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37519 discussions

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

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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.

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