pv.sets, but not sure how to pass value to it. #!/usr/bin/lua math.randomseed(os.time()) ruid = string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf)) print(ruid) ---> Works sr.pv.sets("$var(ruid)", "ruid") ---> In kamailio just static ruid Output AOR: New user Ruid -->[ruid] Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Saturday, 26 November, 2016 23:41:46 Subject: Re: [SR-Users] msilo Hello Everyone, Can't find how actually return actual value from lua script not 1. Because my setup is pass through REGISTER to B2BUA, I can't use registrar module. I though insert manually, but location table require ruid entry. I installed small lua script to generate it. Not sure if this good approach for it. When user send first REGISTER in request do like this if(is_method("REGISTER")) { add_path_received(); xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); sql_query_async("cb","INSERT INTO location(id, username, expires, contact, user_agent, domain, callid, ruid) VALUES(0, '$au', $TS, '$ct', '$ua', '$fd', '$ci', '$var(ruid)')"); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 25 November, 2016 02:14:59 Subject: Re: [SR-Users] msilo Hello Everyone, I am trying run small lua script to generate random formatted string, but always get 1. Is script need be set in special way ? That related to MSILO setup which I am trying to do. xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); Output Nov 25 01:06:54 cavprx00 /usr/sbin/kamailio[3345]: INFO: <script>: AOR: New user Ruid -->[1] Actual output [root@cavprx00 kamailio]# /usr/bin/ruid.lua uloc-8b7dae10-c17-2 [root@cavprx00 kamailio]# cat /usr/bin/ruid.lua #!/usr/bin/lua math.randomseed(os.time()) print(string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf))) Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 12:15:45 Subject: Re: [SR-Users] msilo Hello Everyone, For dump I though do something like this. No sure if possible optimise it. Also how if possible convert inc_time to human readable ? xlog("L_INFO", "[$rm], Got Register request for <$tU> dumping OFFLINE SMS\n"); while(sql_pvquery("cb","SELECT id FROM silo ORDER BY id ASC","$avp(id)")) { sql_pvquery("cb", "SELECT dst_addr, src_addr, body, inc_time FROM silo WHERE id = '$avp(id)'","$avp(dst), $avp(src), $avp(body), $avp(time)"); crypto_aes_decrypt("$var(body)", "decryption key", "$avp(decrypted)"); $uac_req(method)="MESSAGE"; $uac_req(furi)=$avp(src); $uac_req(turi)=$avp(dst); $uac_req(body)= $avp(decrypted) ; uac_req_send(); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 09:34:18 Subject: Re: [SR-Users] msilo Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Re: [SR-Users] msilo
by Slava Bendersky 27 Nov '16

27 Nov '16

Hello Everyone, Can't find how actually return actual value from lua script not 1. Because my setup is pass through REGISTER to B2BUA, I can't use registrar module. I though insert manually, but location table require ruid entry. I installed small lua script to generate it. Not sure if this good approach for it. When user send first REGISTER in request do like this if(is_method("REGISTER")) { add_path_received(); xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); sql_query_async("cb","INSERT INTO location(id, username, expires, contact, user_agent, domain, callid, ruid) VALUES(0, '$au', $TS, '$ct', '$ua', '$fd', '$ci', '$var(ruid)')"); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 25 November, 2016 02:14:59 Subject: Re: [SR-Users] msilo Hello Everyone, I am trying run small lua script to generate random formatted string, but always get 1. Is script need be set in special way ? That related to MSILO setup which I am trying to do. xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); Output Nov 25 01:06:54 cavprx00 /usr/sbin/kamailio[3345]: INFO: <script>: AOR: New user Ruid -->[1] Actual output [root@cavprx00 kamailio]# /usr/bin/ruid.lua uloc-8b7dae10-c17-2 [root@cavprx00 kamailio]# cat /usr/bin/ruid.lua #!/usr/bin/lua math.randomseed(os.time()) print(string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf))) Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 12:15:45 Subject: Re: [SR-Users] msilo Hello Everyone, For dump I though do something like this. No sure if possible optimise it. Also how if possible convert inc_time to human readable ? xlog("L_INFO", "[$rm], Got Register request for <$tU> dumping OFFLINE SMS\n"); while(sql_pvquery("cb","SELECT id FROM silo ORDER BY id ASC","$avp(id)")) { sql_pvquery("cb", "SELECT dst_addr, src_addr, body, inc_time FROM silo WHERE id = '$avp(id)'","$avp(dst), $avp(src), $avp(body), $avp(time)"); crypto_aes_decrypt("$var(body)", "decryption key", "$avp(decrypted)"); $uac_req(method)="MESSAGE"; $uac_req(furi)=$avp(src); $uac_req(turi)=$avp(dst); $uac_req(body)= $avp(decrypted) ; uac_req_send(); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 09:34:18 Subject: Re: [SR-Users] msilo Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

tm:local-request does not intercept locally generated CANCEL
by Andy 25 Nov '16

25 Nov '16

should tm:local-request intercept CANCELs, if not then where can the locally generated CANCELs from t_cancel.c be intercepted -- View this message in context: http://sip-router.1086192.n5.nabble.com/tm-local-request-does-not-intercept… Sent from the Users mailing list archive at Nabble.com.

2 2

protocol/port mismatch
by Yuriy Gorlichenko 25 Nov '16

25 Nov '16

Hi. Have strange issue Calling from kamailio through WS invite and resposes goes ok but with ACK to WS client has this issue get_send_socket2(): protocol/port mismatch (forced tls:kamailioIP:4443, to udp:MyWSClientIP:65451) Guess kamailio thinks that it is UDP client because $ru is sip:10002@MyWSClientIP:65451 But can not understand how to fix this...

2 2

Re: [SR-Users] msilo
by Slava Bendersky 25 Nov '16

25 Nov '16

Hello Everyone, For dump I though do something like this. No sure if possible optimise it. Also how if possible convert inc_time to human readable ? xlog("L_INFO", "[$rm], Got Register request for <$tU> dumping OFFLINE SMS\n"); while(sql_pvquery("cb","SELECT id FROM silo ORDER BY id ASC","$avp(id)")) { sql_pvquery("cb", "SELECT dst_addr, src_addr, body, inc_time FROM silo WHERE id = '$avp(id)'","$avp(dst), $avp(src), $avp(body), $avp(time)"); crypto_aes_decrypt("$var(body)", "decryption key", "$avp(decrypted)"); $uac_req(method)="MESSAGE"; $uac_req(furi)=$avp(src); $uac_req(turi)=$avp(dst); $uac_req(body)= $avp(decrypted) ; uac_req_send(); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 09:34:18 Subject: Re: [SR-Users] msilo Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Re: [SR-Users] msilo
by Slava Bendersky 25 Nov '16

25 Nov '16

Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Re: [SR-Users] msilo
by Slava Bendersky 25 Nov '16

25 Nov '16

Hello Everyone, I am trying run small lua script to generate random formatted string, but always get 1. Is script need be set in special way ? That related to MSILO setup which I am trying to do. xlog("L_INFO", "MSILO: New user request saving contact [$ct]\n"); $var(ruid) = lua_dofile("/usr/bin/ruid.lua"); xlog("L_INFO", "AOR: New user Ruid -->[$var(ruid)]\n"); Output Nov 25 01:06:54 cavprx00 /usr/sbin/kamailio[3345]: INFO: <script>: AOR: New user Ruid -->[1] Actual output [root@cavprx00 kamailio]# /usr/bin/ruid.lua uloc-8b7dae10-c17-2 [root@cavprx00 kamailio]# cat /usr/bin/ruid.lua #!/usr/bin/lua math.randomseed(os.time()) print(string.format("uloc-%08x-%03x-%x", math.random(0xffffffff), math.random(0xfff), math.random(0xf))) Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 12:15:45 Subject: Re: [SR-Users] msilo Hello Everyone, For dump I though do something like this. No sure if possible optimise it. Also how if possible convert inc_time to human readable ? xlog("L_INFO", "[$rm], Got Register request for <$tU> dumping OFFLINE SMS\n"); while(sql_pvquery("cb","SELECT id FROM silo ORDER BY id ASC","$avp(id)")) { sql_pvquery("cb", "SELECT dst_addr, src_addr, body, inc_time FROM silo WHERE id = '$avp(id)'","$avp(dst), $avp(src), $avp(body), $avp(time)"); crypto_aes_decrypt("$var(body)", "decryption key", "$avp(decrypted)"); $uac_req(method)="MESSAGE"; $uac_req(furi)=$avp(src); $uac_req(turi)=$avp(dst); $uac_req(body)= $avp(decrypted) ; uac_req_send(); } Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 09:34:18 Subject: Re: [SR-Users] msilo Hello Daniel, That good idea, I will need help with sqlops to update information properly. Make sure the query is optimal. I will still can use if(m_store("$tu")), because it maintain offline notification, but I can use something like this. For dump I will put uac. route[OFFLINE_MESSAGE] { if(!is_method("MESSAGE")) { return; } if(isflagset(FLAG_FROM_PEER)) { if(!lookup("location")) { xlog("L_INFO", "User $tU domain $td offline. Trying store SMS for later delivery\n"); xlog("L_INFO", "SMS received from $fU to $tU domain [$td] --> storing using MSILO\n"); # MSILO - storing as offline message if(m_store("$tu")) { xlog("L_INFO", "MSILO: offline message stored\n"); # Encrypt stored offline message crypto_aes_encrypt("$rb", "mykey", "$avp(encrypted)"); sql_pvquery("cb", "select * from silo ORDER BY id DESC LIMIT 1","$avp(id)"); sql_query_async("cb","UPDATE silo SET body = $avp(encrypted) WHERE id = $avp(id)"); if(is_request()) { if(!sl_send_reply("202", "Accepted")) { sl_reply_error(); } } else { xlog("L_INFO", "MSILO: offline message NOT stored\n"); #if(!sl_send_reply("503", "Service Unavailable")) { # sl_reply_error(); #} } } t_on_failure("SMS_FAIL_ROUTE"); exit; } } } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 24 November, 2016 06:45:48 Subject: Re: [SR-Users] msilo I expect the encryption/decryption can be done with some triggers in mysql server. In kamailio config you can use crypto module to encrypt a text and store it in a variable: - https://www.kamailio.org/docs/modules/stable/modules/crypto.html Then you can use sqlops to insert into the database. The issue comes when dumping stored messages... probably you can just replace msilo with sqlops+uac at the expense of a more complex configuration file. On the other hand, probably adds some privacy to the local platform operators, which have access only to the mysql, because the key will be in kamailio.cfg. For full privacy, the endpoints should do the encryption/decryption with a key they agreed before, without being known by the server. Cheers, Daniel On 24/11/2016 05:28, Slava Bendersky wrote: Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] msilo
by Slava Bendersky 24 Nov '16

24 Nov '16

Sent message to mailing list From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Thursday, 24 November, 2016 00:25:30 Subject: Re: [SR-Users] msilo Hello Everyone, I want to ask about another improvement for MSILO module. If possible encrypt BODY column in database. That will improve some privacy concerns for storing body in plain text. Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Sent: Wednesday, 23 November, 2016 21:10:53 Subject: Re: [SR-Users] msilo Hello Everyone, Need another set of eyes, can't find how to To: got << in uri. Nov 23 19:50:55 cavprx00 kamailio: 3(9625) DEBUG: <core> [parser/parse_uri.c:1280]: parse_uri(): parse_uri: bad uri, state 0 parsed: <<sip> (4) / <<sip:4200@client_public_ip:49418>;messagetype=SMS> (47) Nov 23 19:50:55 cavprx00 kamailio: 3(9625) ERROR: tm [ut.h:254]: uri2dst2(): ERROR: uri2dst: bad_uri: [<sip:4200@ client_public_ip :49418>;messagetype=SMS] Nov 23 19:50:55 cavprx00 kamailio: 3(9625) ERROR: tm [t_fwd.c:1723]: t_forward_nonack(): ERROR: t_forward_nonack: failure to add branches route[MESSAGE_FORWARD] { xlog("L_INFO", "Incoming new SMS [$rm] from $tU --> $td\n"); if(fnmatch("$rs", "202") && fnmatch("$rr", "Accepted") && allow_trusted("$si", "$proto")) { $sht(b=>from) = $fu; $sht(b=>oldto) = $tu; $avp(new_to) = "<" + $sht(b=>oldto) + ">" + ";messagetype=SMS"; xlog("L_INFO", "MESSAGE: B2BUA send --> [$rs] with [$rr] from [$sht(b=>from)] to [$sht(b=>oldto)] -->[$td]\n"); } if(compare_ips("$td", "10.18.130.27")) { xlog("L_INFO", "Message from B2BUA contain domain --> $td user --> $(sht(b=>oldto){uri.user}) updating...\n"); sql_pvquery("cb", "select contact from location where username = '$(sht(b=>oldto){uri.user})'","$avp(dst)"); #$avp(new_dst) = $(hdr(Route){param.value,received}); xlog("L_INFO", "New destination --> $avp(dst)\n"); $avp(new_to) = "<" + $(avp(dst){tobody.uri}) + ">" + ";messagetype=SMS"; } if(fnmatch("$fU", "offline")) { xlog("L_INFO", "OFFLINE SMS: from [$fU] user --> $fU\n"); $avp(from) = $(sht(b=>oldto){uri.user}); xlog("L_INFO", "OFFLINE SMS: Searching destination user $avp(oexten)\n"); sql_pvquery("cb", "select contact from location where username = '$avp(from)'","$avp(dst)"); xlog("L_INFO", "OFFLINE SMS: set new destination --> $avp(dst)\n"); $avp(new_to) = "<" + $(avp(dst){tobody.uri}) + ">" + ";messagetype=SMS"; } xlog("L_INFO", "OFFLINE SMS: new AVP DST $avp(new_to)\n"); $sht(a=>to) = $avp(new_to); remove_hf("To"); insert_hf("To: $sht(a=>to)\r\n", "To"); $du = $sht(a=>to); xlog("SMS from --> $fU domain $fd\n"); xlog("-------------------------------------\n"); xlog("L_INFO", "FROM --> $avp(from)\n"); xlog("L_INFO", "OLD_TO --> $sht(b=>oldto)\n"); xlog("L_INFO", "NEW_TO --> $avp(new_to)\n"); } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "volga629" <volga629(a)skillsearch.ca>, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 17 November, 2016 05:44:22 Subject: Re: [SR-Users] msilo Hello, if you want to fetch them for config usage is ok, otherwise m_dump() is still better, because it takes care of removing delivered messages, etc... Cheers, Daniel On 16/11/16 16:08, Slava Bendersky wrote: Hello Daniel, Yes that good alternative. I will use sql to fetch or insert staff. Slava From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 16 November, 2016 04:37:24 Subject: Re: [SR-Users] msilo Hello, as an alternative, I think it should work by using sql_query(...) to insert whatever values one would like to have in msilo table. This might be a quick workaround until someone will code the extension to get the body via a function parameter. Cheers, Daniel On 15/11/16 16:07, Federico Cabiddu wrote: BQ_BEGIN Hi Slava, if I understand correctly your scenario, you could call t_on_failure before relaying the MESSAGE. In this way you should have access to the original message in the failure route triggered after receiving a final reply and you should be able to store it with the regular msilo functions. Hope this helps. Cheers, Federico On 15 Nov 2016 3:47 pm, "Slava Bendersky" < volga629(a)skillsearch.ca > wrote: BQ_BEGIN Hello Daniel, I am not programmer, but I want place request if kaamilio devs can complete it . Thanks Slava. From: "Daniel-Constantin Mierla" < miconda(a)gmail.com > To: "sr-users" < sr-users(a)lists.sip-router.org > Sent: Tuesday, 15 November, 2016 03:00:19 Subject: Re: [SR-Users] msilo Hello, for me it is fine to add a new parameter to m_store() that takes the body content. Are you asking for more details of how can be done? Cheers, Daniel On 13/11/16 07:08, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, I asking question/request to add improvement for msilo module where will have ability set $avp(i:body) in mod params. That will allow use it in m_store() by specifying body message $avp. In my case registration is on B2BUA so kamailio pass through/proxy to destination. In order to check user status online/offline kamailio wait for reply from B2BUA on MESSAGE status if it return error 503 that mean user offline and m_store(0 should store message in db, but the issue that error 503 is in final stage and MESSAGE body is not available any more. The improvement will allow specify body $var in m_store() so new way will be m_store($tu, $avp(body)). Any help thank you. Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com

2 2

Howto send rtp stats infos to homer
by Oliver Roth 23 Nov '16

23 Nov '16

Hi there We are using homer to analyce and monitor traffic. I did not get a solution to send x-rtp or p-rtp stats to homer. How can I find infos about jitter, sent/received packages, lost packages, etc.? Our current infos are based on Rtpstat provided by kamailio. At the moment I do $var(xrtpstat) = $(rtpstat{s.striptail,1}); # Work the stats $var(octetsent) = $(var(xrtpstat){s.select,1, }); #octetsent / OS $var(packetsent) = $(var(xrtpstat){s.select,3, }); #PS $var(octetrcv) = $(var(xrtpstat){s.select,8, }); #OR $var(packetrcv) = $(var(xrtpstat){s.select,10, }); #PR $var(errorsent) = $(var(xrtpstat){s.select,5, }); $var(errorrcv) = $(var(xrtpstat){s.select,12, }); if ($var(octetsent) != "" || $var(packetsent) != "") { xlog(,"L_INFO", "WITHINDLG PS=$var(rtp0),PR=$var(rtp1),PL=$var(rtp3) \n"); xlog(,"L_INFO", "WITHINDLG xrtpstats=$var(xrtpstat) \r\n"); xlog(,"L_INFO", "WITHINDLG rtpstats=$rtpstat \r\n"); xlog(,"L_INFO", "WITHINDLG X-RTP-Stat: PS=$var(packetsent),OS=$var(octetsent),PR=$var(packetrcv),OR=$var(octetrcv),PL=$var(errorsent) \r\n"); #append_hf("P-RTP-Stat: $rtpstat\r\n"); append_hf("X-RTP-Stat: PS=$var(packetsent);OS=$var(octetsent);PR=$var(packetrcv);OR=$var(octetrcv);PL=$var(errorsent);JI=600;LA=40;\r\n"); } Other question: Where has this to be placed? Only on bye messages? KR, Oli

1 0

Change from in a failure route
by Nelson Migliaro 23 Nov '16

23 Nov '16

Hello, My sip vendor sends me a 403 error when the caller id (CLI) sent is not valid. I would like to set up a failure route to capture that 403 and set up a default CLI. I tried to use a uac_replace_from() in a failure route but I get: "Command cannot be used in the block" Does anyone know how can I change the CLI in a failure route how how can achieve the default CLI? Thank you

2 1

Force OpenSER to send a Stop radius accounting record in failure route.
by Jan Hazenberg 23 Nov '16

23 Nov '16

Hi all, I have a issue here on a old OpenSER 1.3 system. We have some clients that respond to a BYE message with a 481 resulting in a missing radius accounting record. I have added some code to the failure route to force accounting even if a 481 response in received: if (t_check_status("481") && $rm == "BYE") { xlog("L_INFO", "INFO: [FAILURE] 481 response detected!! Forcing accounting! - SRC=$si:$sp R=$ru ID=$ci FU=$fu M=$rm S=$rs\n"); acc_rad_request("BYE with 481 response"); } Now OpenSER sends a accounting record, but this has the attribute Acct-Status-Type set to Failed. This gives a problem with our accounting service and i would like to change it to a normal Stop record. Is there a way to change this and force OpenSER/Kamailio to send a Stop record instead of a Failed record? Thanks, Jan

2 3

SDP Codec not removed with RTPengine - but with rtpproxy it worked
by Oliver Roth 23 Nov '16

23 Nov '16

Hi there I have the following problem - I need to remove a codec in the initial INVITE. This happens since I changed from rtpproxy to rtpengine. I changed all rtpproxy_manage() to rtpengine_manage(). Originating INVITE with the "clearmode" m=audio 9196 RTP/AVP 8 0 125 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:125 CLEARMODE/8000 a=rtpmap:101 telephone-event/8000 I do the following in the config route[SDP] { xlog(, "L_INFO", "SDP: request method $rm"); # remove CLEARMODE if Colt if($avp(s:todirection) =~"^MyCarrier+") { if (is_method("INVITE")) { xlog(, "L_INFO", "SDP remove: request method $rm"); msg_apply_changes(); sdp_remove_codecs_by_name("CLEARMODE"); #sdp_remove_codecs_by_id("125"); } } } This block gets hit as I can see with the xlog entry. In the sent INVITE the "Clearmode" is still in the sdp header a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:125 CLEARMODE/8000 a=rtpmap:101 telephone-event/8000 So with rtpproxy - it worked With rtpengine not any more ... Any idea? KR, Oli

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IMS related : ims_usrloc_pcscf : "lookup_check_received" setting is useless now
by Dragos Oancea 23 Nov '16

23 Nov '16

Hi The setting "lookup_check_received" is documented here : http://www.kamailio.org/docs/modules/4.4.x/modules/ims_usrloc_pcscf.html#id… But looking through the code I saw it's not used anymore, although it can be passed as a parameter. I was wondering if this is an accident or something intentional. I found some more details here: http://lists.sip-router.org/pipermail/sr-dev/2014-June/024371.html Cheers, Dragos

2 1

async_workers vs. async workers modparam in case of async_route()
by Alex Balashov 22 Nov '16

22 Nov '16

Hi, If I am using async_route() with delay, what is the relationship between the core async_workers parameter value and the async module 'workers' modparam? This doc change was intended to clarify it, I think: https://www.mail-archive.com/sr-dev@lists.sip-router.org/msg25002.html But it does not for me: > Number of worker processes to be started to handle the > asynchronous tasks for async_route() and async_sleep(). So, does that mean the module starts its own processes for these cases? If so, does that mean I don't need core async_workers at all? Or do I? If I do, what happens if I have fewer core async_workers than modparam workers? What if I have more? Should I aim to keep them in sync? Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

log_prefix ignored in resumed routes
by Alex Balashov 22 Nov '16

22 Nov '16

Hi, I have this log_prefix: log_prefix="[$ci $mt] " Which means xlog("L_INFO", "xyz") normally results in log entries like this: Nov 22 00:29:50 allegro-4.evaristesys.com /usr/local/sbin/kamailio[27625]: [1-27702(a)10.150.20.6 1] INFO: xyz But if xlog("L_INFO", "xyz") happens to occur in a route[] block that is resumed after async_route()—I have not tested other async functions—there is no prefix: Nov 22 00:29:52 allegro-4.evaristesys.com /usr/local/sbin/kamailio[27678]: INFO: xyz -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

flags documentation
by Alex Balashov 22 Nov '16

22 Nov '16

Hi, I have this set up in my configuration boilerplate: -- flags DUMMY : 1, PDD_DAMPEN : 2, PROXY_MEDIA_SET : 3, #!ifdef WITH_NAT_TRAVERSAL SIPPING : 4, #!endif DUMMY2 : 12 -- I then use these aliases in transaction flags, e.g. setflag(PDD_DAMPEN). But I can honestly say I don't remember how I arrived at this approach. I wrote it somewhere in early 3.x days. However, when I look at the core documentation for the 'flags' declaration, all I find is: http://www.kamailio.org/wiki/cookbooks/4.4.x/core#flags Or namely: -- flags Alias name: bool -- And there is no entry for "bool". So, I couldn't have possibly arrived at this usage from the documentation. Can anyone help me reverse-engineer this historical phenomenon? And, given that the documentation doesn't support it particularly, is it still an encouraged practice? Or should one use #!defined constants instead, as in the stock config? Thanks, -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

Message flags preservation for replies in onsend_route
by Alex Balashov 22 Nov '16

22 Nov '16

Hi, I have -- flags BLAH : 1, DUMMY : 12 onsend_route_reply=yes route { ... setflag(BLAH); if(!t_relay()) sl_reply_error(); } onsend_route { if(is_reply() && isflagset(BLAH)) do_things(); } -- My discovery has been that isflagset(BLAH) always == FALSE _unless_ an onreply_route has been armed for the request, regardless of whether it is necessary. In other words, the following causes isflagset(BLAH) to == TRUE (if set, when appropriate) in the onsend_route for the reply: -- flags BLAH : 1, DUMMY : 12 onsend_route_reply=yes onreply_route[BOGUS] { exit; } route { ... setflag(BLAH); t_on_reply("BOGUS"); if(!t_relay()) sl_reply_error(); } onsend_route { if(is_reply() && isflagset(BLAH)) do_things(); } -- But without the onreply_route, it doesn't work. -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

1 0

Registering Trunk help please
by Andrei Mahalean 22 Nov '16

22 Nov '16

Hello, I am having a bit of trouble finding information on how to setup a registering trunk with Kamailio. By registering trunk, I mean a trunk between a PBX and Kamailio, of say 15 DID's, where the main DID (Pilot) registers with Kamailio and all the other 14 DID are able to make/receive calls without having to register all DID's individually. If the pilot DID registration goes away, the trunk is then considered to be unreachable/down. I have seen this in the past with a Broadsoft setup, but I can't figure out a way to do it with Kamailio. I have my pilot DID setup and registered with Kamailio ok and I can make calls in and out through my PSTN gateway, but I'm unsure how to group all the other DID's so they are matched against the pilot DID. Is this possible? Any help is appreciated. Regards, Andrei. -- Regards, Andrei.

2 1

Roadmap to Kamailio v5.0
by Daniel-Constantin Mierla 21 Nov '16

21 Nov '16

Hello, following the last IRC devel meeting, I am writing the summary of the main points being discussed there and what's planned to get to Kamailio v5.0. 1) Freezing should be done either before Chrismas holidays or at beginning of January 2017 to aim meeting the deadline for next Debian stable distro 2) Restructuring the source tree To be done in the weeks before freezing, so eventual fixes won't have hard time to be backported to 4.4. This means it will be done in December. Open issues: a) anyone aware of a tool that can update include directives upon relocation of the file? b) who is interested to join an eventual online session to perform the restructuring? C coding is not a must here, but more knowledge of sed/awk/perl/python/... and Makefiles ... 3) Defining a template for generating main file of a module The scope is to have a descriptive file for the module where one lists exported symbols (config params, functions, kemi functions, rpc/mi commands, etc.) which along with a template will generate the c code. The effective implementation of the functions/commands will be done separately, in a different file (e.g, the main file that is generated can have references to parameter variables with extern ... and will also generate the .h which will have only the prototypes for the functions). The main goal here is to get a way of easily generated an index of exported symbols that can be used to discover missing documentation. Right now we sort of do 'gentlemen agreement' on documenting functions and parameters, but many rpc commands are not documented (they are extracted with some tool scanning compiled objects). Also, same concept as with c code can be used to generate some stubs for documentation that can be included in docbook xml files. Few proposals of templating engines were mentioned, coming in my mind now being jinja2 and mustache. Anyone else with more suggestions? Or even propose alternatives of what can be used to solve this relation between code and docs? Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com

1 0

usage of app_mono : how to communicate results back to config file
by Pranathi Venkatayogi 21 Nov '16

21 Nov '16

Hi, I am using app_mono module and building my logic in c# app. I am unable to understand how to send the decision back to kamalio.cfg. More precisely: I would like my routing logic to allow specific user names though they are not registered as subscribers. Like: translate(a)myip.cyracom.com<mailto:translate@myip.cyracom.com> I have the logic in c# app which takes $ru and validates such that: Host is myip.cyracom.com User is translate How do I communicate the decision if it is valid or not back to my config script. Do I need to introduce new pseudo variable for this purpose? Thanks, Pranathi

2 1

Is there a way to detect inactive MSRP chat sessions without using SIP session timers
by Pranathi Venkatayogi 18 Nov '16

18 Nov '16

Hi, I am using MSRP feature of Blink client. I want to track when the chat starts and ends so I can free up server resources. Apparently Blink client does not send BYE when MSRP chat window closes. It does not support SIP session timers even. So potential use of SIP session timers (sst and dialog modules) is ruled out. Only possible way is to track every message that goes through the server and keep the session alive. If no message comes within “expiry time period”, server can send “bye” to all parties and tear up the session. Can you please comment if there are other approaches to solve this problem? If not, are they any modules that will help me implement the above approach easily? Thanks Pranathi

2 2

tcp_set_connection_lifetime - does not close the connection immediately
by Andy 18 Nov '16

18 Nov '16

I am calling tcp_set_connection_lifetime(conid, 1), which I understand is one sec after now, but I see it taking upto 60 secs to kill the connection, is there any other timer which has a larger interval which might affect this. -- View this message in context: http://sip-router.1086192.n5.nabble.com/tcp-set-connection-lifetime-does-no… Sent from the Users mailing list archive at Nabble.com.

2 2

Htable sht_rm_name_re usage
by Grant Bagdasarian 18 Nov '16

18 Nov '16

Hello Community, We were facing a weird issue in one of our Kamailio servers where custom htable entries were removed which did not belong to a particular call. Background Information: We use htable to store the call-id and a uuid when an INVITE message is received and remove these entries when the call is completed/failed (so on CANCEL, BYE, 4XX, 5XX, 6XX, etc). In this case the function sht_rm_name_re was causing incremented call-id's belonging to different calls to be removed from the htable, since the removal is based on regex and not the exact key. On INVITE we do this: $var(reference) = $uuid(g); $sht(calls=>$ci) = $ci; $sht(calls=>$ci::reference) = $var(reference); On BYE/4XX,5XX,6XX, etc we did this: sht_rm_name_re("calls=>$ci"); sht_rm_name_re("calls=>$ci::reference"); SIPP generates incremented call-id's, like this: 1-14875(a)1.0.0.1<mailto:1-14875@1.0.0.1>. When you generate enough calls, for instance 30 calls, the 21st call will have the call-id 21-14875(a)1.0.0.1<mailto:21-14875@1.0.0.1>. If for instance the BYE for the call-id 1-14875(a)1.0.0.1<mailto:1-14875@1.0.0.1> arrives before 21-14875(a)1.0.0.1<mailto:21-14875@1.0.0.1> , both htable entries will be removed by the sht_rm_name_re function, since it matches based on regex. To fix this we made sure the function matches on the exact keys by doing this: $var(callid) = '^' + $ci + '$'; $var(callid_reference) = '^' + $ci + '::reference' + '$'; sht_rm_name_re("calls=>$var(callid)"); sht_rm_name_re("calls=>$var(callid_reference)"); Hope this may help someone in the future facing the same issue. Regards, Grant Bagdasarian CM

2 1

Kamailio for SIP Capture With Registration
by Ali Dakroub~Vanrise Technical Support 18 Nov '16

18 Nov '16

Hello, Kindly note that I am configuring Kamailio for SIP capture with "Registration" module (registrar.so). Before loading the "Sip Capture" module (sipcapture.so) into Kamailio.cfg file, the registration of the SIP Phone (X-lite) is successful. Yet, when we add the "Sip Capture" module (sipcapture.so) into Kamailio.cfg file with the below parameters: modparam("sipcapture", "db_url", "postgres://postgres:postgres@localhost/kamailio") modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "table_name", "sip_capture") modparam("sipcapture", "raw_moni_capture_on", 1) /* IP to listen. Port/Portrange apply only on mirroring port capturing */ modparam("sipcapture", "raw_socket_listen", "10.0.10.1:5060") /* Name of interface to bind on raw socket */ modparam("sipcapture", "raw_interface", "eth0") /* children for raw socket */ modparam("sipcapture", "raw_sock_children", 4) /* insert delayed */ #modparam("sipcapture", "db_insert_mode", 1) /* Linux only */ /* Promiscious mode RAW socket. Mirroring port. */ modparam("sipcapture", "promiscious_on", 1) /* activate BPF */ modparam("sipcapture", "raw_moni_bpf_on", 1) With the following routing logic: ####### Routing Logic ######## request_route { #sip_capture(); # CANCEL processing # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # user location service } route[REGISTRAR] { if (is_method("REGISTER")) { if (!save("location")) { sl_reply_error(); } exit; } } The registration of the SIP Phone fails and we are getting the following error in Kamailio.log file: 2016-11-16T09:49:28.809247-08:00 ubuntu /usr/local/sbin/kamailio[10612]: ERROR: <core> [udp_server.c:550]: udp_send(): sendto(sock,0x7f60c7593910,481,0,192.168.26.4:14661,16): Bad file descriptor(9) 2016-11-16T09:49:28.856080-08:00 ubuntu /usr/local/sbin/kamailio[10612]: ERROR: sl [../../forward.h:192]: msg_send(): udp_send failed 2016-11-16T09:49:28.860251-08:00 ubuntu /usr/local/sbin/kamailio[10612]: ERROR: registrar [reply.c:734]: reg_send_reply(): failed to send 200 OK It would be highly appreciated to check what went wrong here. Thanks in advance. Ali Dakroub

2 1

Kamctl stats
by Alexandru Covalschi 18 Nov '16

18 Nov '16

Hello list, Where can I found any information to completely understand what do values returned by 'kamctl stats' represent? Cheers, Alex

2 1

Re: [SR-Users] msilo
by Slava Bendersky 17 Nov '16

17 Nov '16

Hello Daniel, Yes that good alternative. I will use sql to fetch or insert staff. Slava From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 16 November, 2016 04:37:24 Subject: Re: [SR-Users] msilo Hello, as an alternative, I think it should work by using sql_query(...) to insert whatever values one would like to have in msilo table. This might be a quick workaround until someone will code the extension to get the body via a function parameter. Cheers, Daniel On 15/11/16 16:07, Federico Cabiddu wrote: Hi Slava, if I understand correctly your scenario, you could call t_on_failure before relaying the MESSAGE. In this way you should have access to the original message in the failure route triggered after receiving a final reply and you should be able to store it with the regular msilo functions. Hope this helps. Cheers, Federico On 15 Nov 2016 3:47 pm, "Slava Bendersky" < volga629(a)skillsearch.ca > wrote: BQ_BEGIN Hello Daniel, I am not programmer, but I want place request if kaamilio devs can complete it . Thanks Slava. From: "Daniel-Constantin Mierla" < miconda(a)gmail.com > To: "sr-users" < sr-users(a)lists.sip-router.org > Sent: Tuesday, 15 November, 2016 03:00:19 Subject: Re: [SR-Users] msilo Hello, for me it is fine to add a new parameter to m_store() that takes the body content. Are you asking for more details of how can be done? Cheers, Daniel On 13/11/16 07:08, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, I asking question/request to add improvement for msilo module where will have ability set $avp(i:body) in mod params. That will allow use it in m_store() by specifying body message $avp. In my case registration is on B2BUA so kamailio pass through/proxy to destination. In order to check user status online/offline kamailio wait for reply from B2BUA on MESSAGE status if it return error 503 that mean user offline and m_store(0 should store message in db, but the issue that error 503 is in final stage and MESSAGE body is not available any more. The improvement will allow specify body $var in m_store() so new way will be m_store($tu, $avp(body)). Any help thank you. Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Possible memory leak in mysql driver
by Alexandru Covalschi 17 Nov '16

17 Nov '16

Hello list, We’re using dev version of Kamailio: version: kamailio 5.0.0-dev4 (x86_64/linux) ff63e5 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: ff63e5 compiled on 15:46:49 May 31 2016 with gcc 4.9.2 Sometimes we encounter such issue: Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: <core> [db_row.c:114]: db_allocate_row(): no private memory left Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql [km_row.c:57]: db_mysql_convert_row(): could not allocate row Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql [km_res.c:188]: db_mysql_convert_rows(): error while converting row #16 Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql [km_res.c:217]: db_mysql_convert_result(): error while converting rows Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql [km_dbase.c:261]: db_mysql_store_result(): error while converting result Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: <core> [db_query.c:139]: db_do_query_internal(): error while storing result Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: permissions [trusted.c:91]: reload_trusted_table(): failed to query database Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: <core> [db_row.c:114]: db_allocate_row(): no private memory left Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql [km_row.c:57]: db_mysql_convert_row(): could not allocate row Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: db_mysql [km_dbase.c:444]: db_mysql_fetch_result(): error while converting row #15 Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: ERROR: htable [ht_db.c:234]: ht_db_load_table(): Error while fetching result Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12276]: : <core> [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already freed pointer (0x7f5ebda8ae18), called from db_mysql: km_dbase.c: db_mysql_free_result(305), first free db_mysql: km_dbase.c: db_mysql_free_result(305) - aborting Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12281]: CRITICAL: <core> [pass_fd.c:275]: receive_fd(): EOF on 16 Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12268]: ALERT: <core> [main.c:739]: handle_sigs(): child process 12276 exited by a signal 6 Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12268]: ALERT: <core> [main.c:742]: handle_sigs(): core was not generated Nov 9 23:30:01 sbc01 /usr/local/sbin/kamailio[12268]: INFO: <core> [main.c:754]: handle_sigs(): terminating due to SIGCHLD The thing is we heavily use mysql module, but only to update the in-memory tables by kamcmd. Each N minutes a special script updates the trusted,address and htable executing kamcmd. Kamailio (and kamcmd as well) talks only with localhost mysql server. What I saw when encountered that issue on a live machine is that issue happens only with one of child processes, any other are ok. Interesting thing is that happens at the same time with machines on the same «set», I mean that issue happened simultaneously with two our test machines which actually didn’t have any load on them. The common thing between those machines is that they are in same subnet and local mysql databases are filled by scripts which query same external db. I can’t confirm if there were or there weren’t any networking issues at that time with those machines, but as soon as kamcmd queries localhost that shouldn’t be the source of the issue. So my questions are: 1. Has anyone encountered such thing? 2. Maybe the issue is already localized so it has sense to update? We actually use that on production (pls don’t throw too much rocks at me), so maintenance should be properly planned and I must be sure update won’t break anything. 3. If update is proposed - how to do it? I mean - follow the guide https://www.kamailio.org/wiki/install/devel/git or there are some other tips? I suppose in ideal world I don’t even stop the binary, only restart after make all && make install are done, as everything is in-memory. Am I correct? 4. When can we expect stable 5.0 version? (at least tell if it’s months/years) Thanks in advance!

2 3

Error message not seen before
by Kevin Greene 17 Nov '16

17 Nov '16

Hi fellow Kamailio people: I was using the Siremis (v2.0) to manage my Kamailio application. I received a message labeled "System Internal Error" It seems, assuming I understand it correctly, that a table called "kamailio.uacreg" does not exist in the database. I receive this error when I click on the following: Main System > SIP Admin Menu > Server Services > UACReg List function. Any help in pointing me in the right direction would be great. Thanks. Kevin 2016-11-16 22:29:23 (GMT)] An exception occurred while executing this script: Error message: #0, Error in query:: SELECT T0.id, T0.l_uuid, T0.l_username, T0.l_domain, T0.r_username, T0.r_domain, T0.realm, T0.auth_username, T0.auth_password, T0.auth_proxy, T0.expires FROM uacreg T0 LIMIT 10. SQLSTATE[42S02]: Base table or view not found: 1146 Table 'kamailio.uacreg' doesn't exist Script name and line number of error: /var/www/siremis/openbiz/bin/data/BizDataObj_Lite.php:447 *function:* _run_search ( Array(2) ) @ /var/www/siremis/openbiz/bin/data/BizDataObj_Lite.php 243 *function:* fetch ( ) @ /var/www/siremis/openbiz/bin/easy/EasyForm.php 690 *function:* fetchDataSet ( ) @ /var/www/siremis/openbiz/bin/easy/FormRenderer.php 69 *function:* renderSmarty ( Object(EasyForm), "/var/www/siremis/siremis/modules/ser/template/grid.tpl" ) @ /var/www/siremis/openbiz/bin/easy/FormRenderer.php 45 *function:* render ( Object(EasyForm) ) @ /var/www/siremis/openbiz/bin/easy/EasyForm.php 1648 *function:* renderHTML ( ) @ /var/www/siremis/openbiz/bin/easy/EasyForm.php 1538 *function:* render ( ) @ /var/www/siremis/openbiz/bin/easy/ViewRenderer.php 83 *function:* renderSmarty ( Object(EasyView), "/var/www/siremis/siremis/modules/ser/template/view.tpl" ) @ /var/www/siremis/openbiz/bin/easy/ViewRenderer.php 40 *function:* render ( Object(EasyView) ) @ /var/www/siremis/openbiz/bin/easy/EasyView.php 274 *function:* _render ( ) @ /var/www/siremis/openbiz/bin/easy/EasyView.php 237 *function:* render ( ) @ /var/www/siremis/openbiz/bin/BizController.php 221 *function:* renderView ( "ser.view.UacregListView", "", "", Null, "" ) @ /var/www/siremis/openbiz/bin/BizController.php 107 *function:* dispatchRequest ( ) @ /var/www/siremis/openbiz/bin/BizController.php 32 *function:* include_once ( "/var/www/siremis/openbiz/bin/BizController.php" ) @

2 1

[CFP] reminder! FOSDEM RTC dev-room talks: deadline tomorrow
by FOSDEM RTC Team 16 Nov '16

16 Nov '16

Reminder: speaker's deadline tomorrow, 17 November at 23:59 UTC The Free RTC dev-room has already received some really exciting talk proposals but there is still time for people to propose talks or encourage friends or colleagues to speak. Many other dev-rooms also have a deadline in the next few days and if your topic is applicable to more than one dev-room, you are welcome to make more than one submission. Please contact us or put a note in the memo field at the top of the talk proposal if you do that. All projects are encouraged to consider making a lightning talk too, it is an excellent opportunity to get exposure for your project: even though you only have 15 minutes, it can be a much larger and more diverse audience than in some dev-rooms. For full details, please see the original call for participation: https://danielpocock.com/fosdem-2017-rtc-cfp We invite all potential speakers and participants to discuss the selection process and other aspects of FOSDEM on the Free-RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc

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Re: [SR-Users] msilo
by Federico Cabiddu 16 Nov '16

16 Nov '16

Hi Slava, if I understand correctly your scenario, you could call t_on_failure before relaying the MESSAGE. In this way you should have access to the original message in the failure route triggered after receiving a final reply and you should be able to store it with the regular msilo functions. Hope this helps. Cheers, Federico On 15 Nov 2016 3:47 pm, "Slava Bendersky" <volga629(a)skillsearch.ca> wrote: Hello Daniel, I am not programmer, but I want place request if kaamilio devs can complete it . Thanks Slava. ------------------------------ *From: *"Daniel-Constantin Mierla" <miconda(a)gmail.com> *To: *"sr-users" <sr-users(a)lists.sip-router.org> *Sent: *Tuesday, 15 November, 2016 03:00:19 *Subject: *Re: [SR-Users] msilo Hello, for me it is fine to add a new parameter to m_store() that takes the body content. Are you asking for more details of how can be done? Cheers, Daniel On 13/11/16 07:08, Slava Bendersky wrote: Hello Everyone, I asking question/request to add improvement for msilo module where will have ability set $avp(i:body) in mod params. That will allow use it in m_store() by specifying body message $avp. In my case registration is on B2BUA so kamailio pass through/proxy to destination. In order to check user status online/offline kamailio wait for reply from B2BUA on MESSAGE status if it return error 503 that mean user offline and m_store(0 should store message in db, but the issue that error 503 is in final stage and MESSAGE body is not available any more. The improvement will allow specify body $var in m_store() so new way will be m_store($tu, $avp(body)). Any help thank you. Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] msilo
by Slava Bendersky 16 Nov '16

16 Nov '16

Hello Everyone, I am trying just forward MESSAGE through kamailio and I reported issue with mod_sms in freeswitch https://freeswitch.org/jira/browse/FS-9701. So I was need add work around on kamailio side. The issue that my setup MSILO should work in intercept mode meaning check when user online of offline and act as m_store() or m_dump() without rely on lookup("location");, because registration on B2BUA side. If m_store() will have additional parameter to store body this will allow check user state and send when need it. On reply I put something like this to fix To, as work around and forward MESSAGE to client, but when client send MESSAGE kamailio receive on public interface and push out of public interface instead RELAY through local interface and then to B2BUA. route[MESSAGE_FORWARD] { xlog("L_INFO", "Incoming new SMS [$rm] from $tU --> $td\n"); if(fnmatch("$rs", "202") && fnmatch("$rr", "Accepted") && allow_trusted("$si", "$proto")) { $avp(from) = $fu; $avp(to) = $tu; $avp(new_to) = "<" + $avp(to) + ">" + ";messagetype=SMS"; xlog("L_INFO", "Message Accepted by B2BUA --> [$rs] with [$rr] from [$avp(from)] to [$avp(to)]\n"); if(fnmatch("$fU", "offline")) { xlog("L_INFO", "OFFLINE SMS: from [$fU] user --> $fU\n"); $avp(from) = $(avp(to){uri.user}); xlog("L_INFO", "OFFLINE SMS: Searching destination user $avp(oexten)\n"); sql_pvquery("cb", "select contact from location where username = '$avp(from)'","$avp(dst)"); xlog("L_INFO", "OFFLINE SMS: set new destination --> $avp(dst)\n"); $avp(new_to) = "<" + $(avp(dst){tobody.uri}) + ">" + ";messagetype=SMS"; } $sht(a=>to) = $avp(new_to); xlog("SMS from --> $fU domain $fd\n"); xlog("-------------------------------------\n"); xlog("L_INFO", "FROM --> $avp(from)\n"); xlog("L_INFO", "OLD_TO --> $avp(to)\n"); xlog("L_INFO", "NEW_TO --> $avp(new_to)\n"); } if(compare_ips("$td", "10.18.130.27")) { xlog("L_INFO", "Message from B2BUA contain domain --> $td . Updating...\n"); $avp(new_dst) = $(hdr(Route){param.value,received}); xlog("L_INFO", "New destination --> $avp(new_dst)\n"); $avp(new_to) = "<" + $avp(new_dst) + ">" + ";messagetype=SMS"; $sht(a=>to) = $avp(new_to); remove_hf("To"); insert_hf("To: $sht(a=>to)\r\n", "To"); $du = $sht(a=>to); } } From: "Federico Cabiddu" <federico.cabiddu(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Tuesday, 15 November, 2016 11:07:44 Subject: Re: [SR-Users] msilo Hi Slava, if I understand correctly your scenario, you could call t_on_failure before relaying the MESSAGE. In this way you should have access to the original message in the failure route triggered after receiving a final reply and you should be able to store it with the regular msilo functions. Hope this helps. Cheers, Federico On 15 Nov 2016 3:47 pm, "Slava Bendersky" < volga629(a)skillsearch.ca > wrote: Hello Daniel, I am not programmer, but I want place request if kaamilio devs can complete it . Thanks Slava. From: "Daniel-Constantin Mierla" < miconda(a)gmail.com > To: "sr-users" < sr-users(a)lists.sip-router.org > Sent: Tuesday, 15 November, 2016 03:00:19 Subject: Re: [SR-Users] msilo Hello, for me it is fine to add a new parameter to m_store() that takes the body content. Are you asking for more details of how can be done? Cheers, Daniel On 13/11/16 07:08, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, I asking question/request to add improvement for msilo module where will have ability set $avp(i:body) in mod params. That will allow use it in m_store() by specifying body message $avp. In my case registration is on B2BUA so kamailio pass through/proxy to destination. In order to check user status online/offline kamailio wait for reply from B2BUA on MESSAGE status if it return error 503 that mean user offline and m_store(0 should store message in db, but the issue that error 503 is in final stage and MESSAGE body is not available any more. The improvement will allow specify body $var in m_store() so new way will be m_store($tu, $avp(body)). Any help thank you. Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] msilo
by Slava Bendersky 15 Nov '16

15 Nov '16

Hello Daniel, I am not programmer, but I want place request if kaamilio devs can complete it . Thanks Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Tuesday, 15 November, 2016 03:00:19 Subject: Re: [SR-Users] msilo Hello, for me it is fine to add a new parameter to m_store() that takes the body content. Are you asking for more details of how can be done? Cheers, Daniel On 13/11/16 07:08, Slava Bendersky wrote: Hello Everyone, I asking question/request to add improvement for msilo module where will have ability set $avp(i:body) in mod params. That will allow use it in m_store() by specifying body message $avp. In my case registration is on B2BUA so kamailio pass through/proxy to destination. In order to check user status online/offline kamailio wait for reply from B2BUA on MESSAGE status if it return error 503 that mean user offline and m_store(0 should store message in db, but the issue that error 503 is in final stage and MESSAGE body is not available any more. The improvement will allow specify body $var in m_store() so new way will be m_store($tu, $avp(body)). Any help thank you. Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 15 Nov '16

15 Nov '16

Hello I have tried $Rp before sending this to mailing list, but with $Rp I have different values for example 188, 6847, 3163, 9175 for every call ?! Don't know where are they come from, because in URI I see :6060, and using $rp give my 6060 for examples with $Rp. Only have problem with UAC Panasonic-MPR12-V004.41009/VSIPGW-V3.0000 with doesn't send $rp in URI. Greetings

3 2

RTPProxy benchmark
by Gholamreza Sabery 15 Nov '16

15 Nov '16

I managed to create about 1900 concurrent calls using a single Kamailio and RTPProxy server. But after this number RTPProxy returns 0 and the following error is shown in the Kamailio log files: incorrect port 0 in reply from rtp proxy What is the problem here? Also number of file descriptors that RTPProxy can use are set to a million.

3 9

How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 15 Nov '16

15 Nov '16

Hello My point is to log INVITEs in my Network. My SIP Network work on port 6060. Im am using HEP to do that, and hep clients that are listening on port range 5060-6066 to detect some SIP attack to port 5060 and others. But when I have attack to port 5060 I don't want to insert that INVITE to my "good traffic" table, but place it to "fraud" table. Some devices like Panasonic PBX send INVITE to port 6060 so it is "good traffic" but there is no port 6060 in URI, so I can't detect it in right way, because $dp or $rp are 5060 by default. This kamailio work as capture server as "promiscuous_on", so I can't use any "force_rport" etc. because I am only listening. Tcpdump for that SIP session show me that client send traffic to 6060, but I can't get that information from INVITE header. Greetings

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[CFP] FOSDEM 2017, RTC devroom, speakers, volunteers neeeded
by FOSDEM RTC Team 15 Nov '16

15 Nov '16

FOSDEM is one of the world's premier meetings of free software developers, with over five thousand people attending each year. FOSDEM 2017 takes place 4-5 February 2017 in Brussels, Belgium. https://fosdem.org This email contains information about: - Real-Time communications dev-room and lounge, - speaking opportunities, - volunteering in the dev-room and lounge, - related events around FOSDEM, including the XMPP summit, - social events (the legendary FOSDEM Beer Night and Saturday night dinners provide endless networking opportunities), - the Planet aggregation sites for RTC blogs Call for participation - Real Time Communications (RTC) ======================================================= The Real-Time dev-room and Real-Time lounge is about all things involving real-time communication, including: XMPP, SIP, WebRTC, telephony, mobile VoIP, codecs, peer-to-peer, privacy and encryption. The dev-room is a successor to the previous XMPP and telephony dev-rooms. We are looking for speakers for the dev-room and volunteers and participants for the tables in the Real-Time lounge. The dev-room is only on Saturday, 4 February 2017. The lounge will be present for both days. To discuss the dev-room and lounge, please join the FSFE-sponsored Free RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc To be kept aware of major developments in Free RTC, without being on the discussion list, please join the Free-RTC Announce list: http://lists.freertc.org/mailman/listinfo/announce Speaking opportunities ---------------------- Note: if you used FOSDEM Pentabarf before, please use the same account/username Real-Time Communications dev-room: deadline 23:59 UTC on 17 November. Please use the Pentabarf system to submit a talk proposal for the dev-room. On the "General" tab, please look for the "Track" option and choose "Real-Time devroom". https://penta.fosdem.org/submission/FOSDEM17/ Other dev-rooms and lightning talks: some speakers may find their topic is in the scope of more than one dev-room. It is encouraged to apply to more than one dev-room and also consider proposing a lightning talk, but please be kind enough to tell us if you do this by filling out the notes in the form. You can find the full list of dev-rooms at https://www.fosdem.org/2017/schedule/tracks/ and apply for a lightning talk at https://fosdem.org/submit Main track: the deadline for main track presentations is 23:59 UTC 31 October. Leading developers in the Real-Time Communications field are encouraged to consider submitting a presentation to the main track at https://fosdem.org/submit First-time speaking? -------------------- FOSDEM dev-rooms are a welcoming environment for people who have never given a talk before. Please feel free to contact the dev-room administrators personally if you would like to ask any questions about it. Submission guidelines --------------------- The Pentabarf system will ask for many of the essential details. Please remember to re-use your account from previous years if you have one. In the "Submission notes", please tell us about: - the purpose of your talk - any other talk applications (dev-rooms, lightning talks, main track) - availability constraints and special needs You can use HTML and links in your bio, abstract and description. If you maintain a blog, please consider providing us with the URL of a feed with posts tagged for your RTC-related work. We will be looking for relevance to the conference and dev-room themes, presentations aimed at developers of free and open source software about RTC-related topics. Please feel free to suggest a duration between 20 minutes and 55 minutes but note that the final decision on talk durations will be made by the dev-room administrators. As the two previous dev-rooms have been combined into one, we may decide to give shorter slots than in previous years so that more speakers can participate. Please note FOSDEM aims to record and live-stream all talks. The CC-BY license is used. Volunteers needed ================= To make the dev-room and lounge run successfully, we are looking for volunteers: - FOSDEM provides video recording equipment and live streaming, volunteers are needed to assist in this - organizing one or more restaurant bookings (dependending upon number of participants) for the evening of Saturday, 4 February - participation in the Real-Time lounge - helping attract sponsorship funds for the dev-room to pay for the Saturday night dinner and any other expenses - circulating this Call for Participation to other mailing lists See the mailing list discussion for more details about volunteering: https://lists.fsfe.org/pipermail/free-rtc/2016-October/000285.html Related events - XMPP and RTC summits ===================================== The XMPP Standards Foundation (XSF) has traditionally held a summit in the days before FOSDEM. There is discussion about a similar summit taking place on 2 and 3 February 2017 http://wiki.xmpp.org/web/Summit_21 - please join the mailing list for details: http://mail.jabber.org/mailman/listinfo/summit We are also considering a more general RTC or telephony summit, potentially in collaboration with the XMPP summit. Please join the Free-RTC mailing list and send an email if you would be interested in participating, sponsoring or hosting such an event. Social events and dinners ========================= The traditional FOSDEM beer night occurs on Friday, 3 February. On Saturday night, there are usually dinners associated with each of the dev-rooms. Most restaurants in Brussels are not so large so these dinners have space constraints and reservations are essential. Please subscribe to the Free-RTC mailing list for further details about the Saturday night dinner options and how you can register for a seat: https://lists.fsfe.org/mailman/listinfo/free-rtc Spread the word and discuss =========================== If you know of any mailing lists where this CfP would be relevant, please forward this email. If this dev-room excites you, please blog or microblog about it, especially if you are submitting a talk. If you regularly blog about RTC topics, please send details about your blog to the planet site administrators: All projects http://planet.freertc.org planet(a)freertc.org XMPP http://planet.jabber.org ralphm(a)ik.nu SIP http://planet.sip5060.net planet(a)sip5060.net (Español) http://planet.sip5060.net/es/ planet(a)sip5060.net Please also link to the Planet sites from your own blog or web site as this helps everybody in the free real-time communications community. Contact ======= For any private queries, contact us directly using the address fosdem-rtc-admin(a)freertc.org and for any other queries please ask on the Free-RTC mailing list: https://lists.fsfe.org/mailman/listinfo/free-rtc The dev-room administration team: Saúl Ibarra Corretgé <saghul(a)gmail.com> Iain R. Learmonth <irl(a)debian.org> Ralph Meijer <ralphm(a)ik.nu> Daniel-Constantin Mierla <miconda(a)gmail.com> Daniel Pocock <daniel(a)pocock.pro>

2 1

INVITE timeout timer
by Diogenes Aviles 15 Nov '16

15 Nov '16

I have changed the following tm parameters in order to have an timeout INVITE equal to 20 sec (20000 ms). # ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 0) # default retransmission timeout: 20sec modparam("tm", "fr_timer", 20000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000) However, the timeout is ever 3 min (180 sec.) In accord to tm module the timeout is defined only by "r_timer" Then, I don't know where is the problem. Thanks a lot for your help. -- Diogenes

3 2

msilo
by Slava Bendersky 15 Nov '16

15 Nov '16

Hello Everyone, I asking question/request to add improvement for msilo module where will have ability set $avp(i:body) in mod params. That will allow use it in m_store() by specifying body message $avp. In my case registration is on B2BUA so kamailio pass through/proxy to destination. In order to check user status online/offline kamailio wait for reply from B2BUA on MESSAGE status if it return error 503 that mean user offline and m_store(0 should store message in db, but the issue that error 503 is in final stage and MESSAGE body is not available any more. The improvement will allow specify body $var in m_store() so new way will be m_store($tu, $avp(body)). Any help thank you. Slava.

2 1

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 14 Nov '16

14 Nov '16

Hello Daniel, I resolved all configuration issues and right now call goes both directions include rtp. Again huge thank you for help. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "volga629" <volga629(a)skillsearch.ca> Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 10 November, 2016 12:24:57 Subject: Re: [SR-Users] BYE dispatcher Hello, this logic is definitely wrong -- FreeSwitch can send also a request, it means that you send it back to it. Only the initial request of a dialog should be routed with rules like dispatcher/load balancer/least cost routing/... The requests within dialog should be routed based on loose routing. Of course, one can think of exceptions, but then you should be fully aware of what kind of routing you do. Cheers, Daniel On 10/11/16 16:25, Slava Bendersky wrote: Hello Daniel, My setup is proxy all requests to freeswitch via dispatcher. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "volga629" <volga629(a)skillsearch.ca> , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 10 November, 2016 04:56:53 Subject: Re: [SR-Users] BYE dispatcher Hello, as I said before, the registrations have little to do with calls in sip, unless there is gruu in use. Cheers, Daniel On 09/11/16 18:07, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld> ;tag=SXt3DQQ90a0Dj To: < sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com

2 1

Homer's kamailio config + enabling siptrace module
by Ján Füri 11 Nov '16

11 Nov '16

Hi guys, I'm using captagent on my all kamalios and freeswitches to capture and send sip to homer server. That works very well. But I would like to enable, on homer's kamailio, the module SipTrace (http://www.kamailio.org/docs/modules/4.4.x/modules/siptrace.html) and save captured data into mysql as well. I already set it up using this mysql structure : http://www.kamailio.org/docs/db-tables/kamailio-db-devel.html#idp1279328 Almost everything works except the column "DIRECTION". Each row is saved as "IN" direction. Do you guys think that can be solved ? If so, can you pleas direct me HOW ... Thx Jan F.

2 3

How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 11 Nov '16

11 Nov '16

Hello I have some UAC like Panasonic PBX, that send traffic to port 6060 that I am listening on, but that port isn't included to R-URI. I can see that only at tcpdump, or sip_trace from sip_capture module. Kamailio variables like $dp or $rp have default value of 5060. Thank You,

3 2

Execution Error while getting profile size with jsonrpc-s
by Aqs Younas 11 Nov '16

11 Nov '16

Greetings list, I am trying to get profile size with jsonrpc-s module. Below is jsonrpc-s configuration and a curl command to get the profile size. listen=tcp:0.0.0.0:5060 loadmodule "xhttp" loadmodule "jsonrpc-s" modparam("xhttp", "url_match", "^/rpc_path/") modparam("jsonrpc-s", "pretty_format", 1) event_route[xhttp:request] { if($hu =~ "^/rpc_path") jsonrpc_dispatch(); else xhttp_reply("404", "Not Found", "text/html", ""); } curl -v -H 'Content-Type: application/json' -H 'Call-ID: abc12' -d '{"jsonrpc": "2.0", "method": "dlg.profile_get_size","params":{"accountno": "1234567"}, "id": "0deadb33f"}' http://127.0.0.1:5060/rpc_path/ I am getting this in respose. * Hostname was NOT found in DNS cache * Trying 127.0.0.1... * Connected to 127.0.0.1 (127.0.0.1) port 5060 (#0) > POST /rpc_path/ HTTP/1.1 > User-Agent: curl/7.38.0 > Host: 127.0.0.1:5060 > Accept: */* > Content-Type: application/json > Call-ID: abc12 > Content-Length: 106 > * upload completely sent off: 106 out of 106 bytes *< HTTP/1.1 404 Profile not found: 1234567* < Sia: SIP/2.0/TCP 127.0.0.1:49374 < Call-ID: abc12 < Content-Type: application/json * Server kamailio (4.4.1 (x86_64/linux)) is not blacklisted < Server: kamailio (4.4.1 (x86_64/linux)) < Content-Length: 106 < { "jsonrpc": "2.0", "error": { "code": -32000, "message": "Execution Error" }, "id": "0deadb33f" * Connection #0 to host 127.0.0.1 left intact Whereas fifo command gives me this. root@debian:/usr/local/kamailio/sbin# ./kamctl fifo profile_get_size accountno 1234567 profile:: name=accountno value=1234567 count=2 Why jsonrpc-s is search for profile 1234567 whereas profile should be accountno. Anyhelp what i am doing wrong here is much appreciated. Best Regards.

2 4

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 11 Nov '16

11 Nov '16

Hello Daniel, I really ask for help, here are configuration file https://paste.fedoraproject.org/477652/88413891/ I spent quite a lot of time trying understand loose_route() /record_route() mix. I can get signalling working, call is not disconnects, but no RTP. Or I can get rtp and signalling BYE is not routed properly. My setup is just proxy all requests to freesiwtch boxes base on dispatcher selection where kamailio setup with 2 interfaces public and private. I really appreciate on you time and help. Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Cc: miconda(a)gmail.com Sent: Thursday, 10 November, 2016 23:54:40 Subject: Re: [SR-Users] BYE dispatcher Hello Daniel, What From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com Cc: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 10 November, 2016 11:25:19 Subject: Re: [SR-Users] BYE dispatcher Hello Daniel, My setup is proxy all requests to freeswitch via dispatcher. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "volga629" <volga629(a)skillsearch.ca>, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 10 November, 2016 04:56:53 Subject: Re: [SR-Users] BYE dispatcher Hello, as I said before, the registrations have little to do with calls in sip, unless there is gruu in use. Cheers, Daniel On 09/11/16 18:07, Slava Bendersky wrote: Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld> ;tag=SXt3DQQ90a0Dj To: < sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Kamailio 4.4 VoiceMail
by Gonzalo Gasca Meza 10 Nov '16

10 Nov '16

I have the following call flow: INVITE -> sip:gonzalo@sip.parzee.io -- TLS/TCP/UDP -> *KAMAILIO* - DB Lookup -> INVITE sip:gonzalo58@test.external.com;transport=tls (Phone1) If Phone1 is Busy or No answer, I want call to go to VM. Phone1, is not registered to Kamailio, nor I'm using Realtime Integration. This Phone1 is registered to an external PBX. Currently in sample configuration script, seems to be that value: $avp(oexten) is used to redirect to VM, but in my case this value is null. I didnt find any documentation for this. *Questions:* a) What is $avp(oexten) ? b) What is the best way to pass a Redirect number in SIP INVITE to VoiceMail system (Asterisk or Freeswitch) c) Is there a way to configure CFNA timer per alias/uri ? *Example*: route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE|SUBSCRIBE")) return; # check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail routing enabled but IP not defined\n"); return; } if(is_method("INVITE")) { xdbg("SIP Request: method [$rm] from [$fu] to [$tu]\n"); xlog("VoiceMail routing enabled $avp(oexten)\n"); if($avp(oexten)==$null) return; $ru = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); xlog("SCRIPT: VoiceMail to $tu\n"); } else { if($rU==$null) return; $ru = "sip:" + $rU + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); } route(RELAY); exit; #!endif return; }

2 2

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Daniel, My setup is proxy all requests to freeswitch via dispatcher. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "volga629" <volga629(a)skillsearch.ca>, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Thursday, 10 November, 2016 04:56:53 Subject: Re: [SR-Users] BYE dispatcher Hello, as I said before, the registrations have little to do with calls in sip, unless there is gruu in use. Cheers, Daniel On 09/11/16 18:07, Slava Bendersky wrote: Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld> ;tag=SXt3DQQ90a0Dj To: < sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com , "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: BQ_BEGIN Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com

2 1

How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 10 Nov '16

10 Nov '16

Hello Thank You. In using Kamailio only as: route { if (is_method("INVITE")) { sql_query("cb", "insert into invites (from_user,to_user,callid,src_ip,dst_ip,src_port,dst_port) values ('$fU','$rU','$ci','$si','$rd','$sp','$rp')"); } } I know that sip_capture(); In some way get real DST port even if there is no port in r-URI. Greetings

2 1

Change CLID
by Nelson Migliaro 10 Nov '16

10 Nov '16

Hello, I have several SIP vendors and I must present a different CLID for each of them. I am planning to use failure routers in Kamailio to route calls to a different vendor in case of failure. The problem I face is changing clid in each route. I am planning to use a mysql database to select the clid based on a prefix and sip vendor and in case of failure, select new clid base on new sip vendor. Is mysql the right way to do that, is there a module for this? In order to change the clid I am thinking of using: uac_replace_to(display,uri). Is this correct way to do that? And finally, in case of using uac_replace_to(display,uri), do I have to use uac_restore_from() to restone the correct from to avoid further issues? Thank you in advance Nelson.-

2 1

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Everyone, Here are full trace call. https://paste.fedoraproject.org/476607/14787290/ Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:17:34 Subject: Re: [SR-Users] BYE dispatcher Based on this out put Freeswitch send BYE to kamailio and Route present then kamailio forward BYE to client and no routes. Then client reply 481. Do I need add it ? Is this tag= problem ? 24 is freeswtich and 27 kamailio. IP (tos 0x0, ttl 64, id 56723, offset 0, flags [none], proto UDP (17), length 704) 10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 676 E.......@..B ... ....(....8.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.24:5160;rport;branch=z9hG4bKm80c0USSKv5Bp Route: <sip:10.18.130.27;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Max-Forwards: 70 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x10, ttl 64, id 36705, offset 0, flags [none], proto UDP (17), length 700) proxy_public_ip.llrp > client_public_ip.49383: [bad udp cksum 0x4d15 -> 0x34be!] UDP, length 672 E....a..@..d.E.\c.........M.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp Max-Forwards: 69 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_public_ip.49383 > proxy_public_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:07:11 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Based on this out put Freeswitch send BYE to kamailio and Route present then kamailio forward BYE to client and no routes. Then client reply 481. Do I need add it ? Is this tag= problem ? 24 is freeswtich and 27 kamailio. IP (tos 0x0, ttl 64, id 56723, offset 0, flags [none], proto UDP (17), length 704) 10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 676 E.......@..B ... ....(....8.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.24:5160;rport;branch=z9hG4bKm80c0USSKv5Bp Route: <sip:10.18.130.27;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Max-Forwards: 70 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x10, ttl 64, id 36705, offset 0, flags [none], proto UDP (17), length 700) proxy_public_ip.llrp > client_public_ip.49383: [bad udp cksum 0x4d15 -> 0x34be!] UDP, length 672 E....a..@..d.E.\c.........M.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp Max-Forwards: 69 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_public_ip.49383 > proxy_public_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:07:11 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

usrloc module expires column issues
by Grant Bagdasarian 10 Nov '16

10 Nov '16

Hello, When saving in the location table, the expires and last_modified columns contain the value 000-00-00 00:00:00 which causes issues when Kamailio is restarted and the preload modparam is set to load the location table from db in memory at startup. I've already upgraded Kamailio to 4.4.3, and used the dbtables script from version 4.4 to recreate the location table, but without success. Anyone else having this issue? Any resolution for it? Regards, Grant Bagdasarian CM

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Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Daniel, Here more clear trace https://www.dropbox.com/s/2z6ck71ulidqelh/kamailio-BYE-flow.png?dl=0 BYE send by kamailio to wrong freeswitch box. Slava From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 20:09:53 Subject: Re: [SR-Users] BYE dispatcher Hello Daniel, Please check this sip call flow picture https://www.dropbox.com/s/itdewdg3ph7xcyx/kamailio-fs-BYE-flow.gif?dl=0 Kamilio send BYE to incorrect freeswitch which already responded to BYE from leg one. Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 19:11:56 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, Here are full trace call. https://paste.fedoraproject.org/476607/14787290/ Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:17:34 Subject: Re: [SR-Users] BYE dispatcher Based on this out put Freeswitch send BYE to kamailio and Route present then kamailio forward BYE to client and no routes. Then client reply 481. Do I need add it ? Is this tag= problem ? 24 is freeswtich and 27 kamailio. IP (tos 0x0, ttl 64, id 56723, offset 0, flags [none], proto UDP (17), length 704) 10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 676 E.......@..B ... ....(....8.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.24:5160;rport;branch=z9hG4bKm80c0USSKv5Bp Route: <sip:10.18.130.27;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Max-Forwards: 70 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x10, ttl 64, id 36705, offset 0, flags [none], proto UDP (17), length 700) proxy_public_ip.llrp > client_public_ip.49383: [bad udp cksum 0x4d15 -> 0x34be!] UDP, length 672 E....a..@..d.E.\c.........M.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp Max-Forwards: 69 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_public_ip.49383 > proxy_public_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:07:11 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Re: [SR-Users] BYE dispatcher
by Slava Bendersky 10 Nov '16

10 Nov '16

Hello Daniel, Please check this sip call flow picture https://www.dropbox.com/s/itdewdg3ph7xcyx/kamailio-fs-BYE-flow.gif?dl=0 Kamilio send BYE to incorrect freeswitch which already responded to BYE from leg one. Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 19:11:56 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, Here are full trace call. https://paste.fedoraproject.org/476607/14787290/ Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:17:34 Subject: Re: [SR-Users] BYE dispatcher Based on this out put Freeswitch send BYE to kamailio and Route present then kamailio forward BYE to client and no routes. Then client reply 481. Do I need add it ? Is this tag= problem ? 24 is freeswtich and 27 kamailio. IP (tos 0x0, ttl 64, id 56723, offset 0, flags [none], proto UDP (17), length 704) 10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 676 E.......@..B ... ....(....8.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.24:5160;rport;branch=z9hG4bKm80c0USSKv5Bp Route: <sip:10.18.130.27;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=SXt3DQQ90a0Dj> Max-Forwards: 70 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x10, ttl 64, id 36705, offset 0, flags [none], proto UDP (17), length 700) proxy_public_ip.llrp > client_public_ip.49383: [bad udp cksum 0x4d15 -> 0x34be!] UDP, length 672 E....a..@..d.E.\c.........M.BYE sip:4300@client_public_ip:49383 SIP/2.0 Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp Max-Forwards: 69 From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_public_ip.49383 > proxy_public_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@client_public_ip:49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 13:07:11 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I cleared registrations and tried again and issue still present. Client reply with 481. IP (tos 0x0, ttl 52, id 7731, offset 0, flags [none], proto UDP (17), length 638) client_pub_ip.49383 > proxy_pub_ip.llrp: [udp sum ok] UDP, length 610 E..~.3..4...c....E.\.....j..SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_pub_ip :5084;branch=z9hG4bK3ea6.0c594485bff5b216f30af0f6172cb2b9.0 Via: SIP/2.0/UDP 10.18.130.24:5160;received=10.18.130.24;rport=5160;branch=z9hG4bKm80c0USSKv5Bp From: "Test Extension" <sip:4300@sip.company.tld>;tag=SXt3DQQ90a0Dj To: <sip:4300@ client_pub_ip :49383>;tag=719973534 Call-ID: 1abc150b-2141-1235-b5ad-5254003e39bb CSeq: 99019404 BYE Supported: replaces, path, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 Slava. From: "volga629" <volga629(a)skillsearch.ca> To: miconda(a)gmail.com, "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 12:28:32 Subject: Re: [SR-Users] BYE dispatcher Hello Everyone, I changed dispatcher algorithm from 0 to 1 and start working as expected. Yes group 0 is accepted. route[DISPATCHER] { if(!ds_select_dst("0", "1")) { xlog("L_ERROR","ERROR: Proxy Mapping - Desitnation for $fd not found...request dropped \n"); sl_send_reply("404","Desitination Not Found \n"); drop(); } else { $var(did) = 1; } if($var(did)) { if (!t_relay()) { sl_reply_error(); } #forward(); } t_on_failure("DISPATCHER_FAIL_ROUTE"); exit; } Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 9 November, 2016 04:33:33 Subject: Re: [SR-Users] BYE dispatcher Hello, On 08/11/16 20:42, Slava Bendersky wrote: Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). is group value 0 accepted? I think this may create problems if a function returns the group in the config as return code -- iirc, this was changed maybe for lcr or permissions. On the other hand, the registrations are quite independent in SIP in relation with calls. The BYE should be routed based on record-routing to the freeswitch that was involved in routing initial INVITE, with no relation to new registrations from end devices. Is the BYE sent to the freeswitch that got the initial BYE. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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Kamailio v4.4.4 Released
by Daniel-Constantin Mierla 09 Nov '16

09 Nov '16

Hello, Kamailio SIP Server v4.4.4 stable release is out. This is a maintenance release of the latest stable branch, 4.4, that includes fixes since the release of v4.4.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v4.4.x. Deployments running previous v4.x.x versions are strongly recommended to be upgraded to v4.4.4. For more details about version 4.4.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2016/11/kamailio-v4-4-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

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Count number of early(Ringing) dialogs on gateway
by Aqs Younas 09 Nov '16

09 Nov '16

Greetings list, I am using dialog aur dispatcher modules for calls distribution across multiple gateways. I am able to get number of active dialogs against a gateway but could not get early dialogs going on particular gateway. Dialog information is inserted in database only for state 4. Also, I tried to use dlg_list command to get early dialogs. I see call statistics but "callee_contact::" is empty untill call is answered. for ringing I am getting this. root@debian:/usr/local/kamailio/sbin# ./kamctl fifo dlg_list dialog:: hash=3443:9422 state:: 2 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: GGjl58SQ32LXQB1dV2-CBA.. from_uri:: sip:1015@192.168.10.39:5060;transport=UDP from_tag:: 652ee907 caller_contact:: sip:1015@192.168.10.41:55842;transport=UDP caller_cseq:: 1 caller_route_set:: caller_bind_addr:: udp:192.168.10.39:5060 callee_bind_addr:: to_uri:: sip:1010@192.168.10.39:5060;transport=UDP to_tag:: callee_contact:: callee_cseq:: callee_route_set:: callee_contact:: is empty. Is there any way to get early dialog info againts a gateway.? Or any way to insert early dialog into database with callee_contact not empty? Best Regards, Aqs Younas

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Planning the release of Kamailio v4.4.4
by Daniel-Constantin Mierla 09 Nov '16

09 Nov '16

Hello, I am considering to package a new release out of latest stable branch 4.4, respectively v4.4.4. The target date is one of the days between Nov 8-10, a matter of work load that needs to be done for getting it out. As usual, if you are aware of any issue not yet reported on github bug tracker, do it as soon as possible to give it a chance to be fixed: - https://github.com/kamailio/kamailio/issues Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com

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Re: [SR-Users] fnmatch
by Slava Bendersky 09 Nov '16

09 Nov '16

Hello Daniel, Yes, reply was 202 not 200. Right now it work as expected. Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Monday, 7 November, 2016 04:51:11 Subject: Re: [SR-Users] fnmatch Hello, for clarification -- what's the actual problem here? fnmatch("$rs", "200") is matching the 202? Cheers, Daniel On 07/11/16 08:52, Serge S. Yuriev wrote: Hello, In debug you can see 202 Accepted but match configured on 200. Not matched -- Wbr, Serge via mobile 07.11.2016, 01:08, "Slava Bendersky" <volga629(a)skillsearch.ca> : BQ_BEGIN Hello Everyone, I trying fnmatch to find which sip status is returned by B2BUA. But some reason it doesn't work. Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=456 a=26 n=xlog Nov 6 17:02:15 cavprx00 kamailio: 2(3747) INFO: <script>: Message 1 Accepted by B2BUA --> 202 with Accepted Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=469 a=16 n=if Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=457 a=26 n=fnmatch Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio-asterisk.cfg] l=370 a=2 n=return if(fnmatch("$rs", "200") && (fnmatch("$rr", "Accepted"))) { xlog("L_INFO", "Message Accepted by B2BUA --> $rs with $rr \n"); $var(tu) = $tu; xlog("L_INFO", "New To --> $var(tu)\n"); } Is this fnmatch correct for this operation ? Slava. _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users BQ_END -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

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BYE dispatcher
by Slava Bendersky 09 Nov '16

09 Nov '16

Hello Everyone, My setup is kamailio as proxy with few boxes of freeswitch in the LAN. Having issue with BYE when extensions register on different freeswitch boxes. Here are some trace of the call. Not sure if this tag= miss match or routing. Dispatcher use group 0 with option 4 (round robin). client_public_ip.28972 > proxy_public_ip.llrp: [udp sum ok] UDP, length 660 E....2..5...c....E.\q,.....CBYE sip:mod_sofia@10.18.130.24:5160 SIP/2.0 Via: SIP/2.0/UDP client_public_ip:28972;branch=z9hG4bK1248337638;rport Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=01K2Q1BFa9pKH> Route: <sip:10.18.130.27;r2=on;lr=on;ftag=01K2Q1BFa9pKH> From: <sip:4300@client_public_ip:28972>;tag=487304085 To: <sip:4300@sip.company.tld>;tag=01K2Q1BFa9pKH Call-ID: 1443c4ad-2085-1235-b5ad-5254003e39bb CSeq: 98979026 BYE Contact: <sip:4300@client_public_ip:28972> Max-Forwards: 70 Supported: replaces, path, timer, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 IP (tos 0x10, ttl 64, id 10813, offset 0, flags [none], proto UDP (17), length 930) 10.18.130.27.sip > 10.18.130.24.5160: [bad udp cksum 0x1bf7 -> 0xd85c!] UDP, length 902 E...*=..@.4. ... ......(....BYE sip:mod_sofia@10.18.130.24:5160 SIP/2.0 Record-Route: <sip:10.18.130.27;r2=on;lr=on;ftag=487304085> Record-Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=487304085> Via: SIP/2.0/UDP 10.18.130.27;branch=z9hG4bKa61.22917bc5fdd65f16e39bf09dcbc7316b.0 Via: SIP/2.0/UDP client_public_ip:28972;received=client_public_ip;branch=z9hG4bK1248337638;rport=28972 Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=01K2Q1BFa9pKH> Route: <sip:10.18.130.27;r2=on;lr=on;ftag=01K2Q1BFa9pKH> From: <sip:4300@client_public_ip:28972>;tag=487304085 To: <sip:4300@sip.company.tld>;tag=01K2Q1BFa9pKH Call-ID: 1443c4ad-2085-1235-b5ad-5254003e39bb CSeq: 98979026 BYE Contact: <sip:4300@client_public_ip:28972> Max-Forwards: 69 Supported: replaces, path, timer, eventlist User-Agent: Grandstream Wave 1.2.2 Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Length: 0 IP (tos 0x0, ttl 64, id 38202, offset 0, flags [none], proto UDP (17), length 587) 10.18.130.24.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 559 E..K.:..@... ... ....(...7i.SIP/2.0 200 OK Via: SIP/2.0/UDP 10.18.130.27;branch=z9hG4bKa61.22917bc5fdd65f16e39bf09dcbc7316b.0 Via: SIP/2.0/UDP client_public_ip:28972;received=client_public_ip;branch=z9hG4bK1248337638;rport=28972 From: <sip:4300@client_public_ip:28972>;tag=487304085 To: <sip:4300@sip.company.tld>;tag=01K2Q1BFa9pKH Call-ID: 1443c4ad-2085-1235-b5ad-5254003e39bb CSeq: 98979026 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 IP (tos 0x10, ttl 64, id 782, offset 0, flags [none], proto UDP (17), length 503) proxy_public_ip.llrp > client_public_ip.28972: [bad udp cksum 0x4c50 -> 0xd018!] UDP, length 475 E.......@.+}.E.\c.....q,..LPSIP/2.0 200 OK Via: SIP/2.0/UDP client_public_ip:28972;received=client_public_ip;branch=z9hG4bK1248337638;rport=28972 From: <sip:4300@client_public_ip:28972>;tag=487304085 To: <sip:4300@sip.company.tld>;tag=01K2Q1BFa9pKH Call-ID: 1443c4ad-2085-1235-b5ad-5254003e39bb CSeq: 98979026 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Content-Length: 0 IP (tos 0x0, ttl 64, id 10584, offset 0, flags [none], proto UDP (17), length 764) 10.18.130.26.5160 > 10.18.130.27.sip: [udp sum ok] UDP, length 736 E...)X..@.6@ ... ....(....Y^BYE sip:4200@client_public_ip:48147;rinstance=c4084e18f3f39974;alias=client_public_ip~48147~1 SIP/2.0 Via: SIP/2.0/UDP 10.18.130.26:5160;rport;branch=z9hG4bK33785247SrX3H Route: <sip:10.18.130.27;r2=on;lr=on;ftag=d2e84f4a> Route: <sip:proxy_public_ip:5084;r2=on;lr=on;ftag=d2e84f4a> Max-Forwards: 70 From: <sip:4300@sip.company.tld>;tag=5K0FX0ryeBBeH To: "volga629" <sip:4200@sip.company.tld>;tag=d2e84f4a Call-ID: 137861_rel44MTA1ZGMwNzViODg5NDMwYzQ1NDQzYjlmMGExMzlmMmI CSeq: 98979038 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x10, ttl 64, id 802, offset 0, flags [none], proto UDP (17), length 741) proxy_public_ip.llrp > client_public_ip.48147: [bad udp cksum 0x4d3e -> 0x4f2d!] UDP, length 713 E...."..@.*{.E.\c.........M>BYE sip:4200@client_public_ip:48147;rinstance=c4084e18f3f39974 SIP/2.0 Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK8395.d46858df1d9e02b910446fca09f1f864.0 Via: SIP/2.0/UDP 10.18.130.26:5160;received=10.18.130.26;rport=5160;branch=z9hG4bK33785247SrX3H Max-Forwards: 69 From: <sip:4300@sip.company.tld>;tag=5K0FX0ryeBBeH To: "volga629" <sip:4200@sip.company.tld>;tag=d2e84f4a Call-ID: 137861_rel44MTA1ZGMwNzViODg5NDMwYzQ1NDQzYjlmMGExMzlmMmI CSeq: 98979038 BYE User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Reason: Q.850;cause=16;text="NORMAL_CLEARING" Content-Length: 0 IP (tos 0x0, ttl 52, id 810, offset 0, flags [DF], proto UDP (17), length 504) client_public_ip.48147 > proxy_public_ip.llrp: [udp sum ok] UDP, length 476 E....*@.4..oc....E.\........SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP proxy_public_ip:5084;branch=z9hG4bK8395.d46858df1d9e02b910446fca09f1f864.0 Via: SIP/2.0/UDP 10.18.130.26:5160;received=10.18.130.26;rport=5160;branch=z9hG4bK33785247SrX3H To: "volga629" <sip:4200@sip.company.tld>;tag=d2e84f4a From: <sip:4300@sip.company.tld>;tag=5K0FX0ryeBBeH Call-ID: 137861_rel44MTA1ZGMwNzViODg5NDMwYzQ1NDQzYjlmMGExMzlmMmI CSeq: 98979038 BYE Accept-Language: en Content-Length: 0 IP (tos 0x10, ttl 64, id 63040, offset 0, flags [none], proto UDP (17), length 413) 10.18.130.27.sip > 10.18.130.26.5160: [bad udp cksum 0x19f4 -> 0x6386!] UDP, length 385 E....@..@.j. ... ......(....SIP/2.0 481 Call/Transaction Does Not Exist Via: SIP/2.0/UDP 10.18.130.26:5160;received=10.18.130.26;rport=5160;branch=z9hG4bK33785247SrX3H To: "volga629" <sip:4200@sip.company.tld>;tag=d2e84f4a From: <sip:4300@sip.company.tld>;tag=5K0FX0ryeBBeH Call-ID: 137861_rel44MTA1ZGMwNzViODg5NDMwYzQ1NDQzYjlmMGExMzlmMmI CSeq: 98979038 BYE Accept-Language: en Content-Length: 0 Slava.

2 1

Count number of early(Ringing) dialogs on gateway
by Aqs Younas 09 Nov '16

09 Nov '16

Greetings list, I am using dialog aur dispatcher modules for calls distribution across multiple gateways. I am able to get number of active dialogs against a gateway but could not get early dialogs going on particular gateway. Dialog information is inserted in database only for state 4. Also, I tried to use dlg_list command to get early dialogs. I see call statistics but "callee_contact::" is empty untill call is answered. For ringing I am getting this. root@debian:/usr/local/kamailio/sbin# ./kamctl fifo dlg_list dialog:: hash=3443:9422 state:: 2 ref_count:: 1 timestart:: 0 timeout:: 0 callid:: GGjl58SQ32LXQB1dV2-CBA.. from_uri:: sip:1015@192.168.10.39:5060;transport=UDP from_tag:: 652ee907 caller_contact:: sip:1015@192.168.10.41:55842;transport=UDP caller_cseq:: 1 caller_route_set:: caller_bind_addr:: udp:192.168.10.39:5060 callee_bind_addr:: to_uri:: sip:1010@192.168.10.39:5060;transport=UDP to_tag:: callee_contact:: callee_cseq:: callee_route_set:: callee_contact:: is empty. Is there any way to get early dialog info against each gateway or every dialog stat insertion in database dialog table. Best Regards.

1 0

using set_body_multipart and append_body_part on on_reply routes
by david 08 Nov '16

08 Nov '16

hello all i'm using kam version 4.4.1 and i would like to generate an ISUP part on the 1xx,2xx responses to a remote carrier since the fsw i send the calls to cannot handle ISUP SDP. despite seeing in the doc that those functions cannot be used there and only in REQUEST_ROUTE, FAILURE_ROUTE, BRANCH_ROUTE, i tried and it works.... partially, because it seems it doesnt create the last boundary and it creates a next boundary with an empty "Encapsulated multipart part: " this the the route i call from onreply_route route[SIPISUP] { set_body_multipart(); msg_apply_changes(); $var(isup) = "\x01\x12\x49\x01\x0a\x03\x02\x0a\x08\x84\x90\x33\x41 \x72\x17\x01\x06\x0a\x08\x04\x13\x93\x70\x21\x73\x23\x10\x08\x01\x01 \x00"; append_body_part("$var(isup)","application/isup;version=itu-t92 +","signal;handling=optional"); msg_apply_changes(); sipt_destination($rU, 31, 4); sipt_set_calling($fU, 4, 0, 3); msg_apply_changes(); } am i doing something wrong? how can i set this to end the SDP with last boundary? thanks a lot and regards david

2 3

Re: [SR-Users] kamailio + farm of Elastix + rtp direct
by Kato 08 Nov '16

08 Nov '16

Any comments on this please? Thanks! De: Kato [mailto:elkato@gmail.com] Enviado el: domingo, 6 de noviembre de 2016 11:29 a. m. Para: 'sr-users(a)lists.sip-router.org' <sr-users(a)lists.sip-router.org> Asunto: kamailio + farm of Elastix + rtp direct Hi, I'm just starting with Kamailio, and trying to validate a possible solution design. I started offering hosted PBXs (FreePBX or Elastix as customer wish), each one with a public IP, connected with SIP trunks from Net2Phone provider. RTP from/to customers calls is going in and out through PBX, and not directly customer <----> sip trunk provider. Billing is provided by SIP trunk provider, so with just CDR from freepbx/Elastix is enough. But, I would like to introduce 2 main changes. First a sip router/proxy in front of PBXs like kamailio, and doing just a basic domain based routing to each PBX. Kamailio with 2 interfaces (1 public, 1 internal) and change PBXs to internal IPs. Second, to know if would be possible to redirect RTP from/to customer-net2phone provider directly, not going through PBXs. Net2phone has public IPs for RTP, customer normally in their homes behind NAT with a SOHO router. Elastix 2.5 using Asterisk 11, and FreePBX using Asterisk 12 (could be 11 or 13 too tbe o reach this). I would like to kamilio be as simple and static as possible, not handling REGISTER or LOCATION. Just to route all traffic based on each domain to each corresponding PBX. My idea will be to pre-provide all subdomains to each internal IP using subdomains like "customer0001.domain.com, customer0002.domain.com." and customers would login as 100(a)customer0001.domain.com <mailto:100@customer0001.domain.com> , 101(a)customer0001.domain.com <mailto:101@customer0001.domain.com> to PBX0001 and 100(a)customer0002.domain.com <mailto:100@customer0002.domain.com> , 101(a)customer0002.domain.com <mailto:101@customer0002.domain.com> to PBX0002 for example. Just using as it was a firewall doing NAT and Asterisk (freepbx/elastix) behinds handles everything as they are doing already. Maybe kamailio could check/filter some malicious packets like I'm doing now with fail2ban on PBXs. Can be this possible? In that case where do I need to setup SIP trunk? On kamailio or on internal PBXs as I have it now? For second part, just to know and understand if would be possible customer "talks" RTP directly to Net2phone SIP trunk provider on doing or receiving external calls. Not going to my DC to save BW. How I could do that? Thanks a lot in advance! Regards, elkato

1 0

Next Kamailio IRC Devel Meeting, Nov 09, 2016
by Daniel-Constantin Mierla 08 Nov '16

08 Nov '16

Hello, I am proposing a new IRC meeting to discuss the current major issues and the plans for next Kamailio release, on Wednesday, Nov 09, 2016, with alternatives for previous or next day. If many developers are not available, we can postpone it to another date in the near future (make proposals if that is the case for you). I created a wiki page for it: * https://www.kamailio.org/wiki/devel/irc-meetings/2016b Add there the topics that you want to be discussed. Cheers, Daniel -- Daniel-Constantin Mierla http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda

1 4

Contacts replacing each other when registering
by Sebastian Damm 08 Nov '16

08 Nov '16

Hi, we are running a Registrar server with 4.4.3, the registrar module is configured with the following parameters: modparam("registrar", "max_contacts", 50) modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 0) modparam("registrar", "path_use_received", 1) modparam("registrar", "received_avp", "$(avp(i:801))") modparam("registrar", "default_expires", 600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 600) modparam("registrar", "gruu_enabled", 0) A customer has configured 4 phones with the same username. Normally, all 4 phones would be visible in the location database, and incoming calls would fork out to all 4 phones. However, only one of the devices is always online, the others always replace each other on each registration. Client details: * All phones are snom 715 devices with the same firmware version. * All phones sit behind the same router. Registrations: client 1: * Via: SIP/2.0/UDP 192.168.208.91:35812;branch=z9hG4bK-4q0mmxgu6gm7;rport * Contact: <sip:1794388e3@192.168.208.91:35812;line=gicode6v>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>" client 2: * Via: SIP/2.0/UDP 192.168.208.90:41331;branch=z9hG4bK-3cbxdvbmk0di;rport * Contact: <sip:1794388e3@192.168.208.90:41331;line=0j6veso8>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>" client 3: * Via: SIP/2.0/UDP 192.168.208.92:5360;branch=z9hG4bK-4drsogakez94;rport * Contact: <sip:1794388e3@192.168.208.92:5360;line=0n7jfzih>;reg-id=1;q=1.0;+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>" client 4: * Via: SIP/2.0/UDP 192.168.208.93:5260;branch=z9hG4bK-1ci0qia3yxyl;rport * Contact: <sip:1794388e3@192.168.208.93:5260;line=mh9sgxa1>;reg-id=1;q=1.0 Only client 4 is reachable all the time. The other 3 clients replace other registrations. So after client 1 registers, the contacts returned in the 200 OK looks like this: Contact: <sip:1794388e3@192.168.208.91:35812;line=gicode6v>;q=1;expires=600;received="sip:1.2.3.4:6411";+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>";reg-id=1, <sip:1794388e3@192.168.208.93:5260;line=mh9sgxa1>;q=1;expires=303;received="sip:1.2.3.4:8923" And after client 2 registers, the contacts returned in the 200 OK looks like this: Contact: <sip:1794388e3@192.168.208.90:41331;line=0j6veso8>;q=1;expires=600;received="sip:1.2.3.4:49061";+sip.instance="<urn:uuid:0a14174a-77be-424c-864e-0004137F1C9D>";reg-id=1, <sip:1794388e3@192.168.208.93:5260;line=mh9sgxa1>;q=1;expires=301;received="sip:1.2.3.4:8923" So you see, one contact stays the same, the other one gets replaced. Does anybody know what could cause this kind of behavior? I think, it could be GRUU. I see that those three clients all send the same GRUU parameter. From my understanding, this UUID should be different from client to client. I don't know whether this is a bug in the snom firmware always sending the same GRUU UUID or whether som "intelligent" SIP aware router is putting this in the Contact header. However, since we have disabled gruu for the registrar module (see above), I think, Kamailio should ignore it. Ideas? Thanks in advance. Best Regards, Sebastian

2 4

TCP Keepalive for NAT'd Clients
by Colin Morelli 08 Nov '16

08 Nov '16

Hey all, Looking to figure out the best way to allow TCP connections to stay alive for NAT'd clients, however, to protect against people just opening TCP connections to the server, I'm hoping to only keepalive TCP connections for connections that have sent an INVITE and received a 200 OK. Does this type of per-socket connection lifetime setting exist? For a bit more background: this is specifically for clients that are using the TCP connection for an INVITE. They don't register with the server, and as a result there's no active flow timer for their connection. Is there a way to enable this type of behavior without registering? Best, Colin

2 2

How to get Destination port UDP
by andrzej.ciupek-asterisk.edu.pl 08 Nov '16

08 Nov '16

Hello I am using Kamailio to collect data about active calls. I write INVITEs to database, and delete it on BYE or any REL. I would like to write source and destination port too, but don't know how to get destination port. I have tried variables: $dp, $rp, $Rp, $op But when I don't have port in URI like INVITE IP:PORT, $rp have default value of 5060. I would like to get real UDP port, that INVITE does to. Is it possible using PSEUDO VARIABLES? Greetings Andrzej

2 1

Out of memory in UB 210: OOM killed process 12261 (kamailio) score 0 vm:1614768kB, rss:280200kB, swap:131408kB
by Jurijs Ivolga 08 Nov '16

08 Nov '16

Hi, Our Kamailio server is crashing once per week, with following error: Oct 1 06:25:06 kamailio kernel: [26982632.803789] Out of memory in UB 210: OOM killed process 12261 (kamailio) score 0 vm:1614768kB, rss:280200kB, swap:131408kB Core dump was never created, probably it is because of my environment, but I will try to get it. Server constantly eats memory, maybe some kind of memory leak? Any help is highly appreciated! Jurijs

3 25

NER and ASR
by Gholamreza Sabery 07 Nov '16

07 Nov '16

Are there any real data about number of concurrent calls, number of on-line users, NER (Network Effectiveness Ratio) and ASR (Answer Seizure Ratio) of a real VoIP system? I want this data for making comparisons. Can anyone provide a reference, article, etc..

3 4

cdr on remote POSTGRES server
by beer Ll 07 Nov '16

07 Nov '16

Hi all I have a kamailio server configured with accounting on a separate Postgres SQL server by an internal network with acc module and db_postgres module Everything works good but what can I do if I have problem on the internal network or on the postgres server ? Is it possible cache or save the data if the insert data is failed ?

2 1

fnmatch
by Slava Bendersky 07 Nov '16

07 Nov '16

Hello Everyone, I trying fnmatch to find which sip status is returned by B2BUA. But some reason it doesn't work. Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=456 a=26 n=xlog Nov 6 17:02:15 cavprx00 kamailio: 2(3747) INFO: <script>: Message 1 Accepted by B2BUA --> 202 with Accepted Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=469 a=16 n=if Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[MESSAGE_FORWARD] c=[/etc/kamailio/kamailio-asterisk.cfg] l=457 a=26 n=fnmatch Nov 6 17:02:15 cavprx00 kamailio: 2(3747) ERROR: *** cfgtrace:onreply_route=[DEFAULT_ROUTE] c=[/etc/kamailio/kamailio-asterisk.cfg] l=370 a=2 n=return if(fnmatch("$rs", "200") && (fnmatch("$rr", "Accepted"))) { xlog("L_INFO", "Message Accepted by B2BUA --> $rs with $rr \n"); $var(tu) = $tu; xlog("L_INFO", "New To --> $var(tu)\n"); } Is this fnmatch correct for this operation ? Slava.

3 2

kamailio + farm of Elastix + rtp direct
by Kato 06 Nov '16

06 Nov '16

Hi, I'm just starting with Kamailio, and trying to validate a possible solution design. I started offering hosted PBXs (FreePBX or Elastix as customer wish), each one with a public IP, connected with SIP trunks from Net2Phone provider. RTP from/to customers calls is going in and out through PBX, and not directly customer <----> sip trunk provider. Billing is provided by SIP trunk provider, so with just CDR from freepbx/Elastix is enough. But, I would like to introduce 2 main changes. First a sip router/proxy in front of PBXs like kamailio, and doing just a basic domain based routing to each PBX. Kamailio with 2 interfaces (1 public, 1 internal) and change PBXs to internal IPs. Second, to know if would be possible to redirect RTP from/to customer-net2phone provider directly, not going through PBXs. Net2phone has public IPs for RTP, customer normally in their homes behind NAT with a SOHO router. Elastix 2.5 using Asterisk 11, and FreePBX using Asterisk 12 (could be 11 or 13 too tbe o reach this). I would like to kamilio be as simple and static as possible, not handling REGISTER or LOCATION. Just to route all traffic based on each domain to each corresponding PBX. My idea will be to pre-provide all subdomains to each internal IP using subdomains like "customer0001.domain.com, customer0002.domain.com." and customers would login as 100(a)customer0001.domain.com <mailto:100@customer0001.domain.com> , 101(a)customer0001.domain.com <mailto:101@customer0001.domain.com> to PBX0001 and 100(a)customer0002.domain.com <mailto:100@customer0002.domain.com> , 101(a)customer0002.domain.com <mailto:101@customer0002.domain.com> to PBX0002 for example. Just using as it was a firewall doing NAT and Asterisk (freepbx/elastix) behinds handles everything as they are doing already. Maybe kamailio could check/filter some malicious packets like I'm doing now with fail2ban on PBXs. Can be this possible? In that case where do I need to setup SIP trunk? On kamailio or on internal PBXs as I have it now? For second part, just to know and understand if would be possible customer "talks" RTP directly to Net2phone SIP trunk provider on doing or receiving external calls. Not going to my DC to save BW. How I could do that? Thanks a lot in advance! Regards, elkato

1 0

Kamailio DB lookup
by Gonzalo Gasca Meza 06 Nov '16

06 Nov '16

Hi Kamailions, Kamailio 4.3 will receive calls (SIP TLS) for users in this domain: *sip.parzee.io <http://sip.parzee.io>. *Example sip:gonzalo@sip.parzee.io When call is received by Kamailio I need to overwrite R-URI and transport. I should keep a mapping per user. ( gonzalo -> gonzalo58) Example. sip:gonzalo58@test.external.com;transport=tls I started looking into *dbaliases* table in LOCATION. #!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif In dbaliases table I see 4 columns. Is the following configuration correct? *alias_username*: gonzalo58 *alias_domain*: test.external.com;transport=tls *username*: gonzalo *domain*: sip.parzee.io Thanks -Gonzalo

4 5

Malformed To header, or?
by Tobias 04 Nov '16

04 Nov '16

Hi! Having some issues with Kamailio 4.3 reporting a malformed header, but I cannot seem to figure out what's wrong. The problem seen is: Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [parser/parse_addr_spec.c:678]: parse_addr_spec(): ERROR: parse_to : unexpected char [#015] in status 6: <<+123456789 <sip:123456789@127.0.0.1;tag=as4aa27bd0>> . Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [parser/msg_parser.c:165]: get_hdr_field(): ERROR: get_hdr_field: bad to header Oct 27 13:48:56 /usr/sbin/kamailio[13765]: INFO: <core> [parser/msg_parser.c:338]: parse_headers(): ERROR: bad header field [To: +123456789 <si] Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: tm [t_lookup.c:1050]: t_check_msg(): ERROR: reply cannot be parsed Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [parser/parse_addr_spec.c:678]: parse_addr_spec(): ERROR: parse_to : unexpected char [#015] in status 6: <<+123456789 <sip:123456789@127.0.0.1;tag=as4aa27bd0>> . Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [parser/msg_parser.c:165]: get_hdr_field(): ERROR: get_hdr_field: bad to header Oct 27 13:48:56 /usr/sbin/kamailio[13765]: INFO: <core> [parser/msg_parser.c:338]: parse_headers(): ERROR: bad header field [To: +123456789 <si] Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [msg_translator.c:1457]: adjust_clen(): error parsing content-length Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [msg_translator.c:2214]: generate_res_buf_from_sip_res(): error while adjusting Content-Length Oct 27 13:48:56 /usr/sbin/kamailio[13765]: ERROR: <core> [forward.c:760]: do_forward_reply(): building failed And the To header seen at this point is: To: +123456789 <sip:123456789@127.0.0.1>;tag=as24ed5606. version: kamailio 4.3.4 (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, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: unknown compiled with gcc 4.7.2 I obviously understand Kamailio views this To header as malformed, but what I cannot understand is why? Is it as simple as the number +123456789 is not enclosed with ", i.e. it should be sent as "+123456789"? A similar issue seen, which for some reason doesn't create an ERROR, is when Kamailio receives a reINVITE looking like this which it's supposed to proxy: To: +123456789 <sip:123456789@127.0.0.1>;tag=SD1sf9001--45026-530b969-77c7543f-530b969. which is rewritten into: To: +123456789 <sip:123456789@127*0-)'.:/)&7'...0;user=phone>;tag=SD1sf9001--45026-530b969-77c7543f-530b969. and sent out.. Thanks for your help, /Tobias

3 13

Configuring/Integrating Kamailio Open IMS with FHoSS
by Olvera, Victor 04 Nov '16

04 Nov '16

How does one configure Kamailio Open IMS and FHoSS? In other words, how does one configure IMS and FHoSS so that when soft calls are made they must be processed by the FHoSS before they go through the Open IMS server? Great regards, Victor Olvera

8 13

nathelper not changes IP
by Yuriy Gorlichenko 03 Nov '16

03 Nov '16

I trying to use fix_nated_sdp with 0x02 and 0x08 flags for changing ip addres and the SDP body part like fix_nated_sdp(10,"1.2.3.4") But for now it beaks SDP At the output i see next o=- 7300689428214760503 2 IN IP4 1.1.1.1 c=IN IP4 1.1.1.1 1.2.3.41.2.3.4 <- is just a line that added after fix_nated_sdp atthe end of SDP body kamailio -v version: kamailio 4.4.3 (x86_64/linux) e91aec 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: e91aec compiled on 07:26:14 Oct 18 2016 with gcc 4.9.2

3 4

Cannot send request to OpenSIPS
by Laura 02 Nov '16

02 Nov '16

Dear support, on my Kamailio server (version: kamailio 4.3.6 (x86_64/linux) ) i see lot of these messages.. Nov 2 11:24:42 vm-test media-dispatcher[25540]: error: cannot write request to `/var/run/opensips/socket': No such file or directory Nov 2 11:24:42 vm-test media-dispatcher[25540]: error: failed to end dialog: Cannot send request to OpenSIPS I have only configured Disptachet (mediaproxy) for write on db.. Any idea ? Thanks Laura

2 4

diff between location and kamctl ul show
by Никитенко Виталий 02 Nov '16

02 Nov '16

Hi all! I have kamailio 4.2.3 with options modparam ( "usrloc", "db_url", DBURL) modparam ( "usrloc", "db_mode", 2) But not all registred users displayed in table "location". When viewed "kamctl ul show" displays Domain :: location table = 1024 records = 1172 max_slot = 5 But "location" table has 651 rows. Why not all users saved to mysql table? Thank!

3 2

Kamailio TLS client config selection
by Sergey Basov 02 Nov '16

02 Nov '16

Hi Daniel. After some try to configure kamailio 4.4.3 to act as SPI TLS client for Cisco SIP TLS gateways I have found one issue. If I do client configuration for tls [client:10.1.23.19:5061] verify_certificate = yes ca_list = /etc/kamailio/CAs/ca1.pem [client:10.1.23.29:5061] verify_certificate = yes ca_list = /etc/kamailio/CAs/ca2.pem [client:default] verify_certificate = no require_certificate = no Kamailo always do default profile selection (I do configuration without server_name or server_id, with it kamailio works fine but there are some troubles to make selection of this parameters from config script, I need additional checks and queries) after some research in tls module source code I have added some debug information in file tls_server.c: if (c->flags & F_CONN_PASSIVE) { state=S_TLS_ACCEPTING; dom = tls_lookup_cfg(cfg, TLS_DOMAIN_SRV, &c->rcv.dst_ip, c->rcv.dst_port, 0, 0); } else { state=S_TLS_CONNECTING; sname = tls_get_connect_server_name(); srvid = tls_get_connect_server_id(); // ------------------------------------------------------------- DBG("Entered client config loockup (c->rcv.dst_port %d)\n", c->rcv.dst_port); DBG("Entered client config loockup (&c->rcv.dst_ip %s)\n", ip_addr2a(&c->rcv.dst_ip)); DBG("Entered client config loockup (c->rcv.src_port %d)\n", c->rcv.src_port); DBG("Entered client config loockup (&c->rcv.src_ip %s)\n", ip_addr2a(&c->rcv.src_ip)); // ------------------------------------------------------------- dom = tls_lookup_cfg(cfg, TLS_DOMAIN_CLI, &c->rcv.dst_ip, c->rcv.dst_port, sname, srvid); } After making Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [parser/msg_parser.c:597]: parse_msg(): method: <INVITE> Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [parser/msg_parser.c:599]: parse_msg(): uri: <sip:9098@10.1.23.19:5061;transport=TLS> Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [parser/msg_parser.c:601]: parse_msg(): version: <SIP/2.0> I see Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [ip_addr.c:229]: print_ip(): tcpconn_new: new tcp connection: 10.1.23.19 Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [tcp_main.c:985]: tcpconn_new(): on port 5061, type 3 Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: <core> [tcp_main.c:1295]: tcpconn_add(): hashes: 1394:0:0, 1 Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:197]: tls_complete_init(): completing tls connection initialization Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:160]: tls_get_connect_server_name(): xavp with outbound server name not found Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:140]: tls_get_connect_server_id(): xavp with outbound server id not found Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:219]: tls_complete_init(): Entered client config loockup (c->rcv.dst_port 40123) Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:220]: tls_complete_init(): Entered client config loockup (&c->rcv.dst_ip 10.1.23.23) Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:221]: tls_complete_init(): Entered client config loockup (c->rcv.src_port 5061) Oct 26 09:23:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:222]: tls_complete_init(): Entered client config loockup (&c->rcv.src_ip 10.1.23.19) Where: &c->rcv.dst_ip 10.1.23.23 - it is my local kamailio tls socket ip address to make tls connect from c->rcv.dst_port 40123 - it is my local kamailio tls socket port &c->rcv.src_ip 10.1.23.19 - ip of my TLS device to make tls connection to c->rcv.src_port 5061 - port of my TLS device to make tls connection to so if I change line dom = tls_lookup_cfg(cfg, TLS_DOMAIN_CLI, &c->rcv.dst_ip, c->rcv.dst_port, sname, srvid); to dom = tls_lookup_cfg(cfg, TLS_DOMAIN_CLI, &c->rcv.src_ip, c->rcv.src_port, sname, srvid); I got correct client domain selection Oct 26 09:33:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_server.c:233]: tls_complete_init(): Using initial TLS domain TLSc<10.1.23.19:5061> (dom 0x7fd2eefa3d68 ctx 0x7fd2ef7e70a8 sn []) Oct 26 09:33:56 sip1 /usr/sbin/kamailio[20712]: DEBUG: tls [tls_domain.c:703]: sr_ssl_ctx_info_callback(): SSL handshake started Can you look at this code? Thank you in advance. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com tel: (+38067) 403-62-54

2 3

Kamailio + IMS + sipml5 call to softphone
by Serhat Guler 01 Nov '16

01 Nov '16

Hi, I have a setup as follows: IMS enabled on Kamailio and whereas websockets are enabled for PCSCF for webrtc calls. Calls(both audio and video) between to sipml5 clients using firefox web browser is possible. The session is setup for the calls from sipml5 to Mercuro, but then there isn't audio flow as the codecs are not compatible. Now I want to test it with Boghe which supports G.722, PCMA, PCMU, and OPUS codecs as firefox but this time the session isn't being setup. Boghe replies with "Reason: SIP; cause=488; text="Bad content" " I have seen a similar issue has been mentioned here: https://github.com/c00lz3r0/boghe/issues/157 but the initial invite request from sipml5 does have the SDP with media attributes.

3 8

ACC module and BYE attacks
by Gholamreza Sabery 01 Nov '16

01 Nov '16

If you enable acc module to log CDR data into a database a user can send an infinite number of BYE requests to the server and all of them will be logged into the database as "481 Call/transaction does not exist"! How can we prevent this?

3 5

/usr/sbin/kamailio - unresolvable destination error
by Ginti Saurabh 01 Nov '16

01 Nov '16

Hi, Please help to sort the problem. Getting "Unresolvable destination" error in /var/log/messages.

2 1

Terminating PCSCF unresolvable request URI (Kamailio IMS Websockets)
by Serhat Guler 01 Nov '16

01 Nov '16

Dear all, I am trying to make a call between 2 webrtc clients(sipml5) using my kamailio IMS setup. The invite request fails at the terminating PCSCF with error code "Status: 478 Unresolvable destination (478/TM)". I have traced the packages and saw that the package that arrives at the terminating PCSCF has an unresolvable request URI due to the websocket connection. The package being sent from terminating ICSCF to SCSCF has the "Request-URI: sip:bob@net1.test" The package from terminating SCSCF to terminating PCSCF has the "Request-URI: sip:bob@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws" I guess the terminating SCSCF replaces the request URI with the one it stored during the registration process, but the terminating PCSCF cannot resolve the host name&port number; therefore, it reports back "Unresolvable destination". Is that really the case happening in here ? How can I get over this problem ? Any ideas are welcome. Cheers, Serhat

2 1

Server error for SIP client
by Hermann Norpois 01 Nov '16

01 Nov '16

Hello, if my SIP client tries to connect ... a server error occured (see below). Could you please help me to figure out the reason? Kamailio is installed on an amazon instance with an elastic IP (52.208.249.214, hostname: synth.ddns.net). Thanks Hermann ubuntu@ip-172-31-32-10:~sudo /etc/init.d/kamailio restart * Stopping Kamailio SIP Server: kamailio [ OK ] * Starting Kamailio SIP Server: kamailio loading modules under config path: /usr/lib/x86_64-linux-gnu/kamailio/modules/ Listening on udp: 127.0.0.1:5060 udp: 172.31.32.10:5060 tcp: 127.0.0.1:5060 tcp: 172.31.32.10:5060 Aliases: tcp: ip-172-31-32-10.eu-west-1.compute.internal:5060 tcp: localhost:5060 udp: ip-172-31-32-10.eu-west-1.compute.internal:5060 udp: localhost:5060 *: synth.ddns.net:* ubuntu@ip-172-31-32-10:~$ kamctl db show subscriber +----+----------+----------------+----------+---------------+----------------------------------+----------------------------------+------+ | id | username | domain | password | email_address | ha1 | ha1b | rpid | +----+----------+----------------+----------+---------------+----------------------------------+----------------------------------+------+ | 1 | Johanna | synth.ddns.net | 12345 | | 4fa26cedd456d7105c01a48742e71e42 | 1e249ac341d95701f855fce18bec0fba | NULL | ubuntu@ip-172-31-32-10:~$ sudo ngrep -d any -qt -W byline . port 5060 Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.5ad567803b13610279818912da2fcd81.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.6a2baefed6cf4f6b6d34c7563d829e44.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.6d74cec9e5f5e46dd9cd67338143127f.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.a3a37d6bd0cd5d68fc786568f587f06e.0. Via: SIP/2.0/UDP 188.174.185.176:5060 ;received=188.174.185.176;rport=5060;branch=z9hG4bKPjAgF18hfvv20KOQ9H8G-fqfIPMjjcu9wq. From: <sip:Johanna@synth.ddns.net>;tag=vzlAJNNNES67ir6nsLMjvyFh5D0Thy6s. To: <sip:Johanna@synth.ddns.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.3300. Call-ID: 9Uj4TSeQAVwaMs3UHf8UKIhuoA4xBEZt. CSeq: 39586 REGISTER. Server: kamailio (4.3.6 (x86_64/linux)). Content-Length: 0. . U 2016/10/20 13:44:33.727161 52.208.249.214:5060 -> 172.31.32.10:5060 SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL). Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.2605d82fd11bd8f788cfad582884eb6d.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.e425ad98e0436abc4661d1452a49670d.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.2d047656fb9df0edfccd4a40d20237b7.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.f54ab9c35b22814b15acbe16bcae387f.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.db5960988332c2f4899a5e1c2b57e3c6.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.62f1191bd5af8480698ce118984536c0.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.5ad567803b13610279818912da2fcd81.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.6a2baefed6cf4f6b6d34c7563d829e44.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.6d74cec9e5f5e46dd9cd67338143127f.0. Via: SIP/2.0/UDP 172.31.32.10;received=52.208.249.214;branch=z9hG4bK533f.a3a37d6bd0cd5d68fc786568f587f06e.0. Via: SIP/2.0/UDP 188.174.185.176:5060 ;received=188.174.185.176;rport=5060;branch=z9hG4bKPjAgF18hfvv20KOQ9H8G-fqfIPMjjcu9wq. From: <sip:Johanna@synth.ddns.net>;tag=vzlAJNNNES67ir6nsLMjvyFh5D0Thy6s. To: <sip:Johanna@synth.ddns.net>;tag=b27e1a1d33761e85846fc98f5f3a7e58.3300. Call-ID: 9Uj4TSeQAVwaMs3UHf8UKIhuoA4xBEZt. CSeq: 39586 REGISTER. Server: kamailio (4.3.6 (x86_64/linux)). Content-Length: 0. 2) My logfile /var/log/kamailio/kamailio.log is empty though /etc/rsyslog.d/kamailio.conf exists with the entry: local0.* -/var/log/kamailio/kamailio.log And in /etc/kamailio/kamailio.conf: log_facility=LOG_LOCAL0 Why is the logfile empty? Thanks Herman

2 1

Store and add custom variables for sip registrations
by Markus Bönke 01 Nov '16

01 Nov '16

Hallo all, we want to use kamailio as registrar server in front of freeswitch. When a user registers we need to lookup and store some user variables. To every INVITE that is received from a registered user we want to add the variables as x sip headers before we relay them to freeswitch. The question is what is the best way to implement this? We tried to store the variables in a htable on register and then lookup and attach the variables on every INVITE from there. Is this the correct approach, one issue we see is that htable is not writing new inserts directly to the database. I’ve also seen that there is a table location_atts in the kamailio db which looks like that it is intended to store custom key value pairs for registrations, but the seems to be no module which writes to that table, or is there one? Thanks for any help or suggestions Markus

2 1

TSILO module use failure
by Igor Olhovskiy 31 Oct '16

31 Oct '16

Hi! I’m trying to implement scheme much like Federico Cabiddu is described here https://www.youtube.com/watch?v=4XIrR9bwUkM Done a bit modification to his script, cause it’s not fully correct on 4.4 version So, parts of config looks like …. route[REGISTRAR] { … if (!save("location")) { sl_reply_error(); } else { route(PUSHJOIN); } exit; …. } route[LOCATION] { if (!lookup("location")) { …… if (is_method("INVITE")) { send_reply("100", "Trying"); route(SUSPEND); } } else { if (is_method("INVITE")) { if (t_newtran()) { ts_store("sip:$tU@$fd"); } $sht(vtp=>stored::$rU) = 1; } } route(RELAY); } ….. route[SUSPEND] { if(!t_suspend()) { send_reply("501", "Unknown destination"); exit; } $sht(vtp=>join::$rU) = "" + $T(id_index) + ":" + $T(id_label); } # append branches or resume the transaction route[PUSHJOIN] { … #if was suspended - recover, if not - ts_append ….. ts_append(«location", "sip:$tU@$fd"); return; … t_continue("$var(id_index)", "$var(id_label)", "INVRESUME"); } # lookup and relay after resuming transaction route[INVRESUME] { lookup("location"); t_relay(); ts_store("sip:$tU@$fd"); $sht(vtp=>stored::$rU) = 1; } Branches on ts_store are saved in format (sip:username@domain) and also in this format they are added on ts_append. All regarding to routes working ok, except on ts_append I got ERROR: tm [t_append_branches.c:168]: t_append_branches(): ERROR: t_append_branch: failure to add branches (-1) Also, it’s a mixed environment, like wss + udp + tcp. Any idea why? Thanks. -- Best regards, Igor

2 3

Re: [SR-Users] PCSCF cannot create via header for sipml5 ACK package
by Serhat Guler 30 Oct '16

30 Oct '16

Hi all, I am still stuck with the ACK message not being forwarded by the originating PCSCF. Any advice would be great. Thanks, Serhat On 24 October 2016 at 21:00, Serhat Guler <srtguler(a)gmail.com> wrote: > Hi Daniel, > > I am using only record_route() without any parameters. I do not have a > proper computer atm to draw the network diagram, but I can tell you shortly > about the network setup. > > I have only enabled websockets for the pcscf to allow ws and wss > connections. In that case there is a ws connection that uses UDP protocol. > This is the ACK to complete the session setup. > > the sipml5 client is configured as follows: > WebSocket Server URL: ws://192.168.0.11:880 > SIP outbound Proxy URL: udp://192.168.0.11:4060 > > Mercuro IMS client: uses UDP port as well: 4060 > > The call is made from sipml5 client. The Mercuro phone rings, and when I > reply the call, 200 OK is sent to sipml5 webrtc client, but the ACK from > sipml5 doesn't pass the PCSCF as I explained in the previous message. > > A part of PCSCF cfg file: > > # Check for Subsequent requests: > if (has_totag()) { > # sequential request withing a dialog should > # take the path determined by record-routing > if (loose_route()) { > if ($route_uri =~ "sip:mo@.*") { > setflag(FLT_MO); > } > if(!isdsturiset()) { > handle_ruri_alias(); > } > # RTP-Relay, if necessary > route(RTPPROXY); > t_relay(); > } else { > if ( is_method("ACK") ) { > if ( t_check_trans() ) { > # no loose-route, but stateful ACK; > # must be an ACK after a 487 > # or e.g. 404 from upstream server > t_relay(); > exit; > } else { > xlog("L_INFO", "ACK without matching transaction ... > ignore and discard!!!!!\n"); > # ACK without matching transaction ... ignore and > discard > exit; > } > } > sl_send_reply("404","Not here"); > } > exit; > } > > Cheers, > Serhat > > > > On 24 October 2016 at 20:18, Daniel-Constantin Mierla <miconda(a)gmail.com> > wrote: > >> Hello, >> >> I haven't noticed the log files, it's ok. >> >> From the Route header, I see that there is a proxy that uses WS: >> >> Route: <sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy >> 1nCMEI1mR0RztrB;did=e82.0c3> >> That is the address of the next hop and typically a proxy doesn't use >> websocket connection to another proxy. Can you show a diagram with the sip >> server nodes in your network and what protocols are used between them? >> >> Are you simply use record_route() function, or some other function or >> different parameters to it? >> >> Cheers, >> Daniel >> >> >> On 24/10/16 12:18, Serhat Guler wrote: >> >> Hi Daniel, >> >> Thanks for your reply. I actually attached a log file with debug level 3, >> consisting ACK related messages. If you would like to see more logs, I'll >> send a new log file in the evening. >> >> Cheers, >> Serhat >> >> On 24 October 2016 at 12:13, Daniel-Constantin Mierla <miconda(a)gmail.com> >> wrote: >> >>> Hello, >>> >>> can you get all the log messages for ACK but with debug=3 in the >>> kamailio.cfg? >>> >>> Cheers, >>> Daniel >>> >>> On 23/10/16 22:04, Serhat Guler wrote: >>> >>> Hello, >>> >>> I finally managed to place a call from sipml5 webrtc client to Mercuro >>> IMS client. The phone rings, and when I answer it sends 200 OK to the >>> sipml5 where as sipml5 send back an ACK message which never passes the >>> originating PCSCF. The PCSCF says: >>> >>> 8(3640) WARNING: <core> [msg_translator.c:2729]: via_builder(): TCP/TLS >>> connection (id: 0) for WebSocket could not be found >>> 8(3640) ERROR: <core> [msg_translator.c:1947]: >>> build_req_buf_from_sip_req(): could not create Via header >>> 8(3640) ERROR: <core> [forward.c:548]: forward_request(): building >>> failed >>> >>> I doubt that the WebSocket connection is closed, cause when I terminate >>> the call from Mercuro client a bye request is being sent to the sipml5. >>> >>> The ACK package: >>> >>> ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2. >>> Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9 >>> hG4bKvuly7bmxnN4aqM4zZTIS;rport >>> From: "Bob"<sip:bob@net1.test>;tag=GxzKy1nCMEI1mR0RztrB >>> To: <sip:alice@net1.test>;tag=18823 >>> Contact: "Bob"<sip:bob@df7jal23ls0d.invalid;rtcweb-breaker=no;click2c >>> all=no;transport=ws>;+g.oma.sip-im;language="en,fr" >>> Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6 >>> CSeq: 3887 ACK >>> Content-Length: >>> Route: <sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp> >>> Max-Forwards: 69 >>> Route: <sip:mo@192.168.0.11:880;transport=ws;r2=on;lr=on;ftag=GxzKy >>> 1nCMEI1mR0RztrB;did=e82.0c3> >>> Route: <sip:mo@192.168.0.11:4060;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0Rz >>> trB;did=e82.0c3> >>> Route: <sip:mo@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di >>> d=e82.f062> >>> Route: <sip:mt@192.168.0.11:6060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di >>> d=e82.f062> >>> Route: <sip:mt@192.168.0.11:4060;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;di >>> d=e82.1c3> >>> User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 >>> Organization: Doubango Telecom >>> >>> Have been thinking for quite a while, but couldn't really find a reason >>> why it wouldn't add the v,a header. A debug 3 level log file is also >>> attached. >>> >>> Thanks in advance, >>> Serhat >>> >>> >>> _______________________________________________ >>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users >>> >>> -- >>> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com >>> >>> _______________________________________________ SIP Express Router >>> (SER) and Kamailio (OpenSER) - sr-users mailing list >>> sr-users(a)lists.sip-router.org http://lists.sip-router.org/cg >>> i-bin/mailman/listinfo/sr-users >> >> -- >> Daniel-Constantin Mierlahttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com >> >> >

2 3

digest authentication based on two password columns
by Jayesh Nambiar 28 Oct '16

28 Oct '16

Hello, I was wondering if it is possible to do a failover in digest authentication. For eg: I get two possible valid password values. Can I do a pv_proxy_authenticate in such a way where if authentication fails, I do a pv_proxy_autenticate again on the different value to check if that succeeds? Is it even allowed in digest authentication mechanism. Just getting opinions. Thanks for any help. - Jayesh

3 3

Re: [SR-Users] fetch contact
by Slava Bendersky 28 Oct '16

28 Oct '16

Hello Daniel, No that exactly the one, Slava. From: "Daniel-Constantin Mierla" <miconda(a)gmail.com> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Friday, 28 October, 2016 07:37:31 Subject: Re: [SR-Users] fetch contact Hello, the contact address is one for attributes in $ulc(...), iirc, it is the =>addr -- see the wiki for pv cookbook for more details. Or are you looking for something else? Cheers, Daniel On 27/10/16 21:43, Slava Bendersky wrote: Hello Everyone, How to fetch contact from location table ? I tried look reg_fetch_contacts(), but I don't specific $ulc for contact. Thanks _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 28-30, 2016 - http://www.asipto.com _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Errors when parsing Diversion header with multiple comma-separated diversions
by Björn Bylander 28 Oct '16

28 Oct '16

Hello! Given a Diversion header like the following: Diversion:"Foo Bar"<sip:+9876543211@127.0.0.101;user=phone>;privacy=off;answered-count=2;re ason=deflection;counter=1;answered,"_ somewhere"<sip:+9876543212@127.0.0.101;user=phone>;privacy=off;answered-coun t=1;reason=deflection;counter=1;answered Kamailio 4.4.3 (and probably earlier versions) emit error messages complaining about the comma character. According to RFC 6044 multiple comma-separated diversions in a single Diversion-header is allowed. Full INVITE as an example: INVITE sip:+1234567890@127.0.0.100:5060;user=phone;transport=tcp SIP/2.0 Via:SIP/2.0/TCP 127.0.0.101;branch=z9hG4bKBroadWorks.-1od4eg8-127.0.0.100V5060-0-82844591-13 54681939-1477634439005- From:<sip:+9876543210@127.0.0.101;user=phone>;tag=1354681939-1477634439005- To:<sip:+1234567890@127.0.0.100:5060;user=phone> Call-ID:BW0800390052810161562691307@127.0.0.101 CSeq:82844591 INVITE Contact:<sip:127.0.0.101:5060;transport=tcp> P-Asserted-Identity:<sip:+9876543210@127.0.0.101;user=phone> Privacy:none Diversion:"Foo Bar"<sip:+9876543211@127.0.0.101;user=phone>;privacy=off;answered-count=2;re ason=deflection;counter=1;answered,"_ somewhere"<sip:+9876543212@127.0.0.101;user=phone>;privacy=off;answered-coun t=1;reason=deflection;counter=1;answered Supported:100rel Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE X-BroadWorks-DNC:network-address="sip:+9876543210@127.0.0.101;user=phone" Accept:application/dtmf-relay,application/media_control+xml,application/sdp, application/x-broadworks-call-center+xml,multipart/mixed Max-Forwards:10 Content-Type:application/sdp Content-Disposition:session;handling=required Content-Length:267 v=0 o=BroadWorks 100442303 1 IN IP4 127.0.0.101 s=- c=IN IP4 127.0.0.102 t=0 0 m=audio 29778 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 Example log messages: Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [parser/parse_addr_spec.c:488]: parse_to_param(): ERROR parse_to_param : invalid character ',' in status 21: <<;privacy=off;answered-count=2;reason=deflection;counter=1;answered>> Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [parser/parse_addr_spec.c:254]: parse_to_param(): ERROR: parse_to_param :unexpected char ["] in status 21: <<;privacy=off;answered-count=2;reason=deflection;counter=1;answered,>> . Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [parser/parse_addr_spec.c:906]: parse_addr_spec(): ERROR: parse_to: invalid To - unexpected end of header in state 21 Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [parser/parse_diversion.c:69]: parse_diversion_header(): ERROR:parse_diversion_header: bad diversion header Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [lvalue.c:345]: lval_pvar_assign(): non existing right pvar Oct 28 08:00:39 kam1 /usr/sbin/kamailio[14935]: ERROR: <core> [lvalue.c:405]: lval_assign(): assignment failed at pos: (562,21-562,35) The code on line 562 is $avp(div_user) = $(di{uri.user});. Can this be handled in any way by Kamailio without changing the parser? Best regards, Björn Bylander

2 1

fetch contact
by Slava Bendersky 28 Oct '16

28 Oct '16

Hello Everyone, How to fetch contact from location table ? I tried look reg_fetch_contacts(), but I don't specific $ulc for contact. Thanks

2 1

logfile is empty
by Hermann Norpois 27 Oct '16

27 Oct '16

my logfile /var/log/kamailio/kamailio.log is empty. 1) /etc/rsyslog.d/kamailio.cfg exists with the entry: log_facility=LOG_LOCAL0 2) /etc/rsyslog.con has the entry: # # don't log messages with LOG_LOCAL0 in /var/log/syslog anymore *.*;auth,authpriv.none,local0.none -/var/log/syslog # # log messages with LOG_LOCAL0 in /var/log/kamailio.log local0.* -/var/log/kamailio.log # local0.* -/var/log/kamailio/kamailio/kamailio.log Alternatively I edited /etc/rsyslog.d/kamailio.conf that should be used by rsylog.conf 3) /etc/rsyslog.d/kamailio.conf exists with the entry: local0.* -/var/log/kamailio/kamailio.log I am working with ubuntu on an amazon instance. Why is the logfile empty? Thanks Herman

3 2

How do I see the full SDP?
by Martin Taschen 27 Oct '16

27 Oct '16

Hello, kamailio seems to work fine. After registration I would like to see the complete SDP of an user. . How is this done? Thanks Martin Taschen

2 1

Re: [SR-Users] sr-users Digest, Vol 137, Issue 25
by surendra tiwari 27 Oct '16

27 Oct '16

i used the cnxcc module its successfully terminated the call. but if you want i can help you. On Thu, Oct 27, 2016 at 12:58 AM, <sr-users-request(a)lists.sip-router.org> wrote: > Send sr-users mailing list submissions to > sr-users(a)lists.sip-router.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > or, via email, send a message with subject or body 'help' to > sr-users-request(a)lists.sip-router.org > > You can reach the person managing the list at > sr-users-owner(a)lists.sip-router.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of sr-users digest..." > > > Today's Topics: > > 1. Re: cnxcc doesn't terminate calls (Igor Potjevlesch) > 2. logfile is empty (Hermann Norpois) > 3. Re: Out of memory in UB 210: OOM killed process 12261 > (kamailio) score 0 vm:1614768kB, rss:280200kB, swap:131408kB > (Jurijs Ivolga) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 26 Oct 2016 12:48:29 +0200 > From: "Igor Potjevlesch" <igor.potjevlesch(a)gmail.com> > To: "'Kamailio $SER$ - Users Mailing List'" > <sr-users(a)lists.sip-router.org> > Subject: Re: [SR-Users] cnxcc doesn't terminate calls > Message-ID: <009c01d22f76$7d0c9a30$7725ce90$(a)gmail.com> > Content-Type: text/plain; charset="utf-8" > > Hi, > > > > Yep, the dialog is created. I put the dialog into DB and see it. > > Nothing appears in the logs when the timer is reached. > > > > I succeed to do the same behaviour with dialog module only and > dlg_set_timeout. Looks to do why I want. > > > > I'm a bit frustrating about cnxcc because I really don't understand why > nothing happen after max_time, but, anyway, it works with dialog only. > > > > Regards, > > > > Igor. > > > > De : sr-users [mailto:sr-users-bounces@lists.sip-router.org] De la part > de Arsen > Envoyé : mercredi 26 octobre 2016 11:15 > À : Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> > Objet : Re: [SR-Users] cnxcc doesn't terminate calls > > > > Hi Igor, > > > > Make sure that you create a dialog for this call and $dlg_var(key) is > available (This pseudo-variable will be available only for subsequential > requests after doing loose_route().) > > > > Also check kamailio logs maybe you try to terminate call for wrong > customer id. > > > > On Wed, Oct 26, 2016 at 10:52 AM, Igor Potjevlesch < > igor.potjevlesch(a)gmail.com <mailto:igor.potjevlesch@gmail.com> > wrote: > > Hi, > > > > I'm testing cnxcc module in order to make a simple call duration > limitation. > > My understanding is to use the function like this : > > > > $dlg_var(customer_id) = "customer-" + $fU; > > $var(max_time) = 5; > > > > if (!cnxcc_set_max_time("$dlg_var(customer_id)", > "$var(max_time)")) > > > > Unfortunately, after 5 seconds, nothing happen. I don't even see an error > from dialog module. > > > > Anyone has a better understanding of the way the module runs? Thank you! > > > > Regards, > > > > Igor. > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > > > > > > -- > > Regards, > Arsen. > >

1 0

cnxcc doesn't terminate calls
by Igor Potjevlesch 26 Oct '16

26 Oct '16

Hi, I'm testing cnxcc module in order to make a simple call duration limitation. My understanding is to use the function like this : $dlg_var(customer_id) = "customer-" + $fU; $var(max_time) = 5; if (!cnxcc_set_max_time("$dlg_var(customer_id)", "$var(max_time)")) Unfortunately, after 5 seconds, nothing happen. I don't even see an error from dialog module. Anyone has a better understanding of the way the module runs? Thank you! Regards, Igor.

2 2

app_perl modifies Via header?
by Igor Potjevlesch 25 Oct '16

25 Oct '16

Hello, I try to use app_perl and cnxcc module. As soon as I load the module like this: # --------- app_perl params ---- modparam("app_perl", "filename", "/usr/local/etc/kamailio/AAA/aaa-script.pl") modparam("app_perl", "modpath", "/usr/local/lib64/kamailio/perl/") The Via header is modified for each calls: /usr/local/sbin/kamailio[21064]: INFO: <core> [parser/msg_parser.c:355]: parse_headers(): ERROR: bad header field [Via:ASIP/2.0/UDP 91.] /usr/local/sbin/kamailio[21064]: ERROR: <core> [parser/parse_via.c:2048]: parse_via(): ERROR: parse_via: bad char <A> on state 100 What could explain this behaviour? Regards, Igor.

2 4

Kamailio SIP TLS issue
by Sergey Basov 25 Oct '16

25 Oct '16

Hi All I have some strange behavior of kamailio with TLS. I have configured second IP addres on server, added it to tls listener, and tls.cfg file. But when I try to connect using openssl s_client -showcerts -connect 10.1.23.33:5061 -tls1 -state and openssl s_client -showcerts -connect 10.1.23.23:5061 -tls1 -state I see same certificates (sip2 my config samples are bellow) if I make changes in port number (for ip 10.1.23.33 set port 5091 in both config parts) - I see correct certificates. Does anyone have this problem? Thanks in advance. ----- listen section ---- listen=tls:10.1.23.23:5061 listen=tls:10.1.23.33:5061 ----- tls.cfg ------ [server:default] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/keys/sip1.key certificate = /etc/kamailio/keys/sip1.crt [server:10.1.23.33:5061] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/keys/sip1.key certificate = /etc/kamailio/keys/sip1.crt [server:10.1.23.23:5061] method = TLSv1+ verify_certificate = no require_certificate = no private_key = /etc/kamailio/keys/sip2.key certificate = /etc/kamailio/keys/sip2.crt -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.com

2 3

Sending INVITE’s on REGISTER
by Igor Olhovskiy 25 Oct '16

25 Oct '16

Hi! I have this kind of scenario Receive an INVITE -> send it to user, but if possible, if I’ll receive REGISTER with same credentials I’m calling to - drop an INVITE there also, if call was not answered on first location. Means create other INVITE branch async on the fly, if possible. Where at least to look at? Thanks. -- Best regards, Igor

2 2

How compile Kamailio in freebsd : add carrierroute module
by AFRICALLSHOP 25 Oct '16

25 Oct '16

Hello, I try to compile kamailio in freebsd. I want to add carrierroute module. I do : 1) cd /usr/ports/net/kamailio 2) make install but carrierroute module is not compiled. Could you help me? Regards Abdoul.

3 2

How compile Kamailio in freebsd : add carrierroute module
by Abdoul Osséni 24 Oct '16

24 Oct '16

Hello, I try to compile kamailio in freebsd. I want to add carrierroute module. I did : 1) cd /usr/ports/net/kamailio/ 2) make install but carrierroute module is not compiled. Could you help me? Regards Abdoul.

1 0

Rewriting To on BYE
by Charles Phillips 24 Oct '16

24 Oct '16

I need a sanity check if you don’t mind… I am setting up a stateless proxy for several (separate, not load balanced) Asterisk servers behind Kamailio 4.2 (with domain and dispatcher). All works as expected except when called party hangs up, as Asterisk seems to address the BYE “To” (that is sent to the calling party) to the originating domain, and not the IP of the UA (as it does in other cases). When this happens, stateless Kamailio does not seem to know how to route them, which makes sense. I assume I would need to rewrite the “To” field on BYE message so that it contains the UA’s IP (to match the BYE’s that are routed correctly). The RURI header contains the IP/port of the UA, so I am attempting to use it and here is where I am stuck… Here is the subst from onreply_route: if (is_method("BYE")) { xlog("L_NOTICE", "BYE Detected \n"); subst('/^To:(.*)<sip:(.*)@(.*)>(.*)$/To: <sip:\2@$rd:$rp>\4/ig'); } Here is the “To”: To: "Test User 2" <sip:102@zxcv.asdf.qwerty.net>;tag=8AB53900B9646F1DF05330E7FBF846C8 Any obvious glaring mistakes here? Would I then need additional logic to forward the BYE on? My apologies if this is wordy, or nonsensical...

2 1

PCSCF cannot create via header for sipml5 ACK package
by Serhat Guler 24 Oct '16

24 Oct '16

Hello, I finally managed to place a call from sipml5 webrtc client to Mercuro IMS client. The phone rings, and when I answer it sends 200 OK to the sipml5 where as sipml5 send back an ACK message which never passes the originating PCSCF. The PCSCF says: 8(3640) WARNING: <core> [msg_translator.c:2729]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found 8(3640) ERROR: <core> [msg_translator.c:1947]: build_req_buf_from_sip_req(): could not create Via header 8(3640) ERROR: <core> [forward.c:548]: forward_request(): building failed I doubt that the WebSocket connection is closed, cause when I terminate the call from Mercuro client a bye request is being sent to the sipml5. The ACK package: ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2. Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvuly7bmxnN4aqM4zZTIS;rport From: "Bob"<sip:bob@net1.test>;tag=GxzKy1nCMEI1mR0RztrB To: <sip:alice@net1.test>;tag=18823 Contact: "Bob"<sip:bob@df7jal23ls0d.invalid ;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr" Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6 CSeq: 3887 ACK Content-Length: Route: <sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp> Max-Forwards: 69 Route: <sip:mo@192.168.0.11:880 ;transport=ws;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.0c3> Route: <sip:mo@192.168.0.11:4060 ;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.0c3> Route: <sip:mo@192.168.0.11:6060 ;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.f062> Route: <sip:mt@192.168.0.11:6060 ;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.f062> Route: <sip:mt@192.168.0.11:4060 ;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.1c3> User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04 Organization: Doubango Telecom Have been thinking for quite a while, but couldn't really find a reason why it wouldn't add the v,a header. A debug 3 level log file is also attached. Thanks in advance, Serhat

2 3

configuration kamailio 4.4.2 with Asterisk 13.10.0
by Serge Gabriel Ahnoux 24 Oct '16

24 Oct '16

Hello I am trying to configure kamailio 4.4.2 with asterisk 13.10.0. My Os is Debian 7.4 I followed the tutorial : http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb… The two softwares are installed on my server. So they have the same IP adress but the port is different like they said in the tutorial. I use too Mysql to stock users informations I have 2 databases Kamailio and Asterisk. I use ODBC for communicate beetwen asterisk server and his database. However when my sip client (linephone) connects to kamalio, kamailio doesn't forward information to asterisk. I don't kow the reason. So i can send message but not call If someone knows how to configure, please help me. I join my kamailio configuration file. Regards. ahnouxs(a)gmail.com

2 1

Dropping non-provisional replies in onreply_route
by Alex Balashov 21 Oct '16

21 Oct '16

The core documentation says that in a named onreply_route[], only provisional replies can be drop()'d. To drop any reply, it is necessary to use a global onreply_route. Is there any workaround for this, i.e. so I can drop a 2xx reply from a specific TM transaction? The reason is, to know whether to drop it, I need access to either AVPs or, ideally, dialog variables. Since the global onreply_route is executed by the core, I presume I won't have access to anything that persists through TM there. Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 5

Edge proxy / Registrar Script
by Infinicalls Infinicalls 21 Oct '16

21 Oct '16

Hi, I am trying to get a simple Edge Proxy - Registrar setup working behind NAT. Initially I tried various scripts - including the one mentioned in the Oubound page and Peter Dunkley' example - but got some errors related to PATH. Can somebody give me a working script of both proxy and registrar? Any help would be gladly appreciated. regards Ganesh Kumar -- --- http://www.infinicalls.com

2 2

Basic TM suspend question
by Alex Balashov 21 Oct '16

21 Oct '16

Hi, If a transaction is currently t_suspended() and has not been t_continued(), will t_check_trans() still find that transaction for retransmission-dampening purposes, e.g. if a retransmission of the same request as the original one is received? Thanks, -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 3

Postgres: Exclusive Transaction Locks vs. Vacuumer
by Stefan Kohlhauser 21 Oct '16

21 Oct '16

Hey everyone! We noticed that under high load of registrations and using the pua module we can reach a state where the PUA table bloats our Postgres to several GB. However, a SELECT on pua only shows a few hundred entries. The other half million or so entries are obsoleted but not cleaned up. The reason for this seems to be that the autovacuumer of the Postgres can't get to work because the PUA table is almost permanently locked. Our Postgres logs are full with errors about the autovacuumer being canceled because it couldn't get to work. Looking at modules/db_postgres/km_dbase.c::db_postgres_start_transaction() I noticed that there are two locking modes used: str lock_write_end_str = str_init(" IN EXCLUSIVE MODE"); str lock_full_end_str = str_init(" IN ACCESS EXCLUSIVE MODE"); According to https://www.postgresql.org/docs/9.2/static/explicit-locking.html these modes mean forbidding everything else except reads and forbidding everything else. However, according to this documentation ROW EXCLUSIVE is the standard lock for entry altering operations (INSERT, UPDATE, DELETE). And this would still allow the autovacuumer to do its job. Is there a specific reason why these more restricting locking modes are used instead of ROW EXCLUSIVE? Thanks. Best regards, Stefan

2 1

TCP FIN after 10 minutes
by gmele 20 Oct '16

20 Oct '16

Hello, we have deployed a Kamailio acting as SIP proxy on a RHEL 7.2 machine. Clients (mainly mobile phones) connects to the proxy using a TLS protected TCP connection. In the kamailio config, we've set : #!ifdef WITH_TLS enable_tls=yes tcp_async=yes tcp_connection_lifetime=3605 tcp_accept_no_cl=yes tcp_crlf_ping = yes #!endif Our problem is that, even if we set the tcp_connection_lifetime to > 1 hour, the tcp connection is closed after 10 minutes: on tcp dumps, we see clearly the TCP FIN sent by machine hosting the kamailio proxy... Setting parameters tcp_keepidle/keepintlv/keepcnt in the kamailio config didn't change the behavior. We also set TCP keepalived at system level, but without result... This TCP closure is causing us lot of problems when calls between 2 UAs last more than 10 minutes because the REINVITE or BYE messages are lost. Also, closing the TCP connection will wake up the mobile app and make it resend a REGISTER, thing we absolutely want to avoid. Is there a parameter we can use to avoid this closure? I had a look in previous posts, found people with the same problem as mine, but no real solution... Thx for your help. Giovanni -- View this message in context: http://sip-router.1086192.n5.nabble.com/TCP-FIN-after-10-minutes-tp152646.h… Sent from the Users mailing list archive at Nabble.com.

2 2

Re: [SR-Users] help with NOTIFY
by Slava Bendersky 20 Oct '16

20 Oct '16

Hello Everyone, This message as continue conversation from http://lists.sip-router.org/pipermail/sr-users/2015-March/087557.html. That my previous post about it. I never was be be able forward NOTIFY from asterisk to client through kamailio. Right now in use asterisk 14 pjsip. Any help thank you. volga629 From: "volga629" <volga629(a)skillsearch.ca> To: "sr-users" <sr-users(a)lists.sip-router.org> Sent: Wednesday, 19 October, 2016 18:36:52 Subject: [SR-Users] help with NOTIFY Hello Everyone, I am trying forward NOTIFY to client. My setup is PBX Server ---- local lan ---kamailio ---- internet ---- client The problem with whole setup that kamailio is not forwarding NOTIFY to client. Here are log. U 2016/10/19 14:23:29.581609 10.18.130.50:5160 -> 10.18.130.46:5060 NOTIFY sip:10101@10.18.130.46:5060 SIP/2.0. Via: SIP/2.0/UDP 10.18.130.50:5160;rport;branch=z9hG4bKPjONocd4FvI5LkOmSR6ZCA5Z1f7SYDb3Ty. From: <sip:8101@networklab.ca>;tag=ziApds23FQsPL2zd6q8pCUlwW.jwa6mK. To: "Slava" <sip:10101@networklab.ca>;tag=734f813f. Contact: <sip:10.18.130.50:5160>. Call-ID: A0bJWg8v0nGmtUmiHzUixQ... CSeq: 18547 NOTIFY. Route: <sip:kamailio_local_ip:5060;lr;r2=on;ftag=734f813f>. Route: <sip:kamailio_public_ip:5081;transport=tls;lr;r2=on;ftag=734f813f>. Event: presence. Subscription-State: active;expires=413. Allow-Events: message-summary, presence, dialog, refer. Max-Forwards: 70. User-Agent: FPBX-13.0.188.9(14.0.2). Content-Type: application/pidf+xml. Content-Length: 579. . <?xml version="1.0" encoding="UTF-8"?> <presence entity="sip:8101@10.18.130.50:5160" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:pp="urn:ietf:params:xml:ns:pidf:person" xmlns:es="urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status" xmlns:ep="urn:ietf:params:xml:ns:pidf:rpid:rpid-person"> <note>On the phone</note> <tuple id="8101"> <status> <basic>closed</basic> </status> <contact priority="1">"Slava" <sip:10101@networklab.ca></contact> </tuple> <pp:person> <status> <ep:activities>ep:busy</ep:activities> </status> </pp:person> </presence> U 2016/10/19 14:23:32.045388 10.18.130.46:5060 -> 10.18.130.50:5160 SIP/2.0 408 Request Timeout. Via: SIP/2.0/UDP 10.18.130.50:5160;rport=5160;branch=z9hG4bKPju05rMNlzhfukLvaq4ELmq.QI4GBqDN-m;received=10.18.130.50. From: <sip:8101@networklab.ca>;tag=Yzy2m6O2AFT6rGSDBmo6PjcGaCWT1i5H. To: "Slava" <sip:10101@networklab.ca>;tag=ad5a945e. Call-ID: 2pq5iebmtLdB_OsMUs3EOg... CSeq: 14623 NOTIFY. Server: kamailio (4.3.4 (x86_64/linux)). Content-Length: 0. Any help thank you. volga629 _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

sipml5 fail invite kamailio ims web sockets
by Serhat Guler 20 Oct '16

20 Oct '16

Dear all, I can register the sipml5 client to my kamailio IMS setup successfully, but when I try to make a call the invite fails with "478 Unresolvable destination". From the wireshark trace file I think the PCSCF on the terminating side cannot resolve the destination. After the invite request fails the caller re-registers to the networkj automatically. The wireshark file is attached. The registration process happens successfully but all of the nodes give some errors: 10(3773) INFO: <script>: [REGISTER] from [sip:alice@net1.test] to [sip:alice@net1.test] 10(3773) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=425 a=25 n=xlog 10(3773) ERROR: <script>: REGISTER (sip:alice@net1.test (192.168.0.10:14115) to sip:alice@net1.test, 25c80750-d52d-30c8-d373-c0136401a927) 10(3773) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=437 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[DEFAULT_ROUTE] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=437 a=5 n=route 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=627 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=622 a=25 n=mf_process_maxfwd_header 10(3773) DEBUG: <core> [parser/msg_parser.c:153]: get_hdr_field(): get_hdr_field: cseq <CSeq>: <59840> <REGISTER> 10(3773) DEBUG: <core> [parser/msg_parser.c:187]: get_hdr_field(): DEBUG: get_hdr_body : content_length=0 10(3773) DEBUG: maxfwd [mf_funcs.c:85]: is_maxfwd_present(): value = 70 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=633 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=627 a=26 n=sanity_check 10(3773) DEBUG: <core> [parser/msg_parser.c:89]: get_hdr_field(): found end of header 10(3773) DEBUG: sanity [mod_sanity.c:248]: w_sanity_check(): sanity checks result: 1 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=639 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=633 a=24 n=has_totag 10(3773) DEBUG: siputils [checks.c:97]: has_totag(): no totag 10(3773) DEBUG: <core> [select.c:412]: run_select(): Calling SELECT 0xb6e5fe40 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=645 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=639 a=25 n=is_method 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=649 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=645 a=24 n=t_lookup_request 10(3773) DEBUG: tm [t_lookup.c:1011]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=1 T start=0xffffffff 10(3773) DEBUG: tm [t_lookup.c:466]: t_lookup_request(): t_lookup_request: start searching: hash=21728, isACK=0 10(3773) DEBUG: tm [t_lookup.c:424]: matching_3261(): DEBUG: RFC3261 transaction matching failed 10(3773) DEBUG: tm [t_lookup.c:648]: t_lookup_request(): DEBUG: t_lookup_request: no transaction found 10(3773) DEBUG: tm [t_lookup.c:1080]: t_check_msg(): DEBUG: t_check_msg: msg id=2 global id=2 T end=(nil) 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=652 a=16 n=if 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=649 a=25 n=is_method 10(3773) ERROR: *** cfgtrace:request_route=[REQINIT] c=[/usr/local/etc/kamailio/kamailio-pcscf.cfg] l=650 a=26 n=send_reply 10(3773) DEBUG: sl [sl.c:280]: send_reply(): reply in stateless mode (sl) 10(3773) DEBUG: websocket [ws_conn.c:429]: wsconn_get(): wsconn_get for id [2] 10(3773) DEBUG: websocket [ws_conn.c:437]: wsconn_get(): wsconn_get returns wsc [0xb2329228] refcnt [2] 10(3773) DEBUG: websocket [ws_frame.c:751]: ws_frame_transmit(): Tx message: The SCSCF says (while inviting): 0(3582) ERROR: dialog_ng [dlg_handlers.c:196]: populate_leg_info(): bad sip message or missing Contact hdr 0(3582) ERROR: dialog_ng [dlg_handlers.c:453]: dlg_onreply(): could not add further info to the dlg out Making calls between 2 IMS clients(boghe) is possible. Would appreciate any help (I can share the cfg files if needed). Cheers, Serhat

2 3

Kamailio script config - Generate Documentation
by José Seabra 20 Oct '16

20 Oct '16

Hello there, I'm sending this email in order to know if anyone had work or has any idea how we can generate Documentation from kamailio script using Comments? This would be great because we could write comments explaining the Route method during script development and use it to generate documentation about our kamailio script, saving time and leave our deployment documented. Thank you -- Best Regards José Seabra

3 5

Regarding pua_reginfo, server_address
by Ram Anji 20 Oct '16

20 Oct '16

Hi All, I am configured kamailio to handle reg info event subscriptions, My server is running if i configure the following parameter. #kamailio local ip is 192.168.20.20 *modparam("pua_reginfo", "server_address", "sip:reginfo@192.168.20.20 <sip%3Areginfo(a)192.168.20.20>")* If i am not configured the above server address, i am getting bad configuration Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: auth [auth_mod.c:333]: mod_init(): auth: qop set, but nonce-count (nc_enabled) support disabled Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /etc/kamailio/permissions.allow Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: permissions [permissions.c:607]: mod_init(): default allow file (/etc/kamailio/permissions.al... rule set Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: permissions [parse_config.c:251]: parse_config_file(): file not found: /etc/kamailio/permissions.deny Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: permissions [permissions.c:616]: mod_init(): default deny file (/etc/kamailio/permissions.den... rule set *Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: ERROR: pua_reginfo [pua_reginfo.c:105]: mod_init(): server_address parameter not set* *Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: ERROR: <core> [sr_module.c:944]: init_mod(): Error while initializing module pua_reginfo (/usr/lib/...ginfo.so)* Oct 18 11:31:36 arm /usr/sbin/kamailio[2787]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Oct 18 11:31:36 arm kamailio[2771]: already running ... failed! Oct 18 11:31:36 arm kamailio[2771]: . Oct 18 11:31:36 arm systemd[1]: Started LSB: Start the Kamailio SIP proxy server. Hint: Some lines were ellipsized, use -l to show in full. I want to know what is the significance of this parameter and find no where this value is used. Thanks in advance. Regards, Raman

2 1

RTPProxy
by Rodrigo Moreira 20 Oct '16

20 Oct '16

Hi, What is difference between modules rtpproxy and rtpengine? Best regards. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

5 12

memory is not freed in system, while kamailio's statistic shows OK
by Vasiliy Ganchev 20 Oct '16

20 Oct '16

Hi folk! Have a production server - used as webrtc to sip proxy - it is built on kamailio v 4.4.2 - Server handles connections from ~4000 incoming SIP trunks (transport=TLS). Kamailio provides auth for this trunks. - Kamailio receives also connections from Webrtc app. Calls from Webrtc routed to one of the 4000 sip trunks. Server is running on Debian GNU/Linux 7 The number of calls on the system is small. Most of the load is generated by registration/re-registrations of SIP trunks The problem is following: - output of kamcmd core.shmmem and kamcmd pkg.stats shows constant using of memory - output of htop shows constantly increasing memory usage, e.g. after one week working: PID USER PRI NI VIRT RES SHR S CPU% MEM% TIME+ Command ..... 6013 kamailio 20 0 744M 134M 130M S 0.0 3.6 2:28.80 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 32 -u kamailio -g kamailio 6002 kamailio 20 0 744M 134M 130M S 0.0 3.6 2:30.93 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 32 -u kamailio -g kamailio 6012 kamailio 20 0 744M 134M 130M S 0.0 3.5 2:35.00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 32 -u kamailio -g kamailio 5991 kamailio 20 0 744M 133M 129M S 0.0 3.5 2:05.88 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 32 -u kamailio -g kamailio 6018 kamailio 20 0 744M 133M 129M S 0.0 3.5 2:31.90 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 512 -M 32 -u kamailio -g kamailio ..... Despite running with: PKG_MEMORY=32, Amount of used memory per process is higher. Finishes all with monit killing kamailio, because of memory usage more than allowed studied this tutorial: https://www.kamailio.org/wiki/tutorials/troubleshooting/memory but did not find a way how to dig up the problem's root. Tried also to reproduce the issue in lab environment (having clone of production server). Generated load with incoming registration using PJSIP, but with no success - memory usage is constant. kamailio -v version: kamailio 4.4.2 (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 with gcc 4.7.2 Any suggestion/idea how to troubleshoot deeper would be highly apprecieted Cheers! -- View this message in context: http://sip-router.1086192.n5.nabble.com/memory-is-not-freed-in-system-while… Sent from the Users mailing list archive at Nabble.com.

2 3

help with NOTIFY
by Slava Bendersky 19 Oct '16

19 Oct '16

Hello Everyone, I am trying forward NOTIFY to client. My setup is PBX Server ---- local lan ---kamailio ---- internet ---- client The problem with whole setup that kamailio is not forwarding NOTIFY to client. Here are log. U 2016/10/19 14:23:29.581609 10.18.130.50:5160 -> 10.18.130.46:5060 NOTIFY sip:10101@10.18.130.46:5060 SIP/2.0. Via: SIP/2.0/UDP 10.18.130.50:5160;rport;branch=z9hG4bKPjONocd4FvI5LkOmSR6ZCA5Z1f7SYDb3Ty. From: <sip:8101@networklab.ca>;tag=ziApds23FQsPL2zd6q8pCUlwW.jwa6mK. To: "Slava" <sip:10101@networklab.ca>;tag=734f813f. Contact: <sip:10.18.130.50:5160>. Call-ID: A0bJWg8v0nGmtUmiHzUixQ... CSeq: 18547 NOTIFY. Route: <sip:kamailio_local_ip:5060;lr;r2=on;ftag=734f813f>. Route: <sip:kamailio_public_ip:5081;transport=tls;lr;r2=on;ftag=734f813f>. Event: presence. Subscription-State: active;expires=413. Allow-Events: message-summary, presence, dialog, refer. Max-Forwards: 70. User-Agent: FPBX-13.0.188.9(14.0.2). Content-Type: application/pidf+xml. Content-Length: 579. . <?xml version="1.0" encoding="UTF-8"?> <presence entity="sip:8101@10.18.130.50:5160" xmlns="urn:ietf:params:xml:ns:pidf" xmlns:pp="urn:ietf:params:xml:ns:pidf:person" xmlns:es="urn:ietf:params:xml:ns:pidf:rpid:status:rpid-status" xmlns:ep="urn:ietf:params:xml:ns:pidf:rpid:rpid-person"> <note>On the phone</note> <tuple id="8101"> <status> <basic>closed</basic> </status> <contact priority="1">"Slava" <sip:10101@networklab.ca></contact> </tuple> <pp:person> <status> <ep:activities>ep:busy</ep:activities> </status> </pp:person> </presence> U 2016/10/19 14:23:32.045388 10.18.130.46:5060 -> 10.18.130.50:5160 SIP/2.0 408 Request Timeout. Via: SIP/2.0/UDP 10.18.130.50:5160;rport=5160;branch=z9hG4bKPju05rMNlzhfukLvaq4ELmq.QI4GBqDN-m;received=10.18.130.50. From: <sip:8101@networklab.ca>;tag=Yzy2m6O2AFT6rGSDBmo6PjcGaCWT1i5H. To: "Slava" <sip:10101@networklab.ca>;tag=ad5a945e. Call-ID: 2pq5iebmtLdB_OsMUs3EOg... CSeq: 14623 NOTIFY. Server: kamailio (4.3.4 (x86_64/linux)). Content-Length: 0. Any help thank you. volga629

1 0

Active calls limit
by Ivan Dudko 19 Oct '16

19 Oct '16

Hello! I am trying to implement limit of active calls for subscribers. And for each subscriber this limit must be personal. I find example route in presentation of Daniel-Constantin Mierla. And i need to set $xavp(caller=>active_calls) = 1; for each dialog. But i can't understand how to load this number of active calls from some-thing similar of user profile or subscriber table? route[DIALOG] { if (is_method("CANCEL") || (has_totag() && is_method("INVITE|BYE|ACK"))) { dlg_manage(); return; } if (is_method("INVITE") && !has_totag() && !isflagset(WITH_ACTIVE_CALLS_LIMIT)) { if( $xavp(caller[0]=>active_calls) != $null && $xavp(caller[0]=>active_calls) > 0 ) { if(!get_profile_size("caller", "$fU@$fd", "$var(acsize)")) { send_reply("500", "No more active calls"); exit; } if($var(acsize)>=$xavp(caller[0]=>active_calls)) { send_reply("403", "No more active calls"); exit; } set_dlg_profile("caller", "$fU@$fd"); } setflag(WITH_ACTIVE_CALLS_LIMIT); dlg_manage(); } }

2 5

Kamailio Bench
by Rodrigo Moreira 19 Oct '16

19 Oct '16

Hello, i need your help, I'm having problems with the reverse response of an INVITE passing through a rtpproxy (NAT). Does anyone have any idea what might be going on? UE is informed of the result INVITE, but the UAS does not receive the test bench detail. Greetings. -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

2 1

Problems with CDRs
by Diogenes Marcano 19 Oct '16

19 Oct '16

I am using kamailio with ubuntu, but I can't obtain CDRs. In my conf. file related to CDR I have: modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_start_id", "start_time") modparam("acc", "cdr_end_id", "end_time") modparam("acc", "cdr_duration_id", "duration") But I don't know where is the problem. Regards Diogenes Marcano +58 414 117 2011

2 1

BYE issue
by Nelson Migliaro 18 Oct '16

18 Oct '16

Hello everybody, I am having issues with one SIP vendor. I have a Kamailio in bridge mode (private IP / Public IP) and some Asterisk and Media Gateways. Calls get established and I have two way audio but when the remote party hangs up the call, the BYE arrives to the Kamailio and does not move forward. I think the problem is SIP vendor rewrite the BYE header and change the asterisk IP with the public IP of the kamailio. The IP that appears in the header of the BYE have to be the same that appears in the contact (UAC that send the call, in my case the Asterisk). Vendor should not change that IP. ¿Am I correct? Thank you ----------------------------------------------------------------------------------------------------- INVITE ---------------------------------------------------------------------------------------------------- 2016/10/17 18:50:49.110967 PUBLIC-KAMAILIO-IP:5060 -> VENDOR-IP:6060 INVITE sip:DESTINATION-NUMBER@VENDOR-IP:6060 SIP/2.0 Record-Route: <sip:PUBLIC-KAMAILIO-IP;r2=on;lr=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1MMDIudm9pY2U G9jYWw-;did=09b.9572;nat=yes> Record-Route: <sip:PRIVATE-KAMAILIO-IP;r2=on;lr=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1MMDIudm9pY2U G9jYWw-;did=09b.9572;nat=yes> Via: SIP/2.0/UDP PUBLIC-KAMAILIO-IP;branch=z9hG4bK06a.07540d0e2f32a811ecf9c0a5235dc77a.1 Via: SIP/2.0/UDP PRIVATE-ASTERISK-IP:5060;received=PRIVATE-ASTERISK-IP;branch=z9hG4bK6bb5a7b3;rport=5060 Max-Forwards: 69 From: "SOURCE-NUMBER" <sip:SOURCE-NUMBER@MY-COMPANY>;tag=as5e87b96c To: <sip:DESTINATION-NUMBER@VENDOR-IP> Contact: <sip:SOURCE-NUMBER@PRIVATE-ASTERISK-IP:5060> Call-ID: 025cc3717ba59faa000cf4db6f8be588@PRIVATE-ASTERISK-IP:5060 CSeq: 102 INVITE User-Agent: UAC Date: Mon, 17 Oct 2016 16:53:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 269 v=0 o=root 292850421 292850421 IN IP4 PUBLIC-KAMAILIO-IP s=Asterisk PBX c=IN IP4 PUBLIC-KAMAILIO-IP t=0 0 m=audio 23456 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv a=nortpproxy:yes ----------------------------------------------------------------------------------------------------- BYE ----------------------------------------------------------------------------------------------------- 2016/10/17 18:50:58.241666 VENDOR-IP:6060 -> PUBLIC-KAMAILIO-IP:5060 BYE sip:SOURCE-NUMBER@PUBLIC-KAMAILIO-IP:5060 SIP/2.0 Via: SIP/2.0/UDP VENDOR-IP:6060;branch=z9hG4bKeff4.48943e76.0 Via: SIP/2.0/UDP VENDOR-IP:5060;branch=z9hG4bK1d4e605e4ll19f74fBYE421ce8658050206 Max-Forwards: 34 Route: <sip:PUBLIC-KAMAILIO-IP;lr;r2=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1> Route: <sip:PRIVATE-KAMAILIO-IP;lr;r2=on;ftag=as5e87b96c;vsf=AAAAAAAAAAAAAAAAAABQUk9fRVYAU0UuODY-;vst=AAAAAAQEAw8MDgsAAHYAcVddXkZWRVVDVl1> To: "SOURCE-NUMBER"<sip:SOURCE-NUMBER@YO>;tag=as5e87b96c From: <sip:DESTINATION-NUMBER@PUBLIC-KAMAILIO-IP>;tag=421ce86-co1547-INS001 Call-ID: 025cc3717ba59faa000cf4db6f8be588@PRIVATE-ASTERISK-IP:5060 CSeq: 154701 BYE User-Agent: VENDOR Content-Length: 0 -----------------------------------------------------------------------------------------------------

3 5

420 Bad Extension error
by Infinicalls Infinicalls 18 Oct '16

18 Oct '16

Hi all, I am testing Edge Proxy setup with the example given here : http://kamailio.org/docs/modules/4.2.x/modules/outbound.html and I get this error : U 2016/10/06 09:18:48.331580 10.0.0.4:5060 -> 52.233.25.164:5060 SIP/2.0 420 Bad Extension. Via: SIP/2.0/UDP 10.1.0.4;branch=z9hG4bK162a.e9f73fe5e52ded1f30801aa00ee2a771.0; received=52.233.25.164. Via: SIP/2.0/UDP 10.114.70.81:1228;received=113.193.149.49;rport=1228;branch=z9h G4bKPj3ec7c87e09d6452b9e68e60052cb2651. From: "User1" <sip:user1@infinicalls-proxy1.myserver.com>; tag=ac1e36aec8bf445bbec8e43cf28e9f48. To: "User1" <sip:user1@infinicalls-proxy1.myserver.com>;ta g=b27e1a1d33761e85846fc98f5f3a7e58.b172. Call-ID: a270d285de6145c6b8670b909b104e74. CSeq: 20149 REGISTER. Contact: <sip:user1@10.114.70.81:1228;ob>;expires=300. Unsupported: path. Path: <sip:10.1.0.4;lr>. Supported: outbound. P-Registrar-Error: No support for found Path indicated. Server: kamailio (4.4.3 (x86_64/linux)). Content-Length: 0. Both my proxy and registrar are behind NAT. rtpproxy is installed and started. I already have modparam("registrar", "use_path", 1) in my registrar. Any idea what is missing? Thanks. regards Ganesh Kumar

2 17

Sip route can be as IMS PCSCF node
by Surender Singh 17 Oct '16

17 Oct '16

Hi, Can we use sip router as IMS PCSCF to connectivity with IMS. Does sip router supports the diameter RX interface to connectivity with PCRF . Regards Surender Singh ::DISCLAIMER:: ---------------------------------------------------------------------------------------------------------------------------------------------------- The contents of this e-mail and any attachment(s) are confidential and intended for the named recipient(s) only. E-mail transmission is not guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or may contain viruses in transmission. The e mail and its contents (with or without referred errors) shall therefore not attach any liability on the originator or HCL or its affiliates. Views or opinions, if any, presented in this email are solely those of the author and may not necessarily reflect the views or opinions of HCL or its affiliates. Any form of reproduction, dissemination, copying, disclosure, modification, distribution and / or publication of this message without the prior written consent of authorized representative of HCL is strictly prohibited. If you have received this email in error please delete it and notify the sender immediately. Before opening any email and/or attachments, please check them for viruses and other defects. ----------------------------------------------------------------------------------------------------------------------------------------------------

2 1

sipml5 register to kamailio IMS (Unauthorized - Challenging the UE)
by Serhat Guler 17 Oct '16

17 Oct '16

Hi, I am trying to register sipml5 webrtc client to my kamailio IMS setup. I have tried to register the client both with ws and wss, but it seems to be that the sipml5 doesn't calculate the authentication digest right. The authentication mechanism i set to AKAv1-MD5 as default in the hss. A simple wireshark file is attached. .10 being the host, .11 being the kamailio server. >From the output of scscf we can see that the digests do not match. [REGISTER] from [sip:bob@net1.test] to [sip:bob@net1.test] ims_auth [authorize.c:824]: authenticate(): uri=sip:net1.test nonce=rB2iDuerHwoy+LUStSOsYojAESfWmAAApFZ3XNB8FdA= response=61cbdbeb47c9880ededfca51c3801800 qop=auth-int nc=00000001 cnonce=2d4545cf4c935c8094c8b1da3d4a2976 hbody=d41d8cd98f00b204e9800998ecf8427e ims_auth [authorize.c:872]: authenticate(): UE said: 61cbdbeb47c9880ededfca51c3801800 and we expect 7e27ef414cf37d96cd1c849bb7e59415 ha1 a53988ba0b257941bc747b1026225c77 (REGISTER) tm [tm.c:1265]: w_t_reply(): ERROR: t_reply: cannot send a t_reply to a message for which no T-state has been established Thanks a lot, Serhat

2 2

How to use sql transformations?
by Daniel Tryba 17 Oct '16

17 Oct '16

How to use the sql transformation? https://www.kamailio.org/wiki/cookbooks/4.4.x/transformations#sql_transform… has the following example: xlog("$$rm = $rm = $(rm{s.sql})"); But adding this to the request_route and starting kamailio will fail: ERROR: pv [pv_trans.c:2351]: tr_parse_string(): unknown transformation: sql}/sql/3! ERROR: <core> [pvapi.c:1629]: tr_lookup(): error parsing [{s.sql}] ERROR: <core> [pvapi.c:1010]: pv_parse_spec2(): bad tr in pvar name "rm" ERROR: <core> [pvapi.c:1036]: pv_parse_spec2(): invalid parsing in [$(rm{s.sql})] at (4) ERROR: xlog [xlog.c:512]: xdbg_fixup_helper(): wrong format[$$rm = $rm = $(rm{s.sql})] ERROR: <core> [route.c:1154]: fix_actions(): fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:372 line 372 is the above xlog and sqlops.so is loaded (and works). Anybody got a working example of this? Or an other hint to prevent sql injections when using user supplied variables in sql queries?

1 0

503 Service unavailable to 500
by Dhruvin Desai 17 Oct '16

17 Oct '16

Hi 503 Service unavailable received at kamailio SIP server is forwarded as 500 server internal error to the UAC. Which seems fine based on the fact that there is no retry-after header. Would like to know if there is any way to to control the SIP Headers like adding Proprietary SIP Headers to the 500 "Server Interrnal Errror" created by kamalio stack for 503 service unavailable (without retry-after Header) It seems that Kamalio does not retain the proprietary sip headers received in 503 to 500 response forwarded. Any insight into the issue would be helpful -- Warm Regards Dhruvin Desai.

2 1

RTPEngine & Homer
by Grant Bagdasarian 17 Oct '16

17 Oct '16

Hello, I'm trying to get rtpengine to send its stats to Homer, but I'm unsure to which IP I should configure RTPengine to send the stats? We have a Kamailio server configured with the sipcapture module, which works perfectly for SIP traffic, but RTP stats are not visible in Homer. Do I need to point RTPengine to the sip capture server or directly to Homer? Running RTPEngine 4.5.2 and Kamailio 4.4.1. Regards, Grant

1 0

HOMER / HEP support in SEMS?
by Daniel Pocock 14 Oct '16

14 Oct '16

Does anybody know if SEMS has HOMER / HEP support like Kamailio, or if it is under development? It would be useful to extract RTCP stats from SRTP streams, the standalone HOMER captagent can't see inside them when they are encrypted (SRTCP) Regards, Daniel

2 1

Accounting CDR generation
by Ivan Dudko 14 Oct '16

14 Oct '16

Hello! I am setting this in kamailio.cfg #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "cdrs_table", "acc_cdrs") modparam("acc", "cdr_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_user=$rU;dst_ouser=$tU;dst_domain=$rd;sip_code=$rs;sip_reason=$rr;callid=$dlg(callid);dst_ip=$Ri") modparam("acc", "cdr_start_id", "start_time") modparam("acc", "cdr_end_id", "end_time") modparam("acc", "cdr_duration_id", "duration") #!endif But in CDRs table ('acc_cdrs') field dst_user=$rU is always empty if call is not completed with remote side answer. But in accounting table ('acc') this field isn't empty. Is this behavior is correct? I think it is not correct.

2 7

Websocket with kamailio IMS
by Serhat Guler 13 Oct '16

13 Oct '16

Dear all, I am trying to configure websockets for kamailio IMS. My basic kamailio(without IMS) setup works properly with websockets and without websockets. The IMS functionality also works, but only with IMS clients and now I want to use webrtc clients and make calls. At the moment I am trying to add the websockets support to the pcscf. I have did some changes to the pcscf cfg file. My problem is not really related to the cfg of pcscf as the component is listening for tcp/tls connections, but I cannot connect through the assigned ports. The output of the pcscf component and the the wireshark pcap file for the Websocket are attached. (I basically tried to browse http://server.ip.address:tcp.port or https://server.ip:tls.port without using a webrtc client yet). 192.168.0.10 -> host 192.168.0.11 -> virtual machine running kamailio Thanks in advance, Serhat

2 1

Reload PIPILIMIT table from DB
by Grant Bagdasarian 12 Oct '16

12 Oct '16

Hello, Is it possible to live reload the pipelimit module so it gets new values from the DB without a restart of Kamailio? I couldn't find a command to do this. Regards, Grant Bagdasarian CM

2 3

prefix_route and db_cluster not working
by Martin Nyström 12 Oct '16

12 Oct '16

Hi, I have an issue when I am unable to get prefix_route to work together with db_cluster. If I set the db_url manually it works great, pointing it to a specific database. However we need the redundancy for this module. Dispatcher uses the same cluster without any issues. The following are my configuration sets: # DB CLUSTER (db_cluster.inc) loadmodule "db_cluster.so" modparam("db_cluster", "connection", "KAMDB01=>mysql://kamailio:password123@192.168.55.10/kamailio") modparam("db_cluster", "connection", "KAMDB02=>mysql://kamailio:password123@192.168.55.11/kamailio") modparam("db_cluster", "cluster", "AccCluster=>KAMDB02=9s9s;KAMDB01=4s4s") modparam("db_cluster", "cluster", "Settings=>KAMDB02=9s9s;KAMDB01=4s4s") # KAMAILIO.CFG (kamailio.cfg) # PREFIX ROUTE MODPARAM loadmodule "prefix_route.so" include_file "db_cluster.inc" modparam("prefix_route", "db_url", "cluster://Settings") modparam("prefix_route", "db_table", "prefix_route") modparam("prefix_route", "exit", 1) Syslog output: Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: ERROR: <core> [db_cmd.c:122]: db_cmd(): DB driver cluster does not implement mandatory db_cmd function Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: ERROR: <core> [db_cmd.c:168]: db_cmd(): db_cmd: Cannot create db_cmd structure Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: ERROR: prefix_route [prefix_route.c:126]: pr_db_load(): prefix_route: db_load(): db_cmd() failed Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: : prefix_route [prefix_route.c:214]: mod_init(): prefix_route: db_load() failed Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: ERROR: <core> [sr_module.c:974]: init_mod(): Error while initializing module prefix_route (usr/local/lib64/kamailio/modules/prefix_route.so) Oct 10 09:53:03 kamailio01 /usr/local/sbin/kamailio[5225]: INFO: <core> [sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Ubuntu 14.04LTS version: kamailio 4.3.6 (x86_64/linux) 105a17 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, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 105a17 compiled on 10:06:12 Oct 4 2016 with gcc 4.8.4 Thank you.

2 3

Enabling accounting module
by Diogenes Marcano 12 Oct '16

12 Oct '16

Can I enable accounting modules acc and acc_cdrs with out diameter? Best regard

2 1

Permissions module question
by Ivan Dudko 11 Oct '16

11 Oct '16

Hello! I am new to kamailio and trying to use it vanilla config. Now main question is how to use ip based auth. I found recent post of Daniel-Constantin Mierla: http://lists.sip-router.org/pipermail/sr-users/2011-December/071147.html Here he recommends to use 'address' table from permissions module, I try yo use advice and add this lines at config begin: #!define WITH_MYSQL #!define DBURL "mysql://kamailio:kamailiorw@localhost/kamailio" #!define WITH_AUTH #!define WITH_IPAUTH #!define WITH_USRLOCDB Database is created and kamailio can access it. I am add user 1000 to kamailio via kamctl and successful register it with soft-phone. I create trunk without registration to kamailio on asterisk server. And trying to call from asterisk to user 1000. Call is successful. I try to create file /etc/kamailio/permissions.deny with content 'ALL : ALL'. And retry previous call. It still sucessful. I try to add record with asterisk address to 'address' table with group 1. And retry previous call. It still sucessful. I am confused. I do not now how to disable any address for ip_auth except if it in the 'address' table. And allow any address with if it request kamailio with registration. -- Best Regards, Ivan Dudko

3 4

Kamailio - DMQ dns srv Question
by José Seabra 11 Oct '16

11 Oct '16

Hello, I have a doubt related with DMQ dns behavior, I noticed that when kamailio starts, it tries to resolve DMQ name configured on parameter notification_address as the following sequence: 1. SRV 2. A 3. AAAA Isn't supposed kamailio try first resolve the NAPTR DMQ name, and then SRV? I'm asking this because kamailio is trying resolve the SRV record without any transport protocol specified on query, as my dns server only accepts queries on the format "_Service._Proto.Name" the SRV will never be resolved. Thank you. BR -- José Seabra

4 13

transaction is not suspended ERROR
by david 11 Oct '16

11 Oct '16

Hello all i'm having some errors sometimes and i'm not able to know why they happen i made a lua configuration script where i create some ESL sockets with several freeswitches (10) (around 500-600 sockets in total, depending on the number os children (8) and listen ips we have (5)), and i use the lua script to send some API to the FSW by a function. No matter what the FSW replies. i dont know why, but i think that having that conn:bgapi($command) in the lua script, makes the issue to happen. theorically, the lua script waits to receive the response from the fsw through the tcp socket, and after that the script continues (with an async_sleep command), but after the API sent, i see (only sometimes, few times) an async_sleep seem to fail when doing the t_continue with the following error WARNING: tm [t_suspend.c:186]: t_continue(): transaction is not suspended [20748:225229907] could it be posible that having manye ESL sockets opened in the kamailio server makes this to happen? how many async workers are needed? i have 8 right now is there a way where i can know or correlate the log to a $ci or something known? best regards david

2 3

Kamailio on Redhat 7.2
by kamailio noob 11 Oct '16

11 Oct '16

Hi All, Can anyone please provide me the details that how to install kamailio on Redhat 7.2. I am following the below url guide for installation of Kamailio. https://www.kamailio.org/wiki/install/4.4.x/git when i compile kamailio code I am not able to find certain scripts in /usr/local/sbin Can anyone help regarding this ? Regards, Atif Ramzan.

2 1

Busy return code when a user has an ongoing transaction
by Gholamreza Sabery 10 Oct '16

10 Oct '16

I want to be able to return a Busy code foe a user that is talking to someone else and receives an INVITE for a new call. How can I implement this in Kamailio? Regards

2 2

Order of record-route
by Dirk Teurlings - Signet B.V. 10 Oct '16

10 Oct '16

Hi, Looking at an issue with SIP-Cisco and a Barracuda firewall, I've come across something I need to know more about. The setup is as follows: CLIENT -> KAMAILIO -> ASTERISK CLIENT <- KAMAILIO <- ASTERISK Client sends INVITE, Kamailio forwards to Asterisk, SIP 100 and SIP 183 are being sent and finally a 200 OK. We now expect the Client to send back an ACK packet, but that never reaches us. In this case the SIP 183 is the first packet that contains the full record route of the path, the 200 OK contain the same record-routes being: Record-Route: <sip:192.168.0.200;r2=on;lr;ftag=12D1120C-1C4;did=4c8.9203;nat=yes> Record-Route: <sip:1.2.3.4;r2=on;lr;ftag=12D1120C-1C4;did=4c8.9203;nat=yes> Record-Route: <sip:4.3.2.1;r2=on;lr;did=4c8.3f471377> Record-Route: <sip:10.0.0.101;r2=on;lr;did=4c8.3f471377> 192.168.0.200 is the private address of Kamailio 1.2.3.4 is the public address of Kamailio 4.3.2.1 is the public address of the Client 10.0.0.101 is some private addres of the Client What we're seeing is that the Client is sending his ACK message to our 192.169.200 private address, which he will never reach. So me question in this case is; is the order of the Record-route headers relevant and do I have any say in which order Kamailio outputs these headers? To be clear, the Kamailio config only contains one reference to record_route which is caught for INVITE messages, which only fires when all things like NAT and transaction checks haven't already picked it up. Kind regards, Dirk

3 5

kamctl - Error opening FIFO
by Hermann Norpois 10 Oct '16

10 Oct '16

Hello, I have problems with the behaviour of kamctl ubuntu@ip-172-31-32-10:/etc/kamailio$ kamctl ul show ERROR: Error opening Kamailio's FIFO /var/run/kamailio/kamailio_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo")' in your config ERROR: and also have loaded the mi_fifo module. #my FIFO file ubuntu@ip-172-31-32-10:/etc/kamailio$ ls -l /var/run/kamailio/kamailio_fifo prw-rw---- 1 kamailio kamailio 0 Oct 7 09:01 /var/run/kamailio/kamailio_fifo #FIFO in config ubuntu@ip-172-31-32-10:/etc/kamailio$ grep fifo kamailio.cfg loadmodule "mi_fifo.so" # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/var/run/kamailio/kamailio_fifo") #With sudo it works ubuntu@ip-172-31-32-10:/etc/kamailio$ sudo kamctl ul show Domain:: location table=1024 records=0 max_slot=0 Further ... ubuntu@ip-172-31-32-10:/etc/kamailio$ kamctl restart INFO: Stopping Kamailio : ERROR: No PID file found (/var/run/kamailio.pid)! Kamailio probably not running INFO: check with 'ps axw | /bin/egrep kamailio' #But kamailio is running ubuntu@ip-172-31-32-10:/etc/kamailio$ ps axw|egrep kamailio 4122 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4123 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4124 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4125 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4126 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4127 ? S 0:46 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4128 ? S 0:44 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4129 ? S 0:44 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4130 ? S 0:44 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4131 ? S 0:03 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4132 ? S 909:27 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4133 ? S 0:00 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4134 ? S 5:08 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4135 ? S 25:51 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4136 ? S 26:02 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4137 ? S 26:08 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4138 ? S 26:18 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 4139 ? S 10:17 /usr/sbin/kamailio -f /etc/kamailio/kamailio.cfg -P /var/run/kamailio/kamailio.pid -m 64 -M 8 -u kamailio -g kamailio 27344 pts/1 S+ 0:00 egrep --color=auto kamailio Whats wrong with my usage of kamctl? Thanks Hermann

3 2

Q: (markup) language for defining structures and function prototypes
by Daniel-Constantin Mierla 10 Oct '16

10 Oct '16

Hello, writing to see if anyone is aware of any (well known) markup language for defining structures and function prototypes. The goal is to use it for specifying the functions exported by KEMI (https://www.kamailio.org/wiki/devel/config-engines) so we can generate C code for the exported structure, but more important, to get at least the index with the functions available over the KEMI. I feel it's going to be a big burden to write C code and docs in two places, even worse, to keep both of them sync'ed over the time. At this moment, there are many functions exported via KEMI, but not documented anywhere (apart of the C code). It should allow defining the name of the function, the parameters with their associated types, return code and eventually the description for each of them. At the end we could invent one based on xml format (or something similar), but I would better use something that already exists there. I searched quickly, but couldn't find anything straightaway. A hint could be a format that is used to document the RPC commands. Once the format is decided, we may have to write a tool to generate C code to be compiled with the module and the xml docbook file to be included in the documentation. Any idea to solve this welcome! Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com

1 0

Kamailio stop processing SIP
by Igor Potjevlesch 07 Oct '16

07 Oct '16

Hello, Someone has an idea? Again, Kamailio stop processing SIP during more than 30s. Nothing print in the logs. The processing back on when a script detect no answer and restart the service. Regards, Igor. De : Igor Potjevlesch [mailto:igor.potjevlesch@gmail.com] Envoyé : vendredi 23 septembre 2016 11:05 À : sr-users(a)lists.sip-router.org Objet : Dimension number of child Hello, I'm wondering if there's a best practise to dimension the number of child to setup? I see, very often, Kamailio that stops to reply to SIP traffic for couple a seconds. This cause some calls drop if RE-INVITE are not processed for example. So, I'm wondering if the issue is not due to a lack of child for processing the SIP traffic. Thank you for your inputs. Regards, Igor.

3 6

Migrating from OpenSER config file question.
by Brian McCrary 07 Oct '16

07 Oct '16

Hello group, I know this may be hard to believe but I'm in the process of upgrading an old, but stable, OpenSER 1.0 group of servers to Kamilio. I'm basically going to sort of start from scratch with the database and export my old MySQL database and write some scripts to reimport the data using kamctl scripts since the database structure has changed substantially since then it looks like. I've mostly migrated the config file over fairly easily, except for AVPops. I've got a section of config file that writes into the usr_preferences database for call forwarding, but I seem to get the same error anywhere I try to write to the database using AVPops. Here's the output I get when checking the config: 0(1378) DEBUG: <core> [pvapi.c:268]: pv_cache_add(): PV cache not initialized, doing it now 0(1378) ERROR: <core> [pvapi.c:828]: pv_parse_spec2(): error searching pvar "avp" 0(1378) ERROR: <core> [pvapi.c:1032]: pv_parse_spec2(): wrong char [s/115] in [$avp(s:callfwd)] at [5 (5)] 0(1378) : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/kamailio.cfg, line 145, column 4-18: Can't get from cache: $avp(s:callfwd) ERROR: bad config file (1 errors) Here's what I think are the revelant portions of the config file: modparam("avpops", "db_url", "mysql://user:pass@localhost/kamailio") modparam("avpops", "avp_table", "usr_preferences") modparam("avpops", "uuid_column", "uuid") modparam("avpops", "username_column", "username") modparam("avpops", "domain_column", "domain") modparam("avpops", "attribute_column", "attribute") modparam("avpops", "value_column", "value") modparam("avpops", "type_column", "type") <snip> if (avp_db_load("$from/username", "$avp(s:callfwd)")) { if(!avp_db_delete("$from/username", "$avp(s:callfwd)/usr_preferences")) { sl_send_reply("500", "Server DB error"); } } # avp_write("$ruri","$avp(s:callfwd)"); # Old OpenSER 1.0 syntax... $avp(s:callfwd) = $ruri; # New Kamailio syntax ?? avp_db_store("$from/username","$avp(s:callfwd)/usr_preferences"); t_relay(); return; Basically I just want to write the ruri into the usr_preferences database with an attribute of callfwd but not sure what I'm missing. I see the error "Can't get from cache" but not sure how that pertains to what I'm trying to do. Anyone have any ideas? Thanks, Brian

2 5

Size of passwords fields in db
by Daniel-Constantin Mierla 06 Oct '16

06 Oct '16

Hello, writing here to decide on a topic opened by pull request 779: - https://github.com/kamailio/kamailio/pull/779 what would be a fair size for db column storing a password that one would like to have for proper security? I would like to make it consistent over all tables that have a password column by defining a xml entity for the size of these columns. The pull request suggests 64 chars, has anyone other opinions on making it larger or smaller? If they are defined varchar, then should not be a problem of allocated size, so we can go with 128 if someone feels it worth doing larger now so we don't have to change it again in the near future. This change is about db schema, the modules I expect to work with allocated strings (or have length checks) in this case and should not be affected. Cheers, Daniel -- Daniel-Constantin Mierla http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Oct 24-26, 2016 - http://www.asipto.com

2 4

Fwd: TOPOH Module in Kamialio
by Ginti Saurabh 06 Oct '16

06 Oct '16

Hi, Can you let me know what is TOPOH module do in kamailio? what does it do? and if commented out this line then will it effect the system?

2 6

Load balancing with Kamailio
by Nitesh Bansal 06 Oct '16

06 Oct '16

Hello, I'm planning on doing some smart load balancing with Kamailio. We have a distributed network, with multiple Kamailio boxes in different locations serving as Ingress SBC, these Kamailio boxes are the entry point for a SIP call and then they route the call to a pre-configured Asterisk boxes. I want to move away from this, I would like these Kamailios to be able to distribute the traffic to Asterisk boxes based on the actual load on these boxes, the goal is to be more dynamic? Is there any Kamailio module which could do that? Do I need to integrate some other tool(Homer etc) with Kamailio to achieve this? Any suggestions are welcome. -- *Nitesh Bansal* Business Development Engineer http://www.voxbone.com/

2 1

Serial Forking and t_suspend
by Kelvin Chua 06 Oct '16

06 Oct '16

we have serial forking already setup properly but need to t_suspend(). found out that it completely breaks the destination set. other than manually rebuilding the destination set after t_continue(), are there any other solution out there? Kelvin Chua

2 1

Re: [SR-Users] Identical Kamailio setup on 2 hosts
by Daniel Tryba 05 Oct '16

05 Oct '16

On Fri, Sep 30, 2016 at 10:29:16PM +0530, Infinicalls Infinicalls wrote: > Hi Daniel, > This seems to be ideal setup for my requirement. Would you mind > sharing the kamailio.cfg file with me (with all the important data > masked) ? Not much important stuff to censor. LBIPADDR is the ipadress of the machine itself. REGISTRARNET/30 is the net of the backend machines. The dispatcher.list contains the backend registrars The backend is essentially the default config (with an always on rtpengine) with the following additions: You need to enable path.so and set modparam("path", "use_received", 1) Tell the registrar module to use path (path_mode 1 fixes you 420 problem): modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 1) modparam("registrar", "path_use_received", 1) If you want to send keepalives from the registrar to the enduser: modparam("nathelper", "ping_nated_only", 0) modparam("nathelper", "udpping_from_path", 1) the loadbalancer config has a little blurb about relaying the sipping from the backend manually using the path header. But I'd advice to ping from the an external process to have better control (eg sipsak)

2 3

Addressing a distinct PSI result is: 555 AS Contacting Failed - iFC terminated dialog
by Valentin Christoph 05 Oct '16

05 Oct '16

Hi all, I'm trying to send a call to an AS via a kamailio S-CSCF (release 4.4.3). I'm addressing the AS by a distinct PSI: sip:04325432101@domain.net;user=gsmr. At the first try the S-CSCF sends a SAR to HSS and receives a successful SAA (tshark traces attached), but the S-CSCF rejects the call with 555 AS Contacting Failed - iFC terminated dialog. At the second try (a few seconds later), the S-CSCF doesn't exchange SAR/SAA with the HSS, and now the call is successfully forwarded to the AS. This problem is permanent, it is not the result of a temporary loss of connectivity between S-CSCF and AS. I checked the iFCs that I had configured at the HSS and I think, they are OK. I attached - tshark traces from the node, where the S-CSCF resides: 2016_09_19_error_555_17_scscf_filtered.pcap - WITH_DEBUG logs from kamailio: 2016_09_19_error_555_17_kamailio_1st_try.log - A tarball of /usr/local/etc/kamailio: 2016_09_19_error_555_17_kamailio.tar.gz Can anybody help? Is it a bug in kamailio or is the problem on my side? 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.

2 4

484 Address Incomplete error
by Infinicalls Infinicalls 05 Oct '16

05 Oct '16

Hi all, I am trying a simple proxy (with dispatcher) and a registrar. However, I have this issue U 2016/10/05 08:22:44.057195 10.0.0.4:5060 -> 52.233.25.164:5060 SIP/2.0 484 Address Incomplete. Via: SIP/2.0/UDP 10.1.0.4;branch=z9hG4bK8653.a9c65320000000000000000000000000.0;received=52.233.25.164. To: <sip:51.140.187.167:5060>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b79c. From: <sip:calls@infinicalls.com>;tag=533cb9e91f4b999cf76861cbb9ed54ed-d3d8. CSeq: 10 OPTIONS. Call-ID: 28fcdbee608363cf-5442(a)127.0.0.1. Server: kamailio (4.4.3 (x86_64/linux)). Content-Length: 0. I have set modparam("dispatcher", "flags", 2) Any idea? Thanks. regards Ganesh Kumar

2 1

NatHelper - Received adding on Contact header
by Yasin CANER 04 Oct '16

04 Oct '16

3 4

Sip Proxy server_kamailio Features
by Surender Singh 03 Oct '16

03 Oct '16

Hello, Thanks to all to develop the such type Sip Proxy server. Please suggest whether SIP proxy server is open source ,If yes than it can fulfil the my below requirement . 1) I having some sip clients which are registered on IMS .Can I use the kamailio Sip proxy sever between sip client and IMS. Sip client ------------------Kamailio Sip Proxy ---------------IMS 2) Second things, can I use Kamailio Sip Proxy to interconnect two IMS nodes . Can I create the SIP trunk without registration mode (username and password) between IMS nodes using Kamailio. Because the my IMS node doesn't support sip trunk in registration mode. Its support one normal trunk (SIP-T) My current IMS set up like below. IMS-1 (IBCF)---------------IMS2(IBCF). Regards Surender Singh ::DISCLAIMER:: ---------------------------------------------------------------------------------------------------------------------------------------------------- The contents of this e-mail and any attachment(s) are confidential and intended for the named recipient(s) only. E-mail transmission is not guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or may contain viruses in transmission. The e mail and its contents (with or without referred errors) shall therefore not attach any liability on the originator or HCL or its affiliates. Views or opinions, if any, presented in this email are solely those of the author and may not necessarily reflect the views or opinions of HCL or its affiliates. Any form of reproduction, dissemination, copying, disclosure, modification, distribution and / or publication of this message without the prior written consent of authorized representative of HCL is strictly prohibited. If you have received this email in error please delete it and notify the sender immediately. Before opening any email and/or attachments, please check them for viruses and other defects. ----------------------------------------------------------------------------------------------------------------------------------------------------

2 1

Kamailio number of concurrent calls
by Gholamreza Sabery 03 Oct '16

03 Oct '16

I used SIPp to stress test different scenarios in Kamailio, but how can I simulate a real call with media? SIPp has media sending capabilities but in not enough for example to simulate 1000 calls persecond! How can I test this?

5 7

How to check in Kamailio if a current sip message 200 OK is a response to the BYE.
by AFRICALLSHOP 01 Oct '16

01 Oct '16

Hello, I try to check in Kamailio if a current sip message "200 OK" is a response to a "BYE" request. Could you confirm the following code please? IF (IS_METHOD("BYE") && STATUS == 200) { } Regards Abdoul.

2 3

Proxy unable to connect - 420 Bad Extension error
by Infinicalls Infinicalls 01 Oct '16

01 Oct '16

Hi, I have setup a proxy and able to connect to the registrar. However, after a few attempts, it throws the "420:Bad Extension" error, saying "Unsupported: path. Path: <sip:10.1.0.4;lr>. Supported: outbound. P-Registrar-Error: No support for found Path indicated." Any idea, how to overcome this ? 10.1.0.4 is not my IP anywhere. I don;t know where it takes that value from. Thanks. regards Ganesh Kumar

2 2

Call forking priority
by Igor Potjevlesch 30 Sep '16

30 Sep '16

Hello list, I have a proxy that manage fix calls and a proxy that manage mobile calls. I want to implement call forking so when I call to a fix phone the call goes to the fix proxy and it forks to the mobile proxy who manage the call and viceversa. I want the fix to ring 2 or 3 times before the mobile. In order to do so I relay first to the fixed proxy, which will bounce back to mobile proxy the branch for mobile device. But now if I want to be able to choose which phone (mobile or fixe) rings before I have tried by doing the "async_route("RELAY", "7");" in the fixe proxy like in the mobile proxy but it doesn't work. My code is: Fix proxy: if($rU==123456789){ add_diversion("forking"); $var(priority)=1; append_branch("sip:987654321@proxy_mobile"); } route[INVITE]{ if($var(priority) == "1"){ async_route("RELAY", "7"); }else{ route(RELAY); exit; } Mobile proxy: if($tU==987654321){ $rU=123456789; } route[INVITE]{ if($dir == "forking"){ async_route("RELAY", "7"); }else{ route(RELAY); exit; } } Thanks.

2 2

ims_charging state machine issue
by 4fba81+8i7felu07050＠guerrillamail.com 30 Sep '16

30 Sep '16

Wanted to ask about changing rates while a call is not yet connected. I did this, $var(cc_ret1) = Ro_CCR_Stop("term", "487", "Request Terminated"); $var(cc_ret2) = Ro_CCR("RO_REPLY", "term", 30, "1", "1"); however, upon sending the new CCR, i get this ERROR: cdp [acctstatemachine.c:185]: cc_acc_client_stateful_sm_process(): Received unknown event [1] in state [1] and CRITICAL: ims_charging [ro_timer.c:122]: insert_ro_timer(): Trying to insert a bogus ro tl=0x7f484ab48728 tl->next=0x7f484ab42858 tl->prev=0x7f484aaa1090 looks like there is some statemachine issue. question is, how do we change rate? ---- Sent using GuerrillaMail.com Block or report abuse: https://www.guerrillamail.com/abuse/?a=UlM5EB4FUac1hAen5G1RYR3XANifx8NV1epN…

1 0

Identical Kamailio setup on 2 hosts
by Infinicalls Infinicalls 30 Sep '16

30 Sep '16

Hi, I have two identical hosts running Kamailio with the same set of users. They are located in different locations. Both are running behind NAT and I've enabled rtpproxy for that and also advertised the public IP. Both the MySQL DBs are similar and have the same data. Both the hosts have different sub-domains in alias and I have set auto_alias=no Now, the problem is my users are not able to communicate from one server to another. user1(a)mydomain.com from host1 is not able to communicate to user1(a)mydomain.com at host2, though inter-server communication is happening nicely. Any idea how to solve this? Thanks. regards Ganesh Kumar -- --- http://www.infinicalls.com

3 8

Load balancing traffic with Kamailio
by Nitesh Bansal 29 Sep '16

29 Sep '16

Hello, I'm planning on doing some smart load balancing with Kamailio. We have a distributed network, with multiple Kamailio boxes in different locations serving as Ingress SBC, these Kamailio boxes are the entry point for a SIP call and then they route the call to a pre-configured Asterisk boxes. I want to move away from this, I would like these Kamailios to be able to distribute the traffic to Asterisk boxes based on the performance of the Asterisk ,it would be great if we could take a look at SIP response time from Asterisk and use that as metric for load balancing the traffic) Is there any Kamailio module which could do that? Do I need to integrate some other tool with Kamailio to achieve this? Any suggestions are welcome. -- *Nitesh Bansal* Business Development Engineer http://www.voxbone.com/

3 3

Re: [SR-Users] Distributed Message Queue Module
by Charles Chance 29 Sep '16

29 Sep '16

Hello, That depends entirely on what you need to replicate between servers. DMQ is just a communication channel. How you use it is up to you. For example, if you enable dmq_usrloc, there's no need to share a location database. Htable has an option for replicating over dmq too. The number of direct module integrations like dmq_usrloc/htable is fairly limited, but you can also share arbitrary data between nodes yourself from within your config. I think you'll need to provide a little more information about what you're trying to achieve, however, in order for us to offer any further advice. Cheers, Charles On 29 Sep 2016 08:08, "Moacir Ferreira" <moacirferreira(a)hotmail.com> wrote: Hi, When using the DMQueue module what is the best approach, using one SQL server/database for each member node or having a single SQL server for all nodes? Thanks, Moacir _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Sipcentric Ltd. Company registered in England & Wales no. 7365592. Registered office: Faraday Wharf, Innovation Birmingham Campus, Holt Street, Birmingham Science Park, Birmingham B7 4BB.

1 0

Distributed Message Queue Module
by Moacir Ferreira 28 Sep '16

28 Sep '16

Hi, When using the DMQueue module what is the best approach, using one SQL server/database for each member node or having a single SQL server for all nodes? Thanks, Moacir

1 0

memory problem
by beer Ll 28 Sep '16

28 Sep '16

hi all I have kamailio 4.2.0 on Debian Jessie the parameters on startup of kamailio are : SHM_MEMORY=512 PKG_MEMORY=32 the server is dedicated to the kamailio SIP service with authentication and accounting on a separated db server the server process about 30/50 call setups per second everythings works good but after 5/6 days the processes lock and I received a lot of error log like these: <core> [rvalue.c:1710]: rval_str_add2(): rv eval out of memory <core> [msg_translator.c:362]: received_builder(): out of memory <core> [msg_translator.c:2717]: via_builder(): out of memory ... <core> [msg_translator.c:2423]: build_res_buf_from_sip_req(): out of memory; needs 514 ... <core> [data_lump.c:172]: insert_new_lump_before(): out of memory acc [acc_extra.c:267]: extra2strar(): extra2strar: out of memory. after a kamailio reboot evrethings works again for 5/6 days. what can I do to resolve?

3 3

RPC tcp limit buffer
by pars3c 28 Sep '16

28 Sep '16

Hello, i'm trying to list the dialog (dlg_profile list) in the proxy using an RPC call to jsonrpc-s (dlg_profile list). But when there are many calls , kamailio send an error because the tcp buffer is not big enough. The only solution is to increase the parameter tcp_conn_wq_max every time i've an error ? Thanks

2 2

Accessing cached database table
by Daniel W. Graham 28 Sep '16

28 Sep '16

Is there a way to access a cached database table from the config file, without using a function from a module? Example - Lookup ip_addr from 'address' table or src_ip from 'trusted' table based on the 'tag' column. No function exists in permissions module to do this, but running 'kamctl address show ' or 'kamctl trusted show ', show that the tables along with tag column are cached. Thanks! -Dan

2 2

invalid sendtoparameters
by Linux Vince 27 Sep '16

27 Sep '16

I am getting this error on every outgoing request. Below are log entries i found Sep 27 22:09:06 centos1 /usr/sbin/kamailio[22421]: ERROR: <core> [udp_server.c:550]: udp_send(): sendto(sock,0x7fb76f5da5e0,1568,0, 10.3.4.2:5060,16): Invalid argument(22) Sep 27 22:09:06 centos1 /usr/sbin/kamailio[22421]: CRITICAL: <core> [udp_server.c:555]: udp_send(): invalid sendtoparameters#012one possible reason is the server is bound to localhost and#012attempts to send to the net Sep 27 22:09:06 centos1 /usr/sbin/kamailio[22421]: ERROR: <core> [forward.h:201]: msg_send_buffer(): udp_send failed SNDTO values are SNDTO ,10.3.4.2 PORT 5060 I have confirmed the PID and it is bound to my public IP 22417 main process - attendant 22418 udp receiver child=0 sock=10.3.4.1:5060 22419 udp receiver child=1 sock=10.3.4.1:5060 22420 udp receiver child=2 sock=10.3.4.1:5060 22421 udp receiver child=3 sock=10.3.4.1:5060 22422 udp receiver child=4 sock=10.3.4.1:5060 22423 udp receiver child=5 sock=10.3.4.1:5060 22424 udp receiver child=6 sock=10.3.4.1:5060 22425 udp receiver child=7 sock=10.3.4.1:5060 22426 udp receiver child=0 sock=127.0.0.1:5060 22427 udp receiver child=1 sock=127.0.0.1:5060 22428 udp receiver child=2 sock=127.0.0.1:5060 22429 udp receiver child=3 sock=127.0.0.1:5060 22430 udp receiver child=4 sock=127.0.0.1:5060 22431 udp receiver child=5 sock=127.0.0.1:5060 22432 udp receiver child=6 sock=127.0.0.1:5060 22433 udp receiver child=7 sock=127.0.0.1:5060 22434 slow timer 22435 timer 22436 secondary timer 22437 ctl handler 22438 MI FIFO 22439 WEBSOCKET KEEPALIVE 22440 tcp receiver (generic) child=0 22441 tcp receiver (generic) child=1 22442 tcp receiver (generic) child=2 22443 tcp receiver (generic) child=3 22444 tcp receiver (generic) child=4 22445 tcp receiver (generic) child=5 22446 tcp receiver (generic) child=6 22447 tcp receiver (generic) child=7 22448 tcp main process How do i identify the issue ?

1 1

SIREMIS MANUAL
by Carlyle Edwards 27 Sep '16

27 Sep '16

Hi, is there a manual for SIREMIS?

2 1

Kamailio IMS NAT issue: PRACK/ACK RURI changing in P-CSCF
by Owais 26 Sep '16

26 Sep '16

Hi all, My scenario is as follows: I have All the CSCFs on the same IP using different ports. P-CSCF advertises a public IP. Rest of the nodes including I-CSCF and S-CSCF listen local only. Behavior: When an ACK or PRACK goes from P-CSCF to S-CSCF, the RURI changes such that alias parameter is removed. Hence, P-CSCF ends up sending PRACK and ACK on private IP of UE. Here's what the config looks like: Any hints? -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-IMS-NAT-issue-PRACK-ACK-RU… Sent from the Users mailing list archive at Nabble.com.

2 2

REGISTER between Kamailio 4.4 and Asterisk 13
by Grant Bagdasarian 26 Sep '16

26 Sep '16

Hello, We're trying to get our Asterisk test server to REGISTER with Kamailio. The first attempts are ok, but once the nonce timer is exceeded Kamailio starts rejecting the REGISTER requests from Asterisk, due to the nonce being expired. The auth_check function is the one returning error -4. Is this an issue in Asterisk or Kamailio? Which component is responsible for generating a new nonce value? I'm a bit confused regarding the flow of SIP REGISTER. I hope someone could help me out on this. Thanks! Regards, Grant Bagdasarian

2 2

Config changes needed from 4.4.2 to 4.4.3 (and a bug)
by Sebastian Damm 26 Sep '16

26 Sep '16

Hi, Kamailio release 4.4.3 was tagged as a "bug fix release", no config changes should be needed. However, we had to change two things in our config file to get 4.4.3 running. I thought, I might share those on the list. 1) Defines must not have '-' in the names. In 4.4.2 we had a Define like this: #!define NO-FRAUDCHECK This threw an error when trying to restart with 4.4.3. Replacing it with #!define NO_FRAUDCHECK fixed the problem. 2) Module Order problems. We introduced the usage of http_async_client module a few weeks ago, and this worked perfectly. However, 4.4.3 complained that the pv module was not available before loading http_async_client. After moving up the pv module above http_async_client, everything worked again. 3) There's a bug in uac_auth function. I've opened an issue for that: https://github.com/kamailio/kamailio/issues/797 Best Regards, Sebastian

2 1

Textops replace question
by Igor Olhovskiy 26 Sep '16

26 Sep '16

Hi! I’m trying to use replace function, but got strange results. Idea - modify Contact header What I got as start packet. (202 Accepted on onreply_route) … Contact: <sip:*97@10.0.20.81:5060> … I give a command replace("Contact: <.*>», "Contact: <" + $fu + ">"); $fu = sip:11@master.rufan.at And, as a result I got Contact: <sip:11@master.rufan.at>sip:*97@10.0.20.81:5060 Why it’s adding old Contact part at the end? What can cause this? On onreply_route I have only this command. Thanks! -- Best regards, Igor

3 4

Dialog sharing between Kamailio servers
by Michael Jepson 26 Sep '16

26 Sep '16

Hi all, I am trying to limit the concurrent calls of a customer, but we have multiple Kamailio servers in our platform. I have the dialog module up and running and it is storing its data in a database, but when I use get_profile_size, it will only return the number of dialogs in that profile for the current machine, it does not add the others. How can I get Kamailio to add the others, i.e. how can I get Kamailio to consult the database to get the correct size. Now, Kamailio seems to only write to the database, but not read from it. Best regards, MICHAEL JEPSON

2 1

Load balancing with Kamailio
by NITESH BANSAL 26 Sep '16

26 Sep '16

Hello, I'm planning on doing some smart load balancing with Kamailio. We have a distributed network, with multiple Kamailio boxes in different locations serving as Ingress SBC, these Kamailio boxes are the entry point for a SIP call and then they route the call to a pre-configured Asterisk boxes. I want to move away from this, I would like these Kamailios to be able to distribute the traffic to Asterisk boxes based on the actual load on these boxes, the goal is to be more dynamic? Is there any Kamailio module which could do that? Do I need to integrate some other tool(Homer etc) with Kamailio to achieve this? Any suggestions are welcome.

3 3

LCR routing, and finaly route to media gateway if not found
by Loic Chabert 26 Sep '16

26 Sep '16

Hello List, I'm finishing my implementation of LCR with kamailio for outgoing calls but one use case does not work. Let me explain: - I have some sbc for one destination. If one destination fail, kamailio try to second one etc.. - If no SBC is available for this destination (for exemple, all are down), i want to send my call to an error server (asterisk), play an error sound. To accomplish this, i re-write the RURI: "sip:error1@errorserver.local", and launch RELAY route. But kamailio dont make a DNS query to find IP of "errorserver.local", only change RURI and try to send to the last LCR gateway. I have also try to add a new branch, but same result... Why kamailio do not make a new DNS query ? *route[RELAY] { #if (is_method("INVITE")) { # if(!t_is_set("failure_route")) t_on_failure("MANAGE_FAILURE"); #} if (!t_relay()) { sl_reply_error(); } exit;}route[NOTFOUND]{ $ru = "sip:error1@errorserver.local"; xlog("L_WARN", "[$ci]|$si|Routing this call to $ru (error server)\n"); append_branch($ru); route(RELAY);}route[LCR] { xlog("L_INFO", "[$ci]|$si|Loading gateways...\n"); if (!load_gws(1)) { xlog("L_ERR", "[$ci]|$si|Can't load LCR gateways\n"); sl_send_reply("503", "Unable to load gateways"); exit; } else { $var(i) = 0; while(is_avp_set("$(avp(i:709)[$var(i)])")) { xlog("L_INFO", "[$ci]|$si|This gateway was found: gw_uri_avp[$var(i)]=$(avp(i:709)[$var(i)]) \n"); $var(i) = $var(i) + 1; }; if(is_avp_set("$avp(i:709)")) { xlog("L_INFO", "[$ci]|$si|Trying gateway ‘$avp(i:709)’\n"); } else { xlog("L_INFO", "[$ci]|$si|No more gateways. Goto route NOTFOUND...\n"); route(NOTFOUND); }; # try the first matched gateway if (next_gw()) { xlog("L_INFO", "[$ci]|$si|New request URI built: $ru . Relaying it (and arming backup destination if needed)."); # Route to failure for failover t_on_failure("lcrfailure"); route(RELAY); } else { xlog("L_INFO", "[$ci]|$si|No more gateways (after next_gw()). Goto route NOTFOUND..."); route(NOTFOUND); }; }; exit;}failure_route[lcrfailure] { # the previous gateway is no good if (t_check_status("408|50[34]")) { xlog("L_WARN", "[$ci]|$si|Entering in entering failure_route[lcrfailure]. Error code: $T_reply_code \n"); if(is_avp_set("$avp(i:709)")) { xlog("L_INFO", "[$ci]|$si|Trying gateway ‘$avp(i:709)’\n"); } else { xlog("L_ERR", "[$ci]|$si|AVP is not set\n"); }; if (next_gw()) { xlog("L_INFO", "[$ci]|$si|New request URI built: $ru . Relaying it (and arming backup destination if needed)."); # Route to failure for failover t_on_failure("lcrfailure"); route(RELAY); } else { xlog("L_INFO", "[$ci]|$si|No more gateways (after next_gw()). Goto route NOTFOUND..."); route(NOTFOUND); }; exit; };}* Many thanks, Regards.

3 2

how to config gateway in kamailio
by Gaurav Bmotra 26 Sep '16

26 Sep '16

Hi i want to config gateway in kamailio i want to know how can i config IP Authenticate gateway and user/ pass gateway -- *Regards:* Gaurav Bmotra

2 1

Scripting Issue
by aman jeff 26 Sep '16

26 Sep '16

Hai Team, I can't able to set maxptime in SDP Please help me

1 1

Location Table In db_mode 3
by Gholamreza Sabery 25 Sep '16

25 Sep '16

I have multiple Kamailio servers in an Active/Active scenario (the database is a MySQL server) using db_mode 3. But when the number of clients goes up writes on the location table become too much and it causes database problems. I even put my location table on a separate SSD disk on the server to separate it's IO but still at times I have problems. How can I move location table into memory? Is this a good solution? What other options do I have?

4 12

how to config gateway in kamailio
by Bmotra Infotech 24 Sep '16

24 Sep '16

i want to config gateway in kamailio i want to know how can i config IP Authenticate gateway and user/ pass gateway Bmotra Infotech

2 1

Re: [SR-Users] LCR routing, and finaly route to media gateway if not found
by Loic Chabert 23 Sep '16

23 Sep '16

In fact, kamailio is out from lcr at this point, no more gateway to try (all gateways are down). I just want to send call to a static server for playing sound. To be more specific, if no gateways are not set on database, a sound must be played. Should i change $du and $ru to make it work ? Le 23 sept. 2016 4:36 PM, "Juha Heinanen" <jh(a)tutpro.com> a écrit : Loic Chabert writes: > To accomplish this, i re-write the RURI: "sip:error1@errorserver.local", > and launch RELAY route. But kamailio dont make a DNS query to find IP of > "errorserver.local", only change RURI and try to send to the last LCR > gateway. > > I have also try to add a new branch, but same result... Why kamailio do not > make a new DNS query ? check if both hostname and ip address are given for the gateway. if so, next_gw() also sets destination uri. -- juha _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

Kamailio ims nat issue: PRACK/ACK ruri changing in p-cscf
by Owais 23 Sep '16

23 Sep '16

Hi all, My scenario is as follows: I have All the CSCFs on the same IP using different ports. P-CSCF advertises a public IP. Rest of the nodes including I-CSCF and S-CSCF listen local only. Behavior: When an ACK or PRACK goes from P-CSCF to S-CSCF, the RURI changes such that alias parameter is removed. Hence, P-CSCF ends up sending PRACK and ACK on private IP of UE. Here's what the config looks like: #!define WITH_NAT Any hints? -- View this message in context: http://sip-router.1086192.n5.nabble.com/Kamailio-ims-nat-issue-PRACK-ACK-ru… Sent from the Users mailing list archive at Nabble.com.

1 0

604 Does not exist anywhere - HSS User Unknown
by Rodrigo Moreira 23 Sep '16

23 Sep '16

Hi, After I am registred on Kamailio a try to make a call (INVITE), and S-CSCF returns the messege: 604 Does not exist anywhere - HSS User Unknown What I am doing wrong? Is the contact entry (Alice) wrong? How should the contact entry on the list? sip:alice@mydomain.com Best Regards., -- Rodrigo M. (37) 9132-4539 (34) 9889-3069 rodrigo.moreira2007

3 3

Kamailio Not Starting
by Linux Vince 23 Sep '16

23 Sep '16

I am getting below error when i start kamailio Not starting : invalid configuration file! 0(2360) ERROR: <core> [mem/f_malloc.c:415]: fm_search_defrag(): fm_search_defrag(0x7f732c5ff010, 232); Free fragment not found! 0(2360) ERROR: <core> [mem/f_malloc.c:498]: fm_malloc(): fm_malloc(0x7f732c5ff010, 232) called from core: route_struct.c: mk_action(118), module: core; Free fragment not found! 0(2360) CRITICAL: <core> [route_struct.c:137]: mk_action(): memory allocation failure 0(2360) ERROR: <core> [mem/f_malloc.c:415]: fm_search_defrag(): fm_search_defrag(0x7f732c5ff010, 40); Free fragment not found! 0(2360) ERROR: <core> [mem/f_malloc.c:498]: fm_malloc(): fm_malloc(0x7f732c5ff010, 40) called from core: cfg.y: yyparse(2636), module: core; Free fragment not found! 0(2360) : <core> [cfg.y:3368]: yyerror_at(): parse error in config file /etc/kamailio/proxy.cfg, line 5372, column 3-28: Not enough memory ERROR: bad config file (1 errors) I tried increasing SHM_MEMORY and PKG_MEMORY under /etc/default/kamailio I doubled memory but still getting same error. Is there any other place i should look at ?

2 3

Kamailio ims nat issue: PRACK/ACK ruri changing in p-cscf
by Owais Ahmad 23 Sep '16

23 Sep '16

Hi all, My scenario is as follows: I have All the CSCFs on the same IP using different ports. P-CSCF advertises a public IP. Rest of the nodes including I-CSCF and S-CSCF listen local only. Behavior: When an ACK or PRACK goes from P-CSCF to S-CSCF, the RURI changes such that alias parameter is removed. Hence, P-CSCF ends up sending PRACK and ACK on private IP of UE. Here's what the config looks like: listen=udp:lo.ca.l.ip:6000 advertise pu.bl.ic.ip:6000 alias=pcscf-ims.domain.net #!define WITH_NAT Any hints?

1 0

Problem reseting TMX statistics
by Átila de Moura Tavano Moretto 23 Sep '16

23 Sep '16

I'm currently using version 4.4.2 and am having trouble trying to reset the statistics in the TMX module. According to its documentation, all statistics except "inuse_transactions" can be reset. Checking the tmx_mod.c files for the statistics definition, I saw that the STAT_NO_RESET flag wasn't used for any statistic listed there (which seems odd because I think it should have been set for the inuse_transactions statistic): static stat_export_t mod_stats[] = { {"UAS_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_uas_trans }, {"UAC_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_uac_trans }, {"2xx_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_2xx }, {"3xx_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_3xx }, {"4xx_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_4xx }, {"5xx_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_5xx }, {"6xx_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_6xx }, {"inuse_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_inuse }, {"active_transactions", STAT_IS_FUNC, (stat_var**)tmx_stats_trans_active}, {"rpl_received", STAT_IS_FUNC, (stat_var**)tmx_stats_rcv_rpls }, {"rpl_absorbed", STAT_IS_FUNC, (stat_var**)tmx_stats_abs_rpls }, {"rpl_generated", STAT_IS_FUNC, (stat_var**)tmx_stats_rld_loc_rpls}, {"rpl_relayed", STAT_IS_FUNC, (stat_var**)tmx_stats_rld_rcv_rpls}, {"rpl_sent", STAT_IS_FUNC, (stat_var**)tmx_stats_rld_tot_rpls}, {0, 0, 0} }; All seems ok, but when I try to reset the stats, the command simply doesn't work. The values remain unaltered. The clear command doesn't work either, it just returns the current values. Can anyone help me trying to debug this problem, I'm very new to Kamailio and its code, and don't have a good understanding of how it the statistics gathering/reset works. Thanks. Atila Moretto

1 0

Location Table In db_mode 3
by Gholamreza Sabery 23 Sep '16

23 Sep '16

I have multiple Kamailio servers in an Active/Active scenario (the database is a MySQL server) using db_mode 3. But when the number of clients goes up writes on the location table become too much and it causes database problems. I even put my location table on a separate SSD disk on the server to separate it's IO but still at times I have problems. How can I move location table into memory? Is this a good solution? What other options do I have?

1 0

Dimension number of child
by Igor Potjevlesch 23 Sep '16

23 Sep '16

Hello, I'm wondering if there's a best practise to dimension the number of child to setup? I see, very often, Kamailio that stops to reply to SIP traffic for couple a seconds. This cause some calls drop if RE-INVITE are not processed for example. So, I'm wondering if the issue is not due to a lack of child for processing the SIP traffic. Thank you for your inputs. Regards, Igor.

1 0

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