- sr-users - kamailio.org

forcing socket doesn't work for ACK
by David Villasmil 03 May '19

03 May '19

Hello guys, I have public and private IPs, and i need to force the sending socket replying to internal out externals endpoint, so i'm using this: function to check the destination (i tried using $sndto(ip) but it always returns NULL): route[CHECK_DEST_NET] { if ( $du=~"sip:172\..*" ) { setflag(FLAG_TO_PRIVATE); xlog("L_ERR", "[CHECK_SOURCE] Packet going to PRIVATE -> [$du]\n" ); } else { setflag(FLAG_TO_PUBLIC); xlog("L_ERR", "[CHECK_SOURCE] Packet going to PUBLIC -> [$du]\n" ); } return; } on my relay: route(CHECK_DEST_NET); if (isflagset(FLAG_TO_PRIVATE)) { xlog("L_ERR", "[RELAY] forcing socket to PRIVATE NET\n" ); force_send_socket(LISTEN_INSIDE_IF:LISTEN_INSIDE_PORT); } else { xlog("L_ERR", "[RELAY] forcing socket to PUBLIC NET\n" ); force_send_socket(LISTEN_OUTSIDE_IF:LISTEN_OUTSIDE_PORT); } This works nicely, except for ACKs and potentially other packets sent statelessly? Anyone knows how to fix this? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

6 26

run some extra things before relying message
by Mojtaba 02 May '19

02 May '19

Hello, I have encountered with problem in Kamailio. At first glance it looks simple. Considering the Data Lump feature in Kamailio, All changes in SIP body, does not apply quickly. So if we are using RTPEngine or RTPProxy, How we can get body message (Actually IP and port from SDP after rtpengine_manage function) before relaying message? Suppose i want get some information and run some extra things before relying message? I just added my extra things in t_on_reply block, but it's strange why it doesn't work before relaying! -Mojtaba Esfandiari.S

2 2

Problem with topoh
by Phillman25 Kyriacou 02 May '19

02 May '19

Dear List I have an issue with topoh module. Specifically for in-dialog requests, if the encoded uri is altered i.e. initial encoded contact uri is Contact: <sip:X.X.X.X;line=sr-huAWi7oPBYsF.cC0L2lWOUlPwbSWOUl6wURTw24W> and UAS sends an in-dialog INVITE like this adding tgrp and user=phone: INVITE sip:X.X.X.X;line=sr-huAWi7oPBYsF.cC0L2lWOUlPwbSWOUl6wURTw24W;tgrp=1069912311;user=phone Kamailio cant seem to match/decode this and responds "404 Not Found" I have also set: modparam("topoh", "uri_prefix_checks", 0) As specified here: https://www.kamailio.org/docs/modules/4.4.x/modules/topoh.html#idp37582868 But still doesn't work. Is there a bug with this module? # ==================================================== # ================ TOPOLOGY HIDING =================== # ==================================================== modparam("topoh", "mask_key", "secret") modparam("topoh", "mask_ip", "X.X.X.X") modparam("topoh", "mask_callid", 0) modparam("topoh", "uparam_name", "line") modparam("topoh", "uparam_prefix", "sr-") modparam("topoh", "vparam_name", "branch") modparam("topoh", "vparam_prefix", "z9hG4bKsr-") modparam("topoh", "uri_prefix_checks", 0) Thanks in advance for your support Phillip

2 4

Stack smash error when using auth_ephemeral with KEMI
by Grant Bagdasarian 02 May '19

02 May '19

Hello, I was trying out the auth_ephemeral module inside the xhttp event_route in Python KEMI, and discovered a possible bug. https://pastebin.com/4Ghumfis The following will result in the stack smash error. Whenever I leave out the unixtimestamp the module will complain about not being able to convert to an int, which make sense, but doesn't lead to a stack smash error and a complete crash of Kamailio. KSR.auth_ephemeral.autheph_authenticate("1576734089:grant", "1234"). According to the documentation this module can only be used in REQUEST_ROUTE, but I found a presentation online which also used this module and that particular function in event_route, which is what I need for WebSocket authentication. Hope this helps. Please let me know if you need more information. Regards, Grant

2 1

Kamailio 5.2.2 IMS authorization failed by the SCSCF
by Woscek, Martin W. 02 May '19

02 May '19

Hi, We are getting a SIP 401 Unauthorized -Challenging the UE error originating by the SCSCF. It looks to be related to the nonce/cnonce. Is there something we have misconfigured in our SCSCF module? Log output: May 1 14:01:45 imskamailio /usr/local/sbin/kamailio[18489]: DEBUG: ims_auth [authorize.c:786]: authenticate(): Nonce or response missing: nonce len [-108158976], response16 len[1873606560] May 1 14:01:45 imskamailio /usr/local/sbin/kamailio[18492]: DEBUG: ims_auth [utils.c:168]: get_nonce_response(): Calling find_credentials with realm [ims522.irisims.org] May 1 14:01:45 imskamailio /usr/local/sbin/kamailio[18492]: DEBUG: ims_auth [utils.c:195]: get_nonce_response(): Found nonce response May 1 14:01:45 imskamailio /usr/local/sbin/kamailio[18492]: INFO: ims_auth [authorize.c:827]: authenticate(): uri=sip:irisims.org nonce=9863c63855c8dd888c5c18e79fc61156 response=ce7ce73872b92aff2defa76dd1aff1ac qop=auth-int nc=00000001 cnonce=4fe4daacd69fa3d609c8b396d331eae6 hbody=d41d8cd98f00b204e9800998ecf8427e May 1 14:01:45 imskamailio /usr/local/sbin/kamailio[18492]: DEBUG: ims_auth [rfc2617.c:135]: calc_response(): calc_response(_ha1=7a1b479a0513914e71bcd6d005b3e17d, _nonce=9863c63855c8dd888c5c18e79fc61156, _nc=00000001,_cnonce=4fe4daacd69fa3d609c8b396d331eae6, _qop=auth-int, _auth_int=1,_method=REGISTER,_uri=sip:irisims.org,_hentity=d41d8cd98f00b204e9800998ecf8427e) Thanks, Martin

2 1

New voisonics.com - Kamailio related products and services
by David Cunningham 02 May '19

02 May '19

Hello, Usually it's our client's VoIP technology being updated. Fortunately we found a little spare time to completely revamp the Voisonics website as well! https://voisonics.com/ Voisonics provides products and services in the commercial VoIP and hosted PBX sector; in particular relating to Kamailio, Asterisk, and Enswitch. The business has been specialising in this area for almost 10 years. Please feel free to contact me off-list with any questions. Kind regards, -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782

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Support for JWT token authentication?
by David Dean 01 May '19

01 May '19

I came across this thread from Jan 2015 discussing the addition of JWT token authentication to Kamailio: https://github.com/kamailio/kamailio/issues/29 Was JWT authentication ever added? Or has anyone else on this list got JWT working somehow?

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CSeq not incrementing - app_ruby
by Andrew White 01 May '19

01 May '19

Hi all, I’m still building with app_ruby and loving it - for the most part! I’m having an issue with one of my providers responding with a 482 on auth invite. Researching, it appears that my CSeq isn’t incrementing and this is the reason for the issue. I already have the relevant flag set in my kamailio.cfg however: modparam("dialog", "track_cseq_updates", 1) During my uac_auth, I’ve tried manually setting the diff, as well as running msg_iflag_reset (as suggested by Daniel - https://github.com/kamailio/kamailio/issues/1359 <https://github.com/kamailio/kamailio/issues/1359> - unfortunately the function isn’t exported in KEMI): if KSR::UAC.uac_auth() then KSR.info("UAC authed, relaying") #KSR::COREX.msg_iflag_reset("UAC_AUTH") KSR::PV.sets("$dlg_var(cseq_diff)", "1") KSR.info("CSeq diff: #{KSR::PV.gete("$dlg_var(cseq_diff)")}") KSR::TM.t_relay() Unfortunately in both cases the actual CSeq doesn’t increment in the second INVITE: INVITE 1 (unauthenticated): > CSeq: 102 INVITE INVITE 2 (with Authorization header): > CSeq: 102 INVITE I’m unsure if I’ve missed something silly around the track_cseq_updates flag, or if app_ruby somehow isn’t interacting with the dialog module when running uac_auth to trigger the increment? Thanks! Kind regards, Andrew

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listening at link local IPv6 address
by Juha Heinanen 01 May '19

01 May '19

My K host is behind a mobile WiFi hotspot and I tried to make K listen at the IPv6 address fe80::6e29:95ff:fe7d:37e6/64 that the host has received from the hotspot. Using that address on listen line gives error: Apr 21 08:55:18 salmon /usr/bin/sip-proxy[6887]: ERROR: <core> [core/udp_server.c:395]: udp_init(): bind(8, 0x7ff97dfb3fe4, 28) on FE80:0:0:0:6E29:95FF:FE7D:37E6: Invalid argument Apr 21 08:55:18 salmon /usr/bin/sip-proxy[6887]: ERROR: <core> [core/udp_server.c:401]: udp_init(): might be caused by using a link local address, try site local or global I can ping that address like this: $ ping6 fe80::6e29:95ff:fe7d:37e6%wlp1s0 PING fe80::6e29:95ff:fe7d:37e6%wlp1s0(fe80::6e29:95ff:fe7d:37e6%wlp1s0) 56 data bytes 64 bytes from fe80::6e29:95ff:fe7d:37e6%wlp1s0: icmp_seq=1 ttl=64 time=0.030 ms I then tried to add %wlp1s0 suffix to the listen address, but got error: Apr 21 09:10:35 salmon sip-proxy[7896]: 0(7948) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 408, column 34: ip address, interface name or hostname expected Did I do something wrong? If not, why is it that K cannot listen at a link local address? It is something that Linux networking stack does not support? -- Juha

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Pipelimit with clean_unused
by Cinthia Leung 30 Apr '19

30 Apr '19

Hello, What is your recommendation on using clean_unused with db defined pipes? >>pl_check("systemwide"); If I leave the system idle, they are eventually "cleaned" as expected. In our use case, that means failed calls unfortunately. I am trying to see if I can mix both db/static pipes with dynamic/run-time pipes. Thanks Cindy

1 1

Next hop of reply, in onreply_route
by Alex Balashov 30 Apr '19

30 Apr '19

I am fairly sure it’s been asked a number of times, but is there a way to know the destination, and more especially the transport, of the next hop of a reply message, viewable from onreply_route? For requests, we have $nh and many other PVs which in some way evidence where it’s going. And yes, I’m aware that this can be previewed from the onsend_routes*, but I need to amalgamate that knowledge with some transaction-specific data. Thanks! — Sent from mobile, with due apologies for brevity and errors.

2 2

"ERROR: reply too big" and ctl binrpc size
by Karsten Horsmann 30 Apr '19

30 Apr '19

Hi all, i use kamailio 5.0.7 in production and i noticed that i cant listen the dialogs after a few days of traffic on this server. Then i got "kamcmd dlg.list_ctx " - "ERROR: reply too big". So now my question - i want to debug my "sleeping dialogs" with dlg.list - how big should it change the values of without knowing the size of the current filled up binrpc stuff? Or is there a better option to debug dialog-module? modparam("ctl", "binrpc_max_body_size", ??) modparam("ctl", "binrpc_struct_max_body_size", ?) -- Cheers and see you in Berlin *Karsten Horsmann*

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Re: [SR-Users] SIP basic flow question: R-URI in ACK
by Yufei Tao 30 Apr '19

30 Apr '19

Hi Daniel, Thank you very much for clarifying! Cheers, Yufei On 30 Apr 2019 at 08:10, <Daniel-Constantin Mierla <miconda(a)gmail.com>> wrote: Hello, there is an errata for that RFC, but it refers only to the BYE, where it adds back the transport=tcp, but the ACK R-URI should follow the same rules, see: * https://www.rfc-editor.org/errata/eid5294 Cheers, Daniel On 28.04.19 22:54, Yufei Tao wrote: Hi, I have a question regarding basic SIP flow for establishing a call session, and wonder if anyone can help me clarify. For normal call set up, INVITE-OK-ACK, from RFC3261, I believe the ACK’s R-URI should be a copy of the Contact header of the OK message. However in RFC3665 sections 3.1 and 3.2 for example, the ACK’s R-URI missed the parameters from the Contact header of the OK, e.g. In OK the Contact header is: Contact: <sip:bob@client.biloxi.example.com;transport=tcp> <sip:bob@client.biloxi.example.com;transport=tcp> And ACK the R-URI is: ACK sip:bob@client.biloxi.example.com SIP/2.0 Which has got the parameter ‘transport=tcp’ removed – why is this? In this case if the proxy connected to Bob handles ACK in the usual way as it would for all in-dialog requests, i.e. based on routing headers only and not extra processing, it’ll try to relay the ACK message to Bob using the default transport UDP which is not expected and will fail. Can anyone help explain why the parameters are removed in RFC3665 examples please? Have I missed anything? Thank you very much! Yufei _______________________________________________ Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

1 0

docker image for non x86 arch
by Sergey Safarov 30 Apr '19

30 Apr '19

Is the community involved in kamailio docker image for ARM architecture and others not x86 platforms? ARM ARM 64 IBM POWER IBM Z PowerPC 64 LE

3 3

SIP basic flow question: R-URI in ACK
by Yufei Tao 30 Apr '19

30 Apr '19

Hi, I have a question regarding basic SIP flow for establishing a call session, and wonder if anyone can help me clarify. For normal call set up, INVITE-OK-ACK, from RFC3261, I believe the ACK’s R-URI should be a copy of the Contact header of the OK message. However in RFC3665 sections 3.1 and 3.2 for example, the ACK’s R-URI missed the parameters from the Contact header of the OK, e.g. In OK the Contact header is: Contact: <sip:bob@client.biloxi.example.com;transport=tcp> And ACK the R-URI is: ACK sip:bob@client.biloxi.example.com SIP/2.0 Which has got the parameter ‘transport=tcp’ removed – why is this? In this case if the proxy connected to Bob handles ACK in the usual way as it would for all in-dialog requests, i.e. based on routing headers only and not extra processing, it’ll try to relay the ACK message to Bob using the default transport UDP which is not expected and will fail. Can anyone help explain why the parameters are removed in RFC3665 examples please? Have I missed anything? Thank you very much! Yufei

3 2

remove_record_route - bug?
by Sergiu Pojoga 29 Apr '19

29 Apr '19

Hi ppl, After a branch route fails, I need to re-evaluate the next destination and adjust RR parameters, if need be, like for example adjust the nat=yes|priv. failure_route[MANAGE_PSTN_FAILURE] { ... remove_record_route(); record_route(); route(NATMANAGE); ... route(RELAY); } The result is that the new branch is stripped of "Record-Route: <sip:MYIP>", but the params remained. Moreover, record_route hasn't been re-added as instructed in the failure route. Request headers of the initial branch that will eventually timeout and fail: 2019/04/27 18:29:18.034992 10.22.0.1:5060 -> 10.22.0.100:5060 INVITE sip:1514XXXXXXX@10.22.0.100 SIP/2.0 Record-Route: <sip:10.22.0.1;r2=on;lr=on;did=9e2.3dd2;nat=priv> Record-Route: <sip:65.XX.XX.1;r2=on;lr=on;did=9e2.3dd2;nat=priv> New request branch after failure: 2019/04/27 18:29:19.538292 65.XX.XXX.1:5060 -> 65.XX.XX.2:5060 INVITE sip:1514XXXXXXX@65.XX.XX.2 SIP/2.0 ;lr=on;did=9e2.3dd2;nat=priv> kamailio -v version: kamailio 5.1.6 (x86_64/linux) 7d1964 Much obliged. --Sergiu

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Vide SIP conferencing kamailio with freeswitch
by Gaurav Bmotra 28 Apr '19

28 Apr '19

HI i m gaurav i have installed kamailio 5.1.x on Ubuntu server 18.4 i want a solution for video conferencing with kamailio as i am using freeswitch with it is there reference guide for that plz help -- *Regards:* Gaurav Kumar

2 1

Re: [SR-Users] [sr-dev] Vide SIP conferencing kamailio with freeswitch
by Henning Westerholt 28 Apr '19

28 Apr '19

Hello, please ask this usage related questions on our "sr-users" list. Kamailio is not doing much regarding the video calls. E.g. if your calls work for voice with kamailio, they should also work for video. All the setup needs to be done in freeswitch. I found a (outdated) howto with google: https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video Maybe this helps. Henning Am 27.04.19 um 13:22 schrieb Gaurav Bmotra: HI i m gaurav i have installed kamailio 5.1.x on Ubuntu server 18.4 i want a solution for video conferencing with kamailio as i am using freeswitch with it is there reference guide for that plz help _______________________________________________ Kamailio (SER) - Development Mailing List sr-dev(a)lists.kamailio.org<mailto:sr-dev@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

1 0

How to configure ipv6 in kamailio
by Santosh Kumar Dash 28 Apr '19

28 Apr '19

Hi All, I have a kamailio installed in my setup which is working properly in IPV4 configuration. I want to perform IPV6 configuration, I don't have any idea how to configure it, I have tried few things by changing *kamailio.cfg* file but did not work. I am sharing my *kamailio.cfg *file which is working fine in ipv4 Kindly help me to configure for IPV6. lets say my Ipv6 address is : 2402:3a80:d1a:f45a:fe44:2e54:ed2b:9bb2 Thanks,

3 3

Can't set module parameter
by Tmbc 26 Apr '19

26 Apr '19

Hi all Having this issue with kamailio: Can't set module parameter Apr 25 09:57:10 kamailio: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <dst_avp> of type <1> not found in module <dispatcher> Apr 25 09:57:10 kamailio: CRITICAL: <core> [core/cfg.y:3511]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 119, column 50: Can't set module parameter Apr 25 09:57:10 kamailio: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <grp_avp> of type <1> not found in module <dispatcher> Apr 25 09:57:10 kamailio: CRITICAL: <core> [core/cfg.y:3511]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 120, column 50: Can't set module parameter Apr 25 09:57:10 kamailio: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <cnt_avp> of type <1> not found in module <dispatcher> Apr 25 09:57:10 kamailio: CRITICAL: <core> [core/cfg.y:3511]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 121, column 50: Can't set module parameter here is full conf file: https://dpaste.de/bP3E OS: NAME="CentOS Linux" VERSION="7 (Core)" Software version: kamailio 5.2.2 (x86_64/linux) 67f967 Thanks in advance, tmbc

2 1

Re: [SR-Users] Config to control TCP connection reuse
by R, Rajkumar (Raj) 24 Apr '19

24 Apr '19

I think elaborating on the setup may help someone to provide inputs on this. Below is the setup. Client 1 --> Kamailio proxy --> Backend SIP server Client 2 --> Kamailio proxy --> Backend SIP server Client 3 --> Kamailio proxy --> Backend SIP server After I register all three clients, see only one TCP connection is being used between "Kamailio proxy - Backend SIP server". Is there a way to achieve separate TCP connection to Backend SIP server from kamailio. thanks, raj From: R, Rajkumar (Raj) Sent: Monday, April 22, 2019 3:45 PM To: sr-users(a)lists.kamailio.org Subject: Config to control TCP connection reuse Hi, In my setup(kamailio in stateless mode to a backend server), for any number of client registrations kamailio (in stateless mode) is reusing the TCP connection. Let's say if I register 3 different clients through kamailio to my backend server - see only one TCP connection is used, which I do not want. I have gone through the core modules to see if any TCP config param exists to control on this, but couldn't find one. Could anyone help how to achieve different TCP connections to same destination in kamailio? thanks, raj

2 1

DMQ_USRLOC
by Gholamreza Sabery 24 Apr '19

24 Apr '19

Hi, Recently, I started to use dmq_usrloc module. I successfully set it up. Unfortunately, the documentation does not specify the behavior of dmq_usrloc module with other Kamailio modules. For example, I see a bunch of KDMQ messages are exchanged between nodes, but how the lookup() function behaves with dmq_usrloc, or what is it's effect on a db_mode of 3! Regards

5 6

JSON error - control character 0xd
by David Dean 23 Apr '19

23 Apr '19

I'm trying to send an SDP from a SIP client to Janus using JSON over HTTP. The problem is that Janus is erroring on the SDP content with the following error: JANUS HTTP: Get SDP for echo plugin: {#012 "janus": "error",#012 "error": {#012 "code": 454,#012 "reason": "JSON error: on line 9: control character 0xd near '\"v=0'"#012 }#012} Result code 200 I tried using {s.escape.common} to escape any quotations, commas etc but it doesn't appear to replace the control characters with newlines. Any suggestions on how best to remove these control characters so that Janus will accept the SDP over JSON? Here is the relevant section of kamailio.cfg: # HTTP: Request SDP for echo test plugin sdp_get("$avp(sdp)"); $var(res) = http_connect("janus", "/janus/$var(JANUS-ID)/$var(ECHO-ID)", "application/json", "{ \"janus\" : \"message\", \"transaction\" : \"testtesttest99\", \"body\" : { \"audio\" : true }, \"jsep\" : { \"type\" : \"offer\", \"sdp\" : \"$(avp(sdp){s.escape.common})\" }}", "$avp(janus-pluginsession)"); xlog("L_INFO", "JANUS HTTP: Get SDP for echo plugin: $avp(janus-pluginsession) Result code $var(res)\n");

3 6

Direction of replies
by Duarte Rocha 23 Apr '19

23 Apr '19

Greetings, Is there a way to know in reply_route if a reply has direction downstream or upstream regarding the first INVITE? I know there are vars and methods for finding this out on requests, but i can't find anything for replies. Thanks in advance, Best Regards, Duarte Rocha

2 1

CDR start time of a canceled call
by Jan-Hendrik Dörner 23 Apr '19

23 Apr '19

Hello, I just noticed that the logged „start_time" as well as „end_time" of a canceled call is set to the „invite time“. If done this way, it is harder to calculate the „ringing time“. Is there an option to change this behaviour (i.e. to log the time of the received „bye“ instead)? (If done so, the ringing time would always be the time between start and invite …) Or is there a special reason for this behaviour? My CDR settings are as followed: modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_expired_dlg_enable", 1) # log unanswered / open dialogs modparam("acc", "cdr_on_failed", 1) # log also not answered calls modparam("acc", "cdr_extra", "inviteTime=$dlg_var(inviteTime)" modparam("acc", "cdr_log_enable", 1) # activate write to syslog modparam("acc", "cdr_facility", "LOG_LOCAL6") # syslog modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "time_mode", 0) #syslog timestamp modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) # needed for syslog modparam("acc", "log_missed_flag", FLT_ACCMISSED) # need for logging missed calls in syslog Regards Jan-Hendrik

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