sr-users November 2016

sr-users@lists.kamailio.org

57 participants
123 discussions

Getting destination ip
by David Villasmil 06 Dec '16

06 Dec '16

Hello Guys, I'm wondering how do I get destination IP on a reply? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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Re: [SR-Users] Kamailio Processes Stop Responding
by Peter DiSanto 05 Dec '16

05 Dec '16

OK thank you Alex. I will look into your suggestions next time the problem arises. Each call does generate one SQL LCR lookup, one CDR INSERT, and also a timer route that runs 3 queries every few minutes. Once Kamailio stops responding it will continue that way indefinitely until a process restart. Will report back. Thanks. Pete D

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Local messages are malformed in Kamailio 4.4.X
by Helio Okuyama 01 Dec '16

01 Dec '16

Local messages are malformed in Kamailio 4.4.X. For example after receiving an error response (603) to an INVITE message, the ACK is generated with header Max-Forwards concatenated to header CSeq and double CRLF, which is wrong. Via: SIP/2.0/TLS 173.16.10.61:5061 ;branch=z9hG4bKc782.5b960c7c48a4363a0126963daa797eb0.0;i=7 From: "22861"<sip:22861@173.16.10.61;transport=tls>;tag=snl_OIPEOzJUNy Call-ID: 61b500a71b510a1Pj61qq1k9dPW To: <sip:551138174108@174.16.4.108:5061;transport=tls>;tag=435012226 CSeq: 2351 ACKMax-Forwards: 70 User-Agent:... The fix for that seems to be just the order of append_str in function build_local() from tm\t_msgbuilder.c. append_str( p, method, method_len ); + append_str( p, CRLF, CRLF_LEN ); append_str( p, MAXFWD_HEADER, MAXFWD_HEADER_LEN ); - append_str( p, CRLF, CRLF_LEN );

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help with string modification
by Sergey Basov 01 Dec '16

01 Dec '16

Hello All. I have some troubles with upstream sip switch. It ignores SIP packets which contains: User-Agent: FPBX-2.11.0(11.17.1) or Server: User-Agent: FPBX-2.11.0(11.17.1) If space is present before first "(" then sip switch works as expected So my question is: how corektly make analyze and modify this headers using kamailio? In which routes i must parse it? i need such modifications: User-Agent: FPBX-2.11.0(11.17.1) -> User-Agent: FPBX-2.11.0 (11.17.1) Server: User-Agent: FPBX-2.11.0(11.17.1) -> Server: User-Agent: FPBX-2.11.0 (11.17.1) Thank you. -- Best regards, Sergey Basov e-mail: sergey.v.basov(a)gmail.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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sr-users November 2016