- sr-users - kamailio.org

Memory leak in tm with push notifications
by Ivaylo Markov 14 Jun '18

14 Jun '18

Hello, I am trying to set up Kamailio as a push notifications proxy, closely following the example in the "Kamailio in a Mobile World" presentation (https://www.slideshare.net/FedericoCabiddu/kamailioinamobileworld-51617342). I am running Debian 9 and Kamailio 5.1.3 from the official Debian repositories. I believe the main modules involved in the issue below are tm, tmx, and tsilo. Every call passing through the proxy leads to a small memory leak in the tm module - there is a large amount of "delayed free" memory cells from tm's internal hash table. At some point the shared memory runs out and Kamailio restarts. Using the "kamcmd corex.shm_summary" command I was able to see that the top users of shared memory are "tm: h_table.c: build_cell" and "core: core/sip_msg_clone.c: sip_msg_shm_clone" with the same allocation count. I experimented with removing different parts of the configuration and noticed that commenting out the "t_continue(...)" call in the "PUSHJOIN" route (see slide #22) prevents the leak from happening. Maybe something in that function is incrementing the reference counter to the hash table cell, but it is not decrementing the counter when done? I tried looking around the source code of the tm and tmx modules, but saw nothing suspicious. I also tried using gdb with a breakpoint in t_continue_helper (tm/t_suspend.c:166) hoping to see what else is accessing the htable cell, but was unable to find anything of use. Has someone encountered anything like this? Can you provide more directions on debuggin this? I can provide some bits of configuration, but an entire test setup would be rather difficult, unfortunately. Thank you for your time, Ivo

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integrate huawei HSS with kamailio
by eyas barhouk 14 Jun '18

14 Jun '18

hello dears i'm trying to integrate kamailio IMS with Huawei Hss instead of Fraunhofer HSS , so is there any one has an experiment with this or tested it before ??? if yes, could you please guide me how to do that ? or where to start ? thanks in advance

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Forbidden - Private identity not found (Authorization: username)
by eyas barhouk 14 Jun '18

14 Jun '18

Hello dears when i was trying to build an IMS platform by using kamailio with sip digest authentication scheme and ims_auth module, i couldn't register and the result was as following : 1-the first register message passed to the s-cscf then S-cscf replaying with the message “401 unauthorized - challenging the ue” 2- the second register pass to the S-cscf with the authorization header then the S-CSCF replying with "Forbidden - Private identity not found (Authorization: username)" So based on that could any one tell me what is the reason of this error and how to fix it ?? Thanks in advance

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Using a var as timeout value in dlg_set_timeout_by_profile function
by jenus＠cyberchaos.nl 13 Jun '18

13 Jun '18

I'm trying to use the dlg_set_timeout_by_profile in the xhhtp route to update the dialog timeout for active calls. But if i use a var as the timeout value the call disconnects immediately. This works fine (call disconnects after 80 sec): dlg_set_timeout_by_profile("subscriber_outbound", "$var(SipName)@$var(SipDomain)::$var(DialogCallid)", "80"); This fails (call disconnects immediately: $var(DialogTimeout)= 80; dlg_set_timeout_by_profile("subscriber_outbound", "$var(SipName)@$var(SipDomain)::$var(DialogCallid)", "$var(DialogTimeout)"); Are vars supported in the dlg_set_timeout_by_profile function? And is it possible to use it in any route as described in the docs? I'm running kamailio 5.1.3. Thanks, Jan

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Registration of UACs i n a partially encrypted scenario
by mip FKF 13 Jun '18

13 Jun '18

Hello, we are pretty new to SIP and kamailio, we do have some questions regarding the following scenario: We have a number of UACs in a small network which are required to communicate without encryption because the are not able to consume certificates. We want to use kamailio (as a proxy?) to establish an encrypted connection to a backend UAS. 1. Is it possible to directly register the UACs with the UAS eventhough communication between kamailio and the UAS is encrypted ? 2. How do we need to configure kamailio in order to make this scenario work ? Thank you for your suppport

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Re: [SR-Users] Question on Contact Header on Kamailio 200 OK to PBX
by Floimair Florian 13 Jun '18

13 Jun '18

Yes it should, but only if “rewrite_contact” is set to “no” for the endpoint (chan_pjsip). With best regards Florian Floimair Innovation - Software-Development COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 http://www.commend.com<http://www.commend.com/> Security and Communication by Commend FN 178618z | LG Salzburg From: sr-users <sr-users-bounces(a)lists.kamailio.org> on behalf of "Wilkins, Steve" <swwilkins(a)mitre.org> Reply-To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Date: Tuesday, 12. June 2018 at 16:27 To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] Question on Contact Header on Kamailio 200 OK to PBX Hi All, Kamailio is sending a 200 OK back to an INVITE from Asterisk, and Asterisk is sending the ACK back but the AOR can’t be found in Kamailio 200 OK Kamailio to Asterisk with a contact of Contact: <sip:fgectrdv@9ot28m83bkur.invalid;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;transport=ws;ob> Asterisk ACK to Kamailio Request-Line: ACK sip:fgectrdv@192.33.11.108:5060;transport=TCP;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;ob SIP/2.0 Shouldn’t Asterisk be using fgectrdv@9ot28m83bkur in the ACK? Or did Kamailio send Incorrect Contact? Note: fgectrdv@9ot28m83bkur is the what Kamailio shows as AOR. Thank you, -Steve

1 0

Asterisk inviting Kamailio
by Wilkins, Steve 13 Jun '18

13 Jun '18

Hello All, I have noticed that sometimes when a call is made from one endpoint to another through Asterisk via Kamailio, Asterisk sends an INVITE to Kamailio even after the call has been established. Sometimes this does not happen. When it does happen, calls drop. Why would an INVITE be sent back to Kamailio? Also, Pan, I am working on the response you requested yesterday. Thanks you All, -Steve

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Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio
by Wilkins, Steve 12 Jun '18

12 Jun '18

Got it working. Thank you everyone. -----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Wilkins, Steve Sent: Sunday, June 10, 2018 3:06 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio >>>Email originates from a non-MITRE system. Use caution.<<< Alex, Pan, Daniel,... Could this group => group:BUNDLE audio video in Message Body have anything to do with my Kamailio Video issue. Thank you!! -Steve -----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Wilkins, Steve Sent: Sunday, June 10, 2018 12:14 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio Hi Alex, No, I'm not using rtpengine. It must definitely be some sort of Codec issue though, since I can seem to move the problem around and actually finally get video on the softphone. It was a little strange that I lost Audio on the WebRTC client though, since I only disabled VP8. Since Kamailio is only relaying I am so confused why introducing Kamailio is messing up Video. Thank you, -Steve -----Original Message----- From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Alex Balashov Sent: Sunday, June 10, 2018 11:46 AM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio Unless you're using rtpengine, you're just moving the problem around. Kamailio does nothing with SDP unless by way of rtpengine or told in some other way, such as sdpops. On June 10, 2018 11:44:47 AM EDT, "Wilkins, Steve" <swwilkins(a)mitre.org> wrote: >I just did a test where I disabled VP8 in Kamailio using SDPOPS and I >now get Video on the softphone however, I lost two-way Audio. Kamailio >seems to be doing something with the codecs but still can't put my >finger on it. The WebRTC Client, who is the caller, needs VP8 for >Video, and apparently Audio. > >I'm not sure if I getting closer or just moving the problem around. > >Thanks All! > >From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf >Of Wilkins, Steve >Sent: Saturday, June 9, 2018 12:47 PM >To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> >Subject: [SR-Users] No Video between WebRTC Client and Softphone when >using Kamailio...works without Kamailio > >>>>Email originates from a non-MITRE system. Use caution.<<< >I am desperately trying to resolve a big issue I have when using >Kamailio 5.2.0 and Asterisk 15.3. > >As soon as I go through Kamailio, I get no Video on either side of the >call; I do get two-way audio and the call stays connected. If I simply >go right through Asterisk, I also get two-way Video. I am having a >tough time determining why Kamailio is messing with the Video portion >of the call. >I have included a full pcap file. As a side note when comparing pcap >traces of the Non Proxied vs Proxied call, the INVITE and 200 OK from >Asterisk and Softphone are exactly the same. > >Thank you, >-Steve -- Alex -- Sent via mobile, please forgive typos and brevity. _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

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Question on Contact Header on Kamailio 200 OK to PBX
by Wilkins, Steve 12 Jun '18

12 Jun '18

Hi All, Kamailio is sending a 200 OK back to an INVITE from Asterisk, and Asterisk is sending the ACK back but the AOR can't be found in Kamailio 200 OK Kamailio to Asterisk with a contact of Contact: <sip:fgectrdv@9ot28m83bkur.invalid;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;transport=ws;ob> Asterisk ACK to Kamailio Request-Line: ACK sip:fgectrdv@192.33.11.108:5060;transport=TCP;alias=128.147.123.1~63486~6;alias=128.147.123.1~63486~6;ob SIP/2.0 Shouldn't Asterisk be using fgectrdv@9ot28m83bkur in the ACK? Or did Kamailio send Incorrect Contact? Note: fgectrdv@9ot28m83bkur is the what Kamailio shows as AOR. Thank you, -Steve

1 0

Facing Problem Registration / Invite pass through one side voice
by Ankit vasava 12 Jun '18

12 Jun '18

Hello , I have installed opensips latest version and have free switch. All user is registered with free switch through opensips like registration pass though you can find out my config file into attachments also pcap log. My free switch IP is *192.168.1.147:5060 <http://192.168.1.147:5060>* Opensip IP is : *192.168.1.12:5069 <http://192.168.1.12:5069>* I m sending registration and invite request to public IP and that public IP is bind to open sips. *Problem:* when external user who is registered through public ip when he is dial any extension number who is registered with local ip at that time we are facing one way voice. I need voice(RTP) could be transfer both the side. -- *Regards,Ankit vasava* Sr. VoIP Developer *Phone: *+91-9879491525 *Skype: *ankit.vasava Http://in.linkedin.com/in/ankitvasava

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unable to Kamailio IMS 5.1.4?
by vinay kumar 12 Jun '18

12 Jun '18

Hi All, Im getting following error when i start kamailio.cfg of icscf. ERROR: ctl [init_socks.c:115]: init_unix_sock(): ERROR: init_unix_sock: bind: No such file or directory [2] ERROR: ctl [ctl.c:290]: mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed ERROR: <core> [core/sr_module.c:990]: init_mod(): Error while initializing module ctl (/usr/local/lib64/kamailio/modules/ctl.so) can anyone help me on this?? Regards, Vinay

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unable to bring up kamailio 5.1.4
by vinay kumar 12 Jun '18

12 Jun '18

Hi, Im facing following issue when I tried to bring up kamailio IMS 5.1.4. ERROR: <core> [core/modparam.c:140]: set_mod_param_regex(): parameter <hashing_type> of type <2> not found in module <ims_usrloc_pcscf> CRITICAL: <core> [core/cfg.y:3449]: yyerror_at(): parse error in config file /usr/local/etc/kamailio/pcscf/kamailio.cfg, line 354, column 47: Can't set module parameter can anyone share complete installation steps for Kamailio IMS 5.1.4 for volte call?? Regards, Vinay

1 0

Problem with TLS
by Arik Halperin 12 Jun '18

12 Jun '18

Hello, I’m using TLS After receiving 200OK from kamailio: r2voip.clear2voipdialer I/(NativeSdk_2_0) 1528174138320 PJSIP: (NativeSdk_2_0) 1528174138320 PJSIP:2018-05 07:48:58.319 pjsua_core.c RX 2203 bytes Response msg 200/INVITE/cseq=8107 (rdata0x7a2c56fb38) from TLS 70.36.25.65:443: SIP/2.0 200 OK Via: SIP/2.0/TLS 10.134.232.109:44097;received=109.253.173.146;rport=31373;branch=z9hG4bKPj4MV5llP9SW5ufk-OcFB-Qh78PmIQFrRk;alias Record-Route: <sips:10.168.10.227:5099;r2=on;lr=on;ftag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO;nat=yes> Record-Route: <sips:70.36.25.65:443;transport=tls;r2=on;lr=on;ftag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO;nat=yes> From: "number" <sips:972523391991@kamprod.telemessage.com<mailto:972523391991@kamprod.telemessage.com>>;tag=mgMLDFMLmCZGzcpASoODG8XgeFJVtcRO To: <sips:1111111@kamprod.telemessage.com<mailto:1111111@kamprod.telemessage.com>>;tag=64H63g861ajHj Call-ID: Sq4jR85o3Caz2XTXo-71FKAdbJ1x9vz2 CSeq: 8107 INVITE Contact: <sip:1111111@10.168.10.200:5080;transport=tls> User-Agent: FreeSWITCH-mod_sofia/1.6.20+git~20180123T214909Z~987c9b9a2a~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Require: timer Supported: ti PJSIP responds with: Secure dialog requires SIPS scheme in Contact and Record-Route headers, ending the session What is the reason for this? How can I fix this issue? Thanks, Arik Halperin

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Disable basic authentication?
by Skip Morse 11 Jun '18

11 Jun '18

Hi All, I couldn't get a clear answer from the documentation. Based on the description of www_authenticate and proxy_authenticate: " The function verifies credentials according to RFC2617. If the credentials are verified successfully then the function will succeed and mark the credentials as authorized (marked credentials can be later used by some other functions). If the function was unable to verify the credentials for some reason then it will fail and the script should call www_challenge which will challenge the user again." The RFC outlines basic and digest auth. Is there a way to disable the ability for a UA to use basic auth? Or maybe these is disabled already? I'm likely understanding this wrong, it seems to me that it's allowed for a UA to auth with basic even if the challenge is for digest. Any clarification would be helpful. Thanks! -Skip

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SIP/2.0 603 Failed to get local SDP for sip to webrtc (SIPML5)call
by vinod mn 11 Jun '18

11 Jun '18

Hi i configured sipml5 and kamailio , sipml5 clients registers to kamailio when i make a call to kamailio call lands on the sipml5 client but i am getting this error *"SIP/2.0 603 Failed to get local SDP" *on chrome/firefox/opera webbrowsers. kamilio configuration as per http://www.kamailio.org/docs/ modules/4.4.x/modules/websocket.html#idp53411716 -- Thanks and regards Vinod.M.N

2 1

sip digest authentication
by eyas barhouk 11 Jun '18

11 Jun '18

Hello dears I’m trying to setup Kamailio with sip digest authentication scheme by using the following configuration from s-cscf side : modparam("ims_auth", "registration_default_algorithm", "HSS-Selected") and the result was as following : 1. when the register message passed form the s-cscf to the hss and in the diameter trace I found the “SIP-AUTHENTICATION-SCHEME” in the MAR request as “Digest-MD5” 1. After that the HSS send MAA message with “DIAMETER-ERROR-AUTH-SCHEME-NOT-SUPPORTED” 2. S-cscf replay with the message “Forbidden – privet identity not found (Authorization : username)” So based on that could any one tell where was my mistake or how can I setup Kamailio with SIP-DIGEST scheme Thanks

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JSON RPC Error: LUMP_RPL_BODY already added!
by Asgaroth 11 Jun '18

11 Jun '18

Hi All, We are trying to query the json rpc server on our registrar for the contact of an aor, but we are seeing the following error(s) being displayed in the log when we query the location table: ERROR: <core> [core/data_lump_rpl.c:83]: add_lump_rpl2(): LUMP_RPL_BODY already added! ERROR: xhttp [xhttp_mod.c:410]: xhttp_send_reply(): Error while adding reply lump We are getting the aor details back in response from the rpc call, but the error's in the log are a little concerning and I cannot see where we are doing something wrong. As an example, the following curl request returns the correct response, but the above error is displayed in the log, Am I doing something wrong here, I cannot see what the issue is: curl --header 'Content-Type: application/json' --data-binary '{ "jsonrpc": "2.0", "method": "ul.lookup", "params": [ "location", "user(a)example.com" ], "id": 3 }' http://registrar.example.com/rpc A secondary question is, is there an upper limit on the size of the "id" parameter? This kamailio instance is v5.0.6. Any thoughts as to what this error means? Thanks

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(N)DB_FIREBASE module?
by Pan Christensen 11 Jun '18

11 Jun '18

Greetings all. We are currently developing web and mobile apps using firebase: https://firebase.google.com/products/realtime-database/ . We have been discussing the possibility of Kamailio querying this database. I assume that it will be possible to query it using HTTP and JSON modules, but it would probably be easier to use the DB API. Any plans to develop such a module? With kind regards Pan B. Christensen Developer Phonect AS Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway E-mail: pan.christensen(a)phonect.no<mailto:pan.christensen@phonect.no> Mobile: 41 88 88 00 [cid:image007.png@01D3A0E8.376921D0] <http://www.phonect.no/> [facebook_2]<https://www.facebook.com/phonectno> [LinkedIn_logo_initials (1)] <https://www.linkedin.com/company/44983?trk=tyah&trkInfo=clickedVertical%3Ac…>

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Re: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio
by Wilkins, Steve 10 Jun '18

10 Jun '18

I just did a test where I disabled VP8 in Kamailio using SDPOPS and I now get Video on the softphone however, I lost two-way Audio. Kamailio seems to be doing something with the codecs but still can't put my finger on it. The WebRTC Client, who is the caller, needs VP8 for Video, and apparently Audio. I'm not sure if I getting closer or just moving the problem around. Thanks All! From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Wilkins, Steve Sent: Saturday, June 9, 2018 12:47 PM To: Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org> Subject: [SR-Users] No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio >>>Email originates from a non-MITRE system. Use caution.<<< I am desperately trying to resolve a big issue I have when using Kamailio 5.2.0 and Asterisk 15.3. As soon as I go through Kamailio, I get no Video on either side of the call; I do get two-way audio and the call stays connected. If I simply go right through Asterisk, I also get two-way Video. I am having a tough time determining why Kamailio is messing with the Video portion of the call. I have included a full pcap file. As a side note when comparing pcap traces of the Non Proxied vs Proxied call, the INVITE and 200 OK from Asterisk and Softphone are exactly the same. Thank you, -Steve

2 3

No Video between WebRTC Client and Softphone when using Kamailio...works without Kamailio
by Wilkins, Steve 09 Jun '18

09 Jun '18

I am desperately trying to resolve a big issue I have when using Kamailio 5.2.0 and Asterisk 15.3. As soon as I go through Kamailio, I get no Video on either side of the call; I do get two-way audio and the call stays connected. If I simply go right through Asterisk, I also get two-way Video. I am having a tough time determining why Kamailio is messing with the Video portion of the call. I have included a full pcap file. As a side note when comparing pcap traces of the Non Proxied vs Proxied call, the INVITE and 200 OK from Asterisk and Softphone are exactly the same. Thank you, -Steve

1 0

Combining lookup_branches() and t_(next|load)_contacts()
by Nathan Ward 09 Jun '18

09 Jun '18

Hi, About a year ago I posted this, but haven’t had much chance to follow up on it: https://lists.kamailio.org/pipermail/sr-users/2017-March/096337.html I’ve been doing some work on this issue (we worked around it until now), and have reduced the config right down to see where things are going wonky. The scenario I have, is inbound calls (to let’s say “A@domain") which I have aliased to two or more different numbers (let’s say “B@domain” and “C@domain”). Both B and C are registering, but are not always registered. I resolve the aliases with alias_db_lookup() (in my sample below, I use seturi() and append_branch() to avoid any confusion about where things may be going weird), then look them up in the location table with lookup_branches(). I then need to do t_load_contacts() and t_next_contacts(), as in some call scenarios I add a serial fork before or after the main call (pre-call announcements etc.). Then, I call t_relay(). This works great most of the time - however, if for example B is registered, but C is not registered, lookup_branches() notes that B is "Not found in usrloc”, and then t_load_contacts() seems to not realise that this isn’t usable, and it attempts to do a DNS lookup for the domain and so on, which I don’t really want it to be doing. Should lookup_branches() be removing a branch it can’t resolve? I would expect it to behave like that, anyway. Can I configure it to do that, or do I need to manually loop through them or something to make sure they resolved, and remove the ones that didn’t? I’ve tried swapping it around, so I call lookup_branches() after t_next_contacts() but the behaviour is the same. I imagine I can’t be the only person trying to do this - how have others made it work? Config sample: seturi('sip:a@domain'); append_branch('sip:b@domain'); xlog("L_DBG", "1 ds: [$ds]\n"); lookup_branches('location'); xlog("L_DBG", "2 ds: [$ds]\n"); t_load_contacts(); xlog("L_DBG", "3 ds: [$ds]\n"); t_next_contacts(); xlog("L_DBG", "4 ds: [$ds]\n"); -- Nathan Ward

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Peculiar Kamailio Asterisk behavior on outbound calls
by Wilkins, Steve 09 Jun '18

09 Jun '18

Hi All, Issue: when a Call is made through Kamailio and Asterisk. Asterisk uses incorrect Video RTP Payload Type when sending Video packets. I have a situation where I make a call from Device A to Device B and Device A is Registered in Kamailio. When Device A Calls Device B, Kamailio sends an 'INVITE' to Asterisk, Asterisk then 'INVITES' Device B. I get two-way Audio, the call stays connected, however, when Video packets are sent to Device B, the RTP Payload Type is incorrect. The port is correct, but just not the Payload Type. Here is where I think Kamailio is involved. In the first Invite from Kamailio to Asterisk, one of the offered Video codecs is '100 H264'; interesting enough, Device B wants to use '115 H264' and when Asterisk sends out Video packets, it is using '100' instead of '115', and of course I have no Video. I don't know if this is just a coincidence but it sure seems like that is where the issue may lie. Has anyone ever seen this behavior? The Asterisk teams does not think it's an Asterisk issue. Thank you, -Steve

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Randomizing t_load_contacts() with same q value?
by Daniel Tryba 08 Jun '18

08 Jun '18

I'm looking for a way to randomize the contacts for serial forking if they have the same q value (so instead of doing parallel forking to those targets). Without taking the q value into account I could do something with the defined contacts_avp before calling t_next_contacts(). But I have no idea how to randomize and serialize the branches with the same q value after calling t_next_contacts(). Any ideas?

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Monitoring Dispatcher
by Tim Bowyer 08 Jun '18

08 Jun '18

Hi All, Currently trying to work out the best way to keep track of how many calls are being sent to any given dispatcher destination. I remember being able to do this years ago with the load balancer module in OpenSIPS: opensipsctl fifo lb_list. Dispatching algorithm '10' seems to be the best fit for my requirements, as I'm essentially wanting to send calls to a large quantity of FreeSwitch docker containers acting as fax receiving nodes. Equal weight, just filling them in some balanced fashion. Any hints or ideas would be much appreciated! Kind regards, Tim Bowyer

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removing not update packaging scripts
by Sergey Safarov 07 Jun '18

07 Jun '18

Hello I created PR to remove not updated CentOS packaging files ( https://github.com/kamailio/kamailio/pull/1555/files) Now CentOS packaging is located at https://github.com/kamailio/kamailio/tree/master/pkg/kamailio/obs Is there anyone considers it necessary to leave a legacy packaging scripts for CentOS Sergey

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hot switch debug mode
by Sergey Safarov 06 Jun '18

06 Jun '18

I experienced issue with kamailio after some time of uptime. Could you suggest way to enable debug output (equivalent of -ddd options for kamailio executable) for running daemon? May be using kamcmd? Sergey

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RFC: dispatcher refactoring to use xavp and options for attributes
by Daniel-Constantin Mierla 06 Jun '18

06 Jun '18

Hello, I am starting a discussion here about my plans to do a bit of refactoring to dispatcher module and gather some feedback if people think of other improvements. 1) First is about moving from using avps to xavps. It should simplify at least the config in terms of modparams, but also makes it more coherent in storing the details of destination record for serial forking. Practically, from a bunch of avp parameters, there will be one to set the root xavp name, and the old "avps" will be fields inside the xavp. 2) The second is about the attributes field. I am considering to add the option to load additional attributes from dedicated columns when using database. So besides the attrs column, one can specify via a mod param additional columns to be loaded and added to attributes, like: modparam("dispatcher", "xattrs", "weight,socket,maxload") The attributes string will be the value of "attrs" column concatenated with "weight=val1;socket=val2;maxload=val3" (the valX being the corresponding values in those columns). This should make easier the management of the database records. For the option with a text file, no changes will be done. 3) Anything else that one thinks would be good to add to dispatcher? Cheers, Daniel PS. I wrote to both sr-dev and sr-users, to get the attention of the two communities. However, you can reply only to sr-users (larger community) if you are not on sr-dev, because this discussion is about features of the module, not technical aspects of coding. -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com

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Re: [SR-Users] Install IMS Modules over KAMAILIO 5.1
by John M 06 Jun '18

06 Jun '18

First compile Kamailio 5.1 binaries and start the services with Kamailio 4.x IMS configuration files. You need to remove few of the deprecated modules. Then compare the working configuration with 5.X IMS cfg files. You can use OpenIMS HSS application. Configuration is straight forward. Calls can be tested using Boghe IMS client. Regards, Uts Hello, not much into IMS myself, and I see you somehow sorted out the Kamailio part, but in case it helps, you can find sample configuration files in the source code tree at misc/examples/ims/ As for FHOSS, I am not sure if that project is active/maintained. Maybe other guys here using IMS extensions can comment more. Cheers, Daniel On 05.04.18 17:37, Roi Arias Salve wrote: >>* Hi all. *>> >>* I would like to thank you all the documents you have to install a *>* Kamailio server. It’s quite easy and fast. *>> >>* Currently I’m working on Azure, witha Debian 9 OS, and I’d followed *>* the instructions to install the Kamailio OPENSER, and everything Works *>* perfect. *>> >>* But now I’m stuck on my next step. I would like to install all the IMS *>* components, P-cscf/I-cscf/S-cscf/Fhoss following this document *>* https://www.kamailio.org/wiki/tutorials/ims/installation-howto <https://www.kamailio.org/wiki/tutorials/ims/installation-howto> *>> >>* But, I have no success. Maybe I’ve a mistake on the selection for the *>* Debian OS installed on the server. I say this because of this sentence: *>>* “ *Note: This document is work in progress and yet unfinished!* *>>* This is a short installation howto for Debian Wheezy (stable). *>>* “ *>> >>* So, I asume that maybe here I have a problem with the OS version, I’ll *>* try to summarize where I have problems: *>> >>* 1.- I’ve installed the , P-cscf/I-cscf/S-cscf not a problem. *>> >>* 2.- I am not able to follow this steps related to the repository: *>> >>* First, install the Key for our repository: *>>* wget -O - http://repository.ng-voice.com/PublicKey <http://repository.ng-voice.com/PublicKey> | apt-key add - *>>* Then add the repository to your “/etc/apt/sources.list”: *>>* echo "deb http://repository.ng-voice.com <http://repository.ng-voice.com/> jessie ims rtpproxy" >> *>* /etc/apt/sources.list *>> >>* and also I have problems with this action over the repository *>> >>* Note: You will need to add the “non-free” packages to your repository: *>>* deb http://ftp.de.debian.org/debian <http://ftp.de.debian.org/debian> lenny main non-free *>* deb http://security.debian.org/ <http://security.debian.org/> lenny/updates main non-free *>> >> >> >>* 3.- I decided to install directly the modules from an apt-get *>* install action, because I followed the instructions on the Log created *>* on the Kamailio starting box, where you can find all the steps related *>* to IMS modules, and the suggestion there it is to compile your own *>* modules. On this step I detected a problem related with the dialog_ng *>* module, according to the .log document on the Starting box, you have *>* to include all the modules before you execute the make all instruction: *>> >>* modify modules.lst for included modules and excluded modules (use *>* saved file) *>>* - include: ims_auth ims_charging ims_icscf ims_isc *>* ims_qos ims_registrar_pcscf ims_registrar_scscf ims_usrloc_pcscf *>* ims_usrloc_scscf dialog_ng cdp cdp_avp *>> >>* 4.- After this issue, I tried to install FHOSS, but here I can’t go *>* further, I am not able to compile it, neither doing my own compilation *>* following this step *>> >>* ant compile deploy | tee ant_compile_deploy.txt *>> >>* nor following *>> >>* apt-get install openimscore-fhoss *>> >>* I was no table to get connected to the repository, so I can’t go further. *>> >>* I’m sorry for all this information, can anyone help? Any suggestion? *>> >>* Regards *>>* Roi *>> >>>* <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai… <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…>> *>* Libre de virus. www.avast.com <http://www.avast.com> *>* <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai… <https://www.avast.com/sig-email?utm_medium=email&utm_source=link&utm_campai…>> *>>>>>* _______________________________________________ *>* Kamailio (SER) - Users Mailing List *>* sr-users at lists.kamailio.org <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> *>* https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users> * -- Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - April 16-18, 2018, Berlin - www.asipto.com Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

Re: [SR-Users] ASN.1 HI2- IRI-Parameters
by Mojtaba 06 Jun '18

06 Jun '18

Any idea in this regards? I understood it's structure of HI2Operations as ASN.1 IRI-HI2, I attached it. I think i should convert SIP signalling flow to this structure for transmission. Is it right way that i am doing? With Regards.Mojtaba On Thu, Apr 26, 2018 at 12:50 PM, Mojtaba <mespio(a)gmail.com> wrote: > Hello, > The declaration of details are in 3GPP TS 101-671 Annex D.5 (ASN.1 > description of IRI (HI2 interface)). > I need a sample packet of HI2Operations for underestanding, so that > i'll develop this HI in kamailio. > With Regards.Mojtaba > > > On Thu, Apr 26, 2018 at 11:20 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: >> Hello, >> >> can you provide more details or references (web links) about the operations >> you want to do? Like what sip message comes to kamailio and what you need to >> change to it. >> >> Cheers, >> Daniel >> >> >> On 25.04.18 06:00, Mojtaba wrote: >> >> Hello. >> I need to use HI2Operations for transfering IRI-Parameters in kamailio.Dose >> anybody have experience in this regards? >> Thanks. >> Mojtaba >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users(a)lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> >> -- >> Daniel-Constantin Mierla >> www.twitter.com/miconda -- www.linkedin.com/in/miconda >> Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com > > > > -- > --Mojtaba Esfandiari.S -- --Mojtaba Esfandiari.S

2 2

TOPOS module errors
by Enrico Bandiera 06 Jun '18

06 Jun '18

Hi, I'm trying to use the TOPOS module to solve some MTU issues we have (we have multiple kamailio servers in our architecture and route/via headers are becoming a problem) The module seems to work but I'm getting these errors on the log: 0(94) ERROR: topos [tps_storage.c:480]: tps_db_insert_dialog(): No database handle - misconfiguration? 0(94) ERROR: topos [tps_storage.c:412]: tps_storage_record(): failed to store I get these errors for some time after running kamailio and then they stop appearing. Data is inserted on topos_d and topos_t on the db, so I'm not sure what the error is referring to. Any idea? Thanks, Enrico.

4 14

problems to parse Cseq 0 value
by David Escartin 05 Jun '18

05 Jun '18

hello all we have seen that when an OPTIONS is sent with CSep: 0 OPTIONS, message is not parsed. Jun 4 10:12:09 proxy-1-aws /usr/local/kamailio/sbin/kamailio[2205]: ERROR: <core> [core/parser/parse_cseq.c:61]: parse_cseq(): no method found Jun 4 10:12:09 proxy-1-aws /usr/local/kamailio/sbin/kamailio[2205]: ERROR: <core> [core/parser/parse_cseq.c:92]: parse_cseq(): bad cseq Jun 4 10:12:09 proxy-1-aws /usr/local/kamailio/sbin/kamailio[2205]: ERROR: <core> [core/parser/msg_parser.c:144]: get_hdr_field(): bad cseq Jun 4 10:12:09 proxy-1-aws /usr/local/kamailio/sbin/kamailio[2205]: ERROR: <core> [core/parser/msg_parser.c:331]: parse_headers(): bad header field [CSeq: 0 Jun 4 10:12:09 proxy-1-aws /usr/local/kamailio/sbin/kamailio[2205]: WARNING: <core> [core/receive.c:198]: receive_msg(): parsing relevant headers failed as far i see in rfc3261 "The sequence number value MUST be expressible as a 32-bit unsigned integer and MUST be less than 2**31." i don't know if you already discussed about this, but do you think it would be something to fix? best regards and thanks a lot david escartin

2 4

Kamailio v5.1.4 Released
by Daniel-Constantin Mierla 05 Jun '18

05 Jun '18

Hello, Kamailio SIP Server v5.1.4 stable release is out. This is a maintenance release of the latest stable branch, 5.1, that includes fixes since the release of v5.1.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.1.x. Deployments running previous v5.1.x versions are strongly recommended to be upgraded to v5.1.4. For more details about version 5.1.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/06/kamailio-v5-1-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

WARNING: topos [tps_storage.c:400]: tps_storage_record(): no local address
by Minh Phan 05 Jun '18

05 Jun '18

Hi List, I have this warning message in kamailio log when Kamailio handle an OPTIONS message with topos module enabled (I am using Kamailio 5.1.2): WARNING: topos [tps_storage.c:400]: tps_storage_record(): no local address - do record routing for all initial requests In my config file, OPTIONS message is handled this way: if ( is_method("OPTIONS") ) { t_newtran(); record_route(); t_reply("200","OK"); exit; } I have found this issue on Github but it is closed and I am not sure what was the solution: #1483 <https://github.com/kamailio/kamailio/issues/1483> My server receives 4 OPTIONS messages every 10 seconds that give as many warning in my log file. What could I do to get rid of them? Thanks, Minh

3 5

Releasing Kamailio v5.1.4
by Daniel-Constantin Mierla 05 Jun '18

05 Jun '18

Hello, I am considering to release a new minor version out of branch 5.1, respectively the v5.1.4, sometime next week, likely on Tuesday, June 5, 2018. As usual, if there is any issue not yet reported on github tracker, do it as soon as possible to give it a chance to be fixed. Soon I am going to do the backports of commits with fixes in master that apply in branch 5.1, so keep a close eye and if someone is missing, let me know to push it in the stable branch. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference -- www.kamailioworld.com

3 3

Transaction flags and branch_routes
by Alex Balashov 05 Jun '18

05 Jun '18

Hi, Given: flags PROXY_MEDIA_SET : 10, SIPPING : 11, DUMMY2 : 12 And: request_route { ... route(XYZ); } route[XYZ] { setflag(PROXY_MEDIA_SET); t_on_branch("BRANCH_ROUTE"); if(!t_relay()) sl_reply_error(); } branch_route[BRANCH_ROUTE] { if(isflagset(PROXY_MEDIA_SET)) { ... } ... } Is PROXY_MEDIA_SET meant to be visible inside the branch_route? Because it doesn't seem to be. Thanks! -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/

2 1

ACK to 200 OK without to tag
by Kjeld Flarup 04 Jun '18

04 Jun '18

We have a SIP doorstation Grandstream GDS3710. When I connect it via kamailio, it does not setup call correct. Kamailio sends a 200 OK with this to field: t: <sip:0001@192.168.1.150:5069>;tag=e33a13d3-c18c-4df1-b68f-446e42957de6 Grandstream answers with: To: <sip:0001@192.168.1.150:5069> As the code follows this example: https://kamailio.org/docs/modules/4.3.x/modules/dispatcher.html#dispatcher.…, it seems obvious that this will fail, as there is not route[WITHINDLG] { if (has_totag()) { There simply is not to tag. Is this a bug in the grandstream? If so how could that go unnoticed, or is it me who misses something? I have attached the full sip text for the 200OK and the ACK -- -------------------- Med Liberalistiske Hilsner ---------------------- Civilingeniør, Kjeld Flarup - Mit sind er mere åbent end min tegnebog Sofienlundvej 6B, 7560 Hjerm, Tlf: 40 29 41 49 Den ikke akademiske hjemmeside for liberalismen - www.liberalismen.dk

2 3

Jumping on new TCP connection during a SIP session?
by Abdoul Osséni 04 Jun '18

04 Jun '18

Hello group, The call flow is: useragent (TLS) -> Kamailio -> Sems Server. During the establishment of a call (before the 200OK or during a retransmission of the 200OK issued by Sems Server via Kamailio), if a user agent (TCP) loses its connection (the user agent is on a mobile network for example or the network is not reliable.) and sends a "register" to Kamailio (same source IP address but a different source TCP port), is it possible for Kamailio to take into account this new TCP port with the function t_relay () ? Abdoul

4 4

Module sanity - memory leaks
by Konstantin Polyakov 04 Jun '18

04 Jun '18

Hello, We had memory leak on a production server and to find it we removed optional modules one by one. As result we found that problem is in the sanity module. Unfortunately I don't have time to fix it since I am going to vacation. We used the following checks: ruri_sip_version (1), required_headers (4), cseq_method (32), cseq_value (64), cotent_length (128), expires_value (256), proxy_require (512), parse uri's (1024). So be aware that this module is not ready to use in production. Best regards, Konstantin

1 0

configuring websockts in kamilio
by vinod mn 04 Jun '18

04 Jun '18

hi configured websockets on kamilio 5.1.3 I am trying to register the client based on webrtc client ,(sip.js) the registration is fine but i am not getting the correct address if i run "*kamctl ul show"* * In the address filed I want the sip:<id>@<ip>:<port>ex: sip:12345@127.0.0.1:12345 <http://sip:12345@127.0.0.1:12345> * output shows: { "jsonrpc": "2.0", "result": { "Domains": [{ "Domain": { "Domain": "location", "Size": 1024, "AoRs": [{ "Info": { "AoR": "1xxxxxxxxx", "HashID": 1xxxxx, "Contacts": [{ "Contact": { *"Address": "sip:eg84vjhr@p6ggglrk0gsh.invalid;transport=ws"*, "Expires": 397, "Q": 1, "Call-ID": "ma4feu2i43bnu6ophe4k78", "CSeq": 86, "User-Agent": "SIP.js/0.5.0-devel BB", "Received": "sip:X.X.X.X:56087;transport=ws", "Path": "[not set]", "State": "CS_NEW", "Flags": 0, "CFlags": 192, "Socket": "tcp:X.X.X.X:8080", "Methods": 783, "Ruid": "uloc-5b116457-1dc1-03", "Instance": "<urn:uuid:0e879ebf-0a31-45a8-9fd9-6b937d3328d4>", "Reg-Id": 1, "Server-Id": 0, "Tcpconn-Id": 259, "Keepalive": 1, "Last-Keepalive": 1527876355, "Last-Modified": 1527876355 } }] } } ], "Stats": { "Records": 1, "Max-Slots": 1 } } }] }, "id": 18958 } In the address filed I want the sip:<id>@<ip>:<port> ex: sip:12345@127.0.0.1:12345 Please give me any suggestions on this, which webrtc client can i use to register to kamailio and able to receive on the client and to make outbound calls from the client. -- Thanks and regards Vinod.M.N

2 1

Kamailio Behind TCP Load Balancer
by Colin Morelli 04 Jun '18

04 Jun '18

Hey all, I'm looking to put Kamailio behind a TCP load balancer that is SIP-unaware. My application is deployed in AWS and I'm tying to place Kamailio behind an ELB. For the most part, everything is fine. For my specific implementation I'm disabling UDP as a signaling transport and using only TLS. This enables me to not have to worry about a SIP-aware LB at the edge because replies to an incoming request will be sent over the existing established TCP socket (avoiding any crazy routing requirements). However - this poses an issue with source addresses. Does Kamailio support anything like the proxy protocol ( http://www.haproxy.org/download/1.5/doc/proxy-protocol.txt) for getting TCP stream information from a load balancer? Do I need to go back to exposing it directly to the world so that I can get source addresses? Thanks, Colin

3 4

Re: [SR-Users] configuring websockts in kamilio
by Konstantin Polyakov 02 Jun '18

02 Jun '18

Hello Vinod, We used Doubango sipML5 web phone https://www.doubango.org/sipml5/call.htm?svn=252 It was quite good for debugging and testing webrtc calls. Most of work in kamailio config I made by using that client. Best regards. Konstantin > >Message: 8 >Date: Fri, 1 Jun 2018 23:50:50 +0530 >From: vinod mn < vinodmn.dbit(a)gmail.com > >To: sr-users(a)lists.kamailio.org >Subject: [SR-Users] configuring websockts in kamilio >Message-ID: >< CAHeWXy7h4Tcp4Sg9_eM3teCHBdJ5bwHa9TYSJy+7P4e+bm1hqQ(a)mail.gmail.com > >Content-Type: text/plain; charset="utf-8" > >hi configured websockets on kamilio 5.1.3 >I am trying to register the client based on webrtc client ,(sip.js) > the registration is fine but i am not getting the correct address if i >run "*kamctl ul show"* > > >* In the address filed I want the sip:<id>@<ip>:<port>ex: >sip:12345@127.0.0.1:12345 < http://sip:12345@127.0.0.1:12345 > * >output shows: >{ > "jsonrpc": "2.0", > "result": { > "Domains": [{ > "Domain": { > "Domain": "location", > "Size": 1024, > "AoRs": [{ > "Info": { > "AoR": "1xxxxxxxxx", > "HashID": 1xxxxx, > "Contacts": [{ > "Contact": { > *"Address": >"sip:eg84vjhr@p6ggglrk0gsh.invalid;transport=ws"*, > "Expires": 397, > "Q": 1, > "Call-ID": "ma4feu2i43bnu6ophe4k78", > "CSeq": 86, > "User-Agent": "SIP.js/0.5.0-devel BB", > "Received": "sip:X.X.X.X:56087;transport=ws", > "Path": "[not set]", > "State": "CS_NEW", > "Flags": 0, > "CFlags": 192, > "Socket": "tcp:X.X.X.X:8080", > "Methods": 783, > "Ruid": "uloc-5b116457-1dc1-03", > "Instance": >"<urn:uuid:0e879ebf-0a31-45a8-9fd9-6b937d3328d4>", > "Reg-Id": 1, > "Server-Id": 0, > "Tcpconn-Id": 259, > "Keepalive": 1, > "Last-Keepalive": 1527876355, > "Last-Modified": 1527876355 > } > }] > } > } > ], > "Stats": { > "Records": 1, > "Max-Slots": 1 > } > } > }] > }, > "id": 18958 >} > >In the address filed I want the sip:<id>@<ip>:<port> >ex: sip:12345@127.0.0.1:12345 > >Please give me any suggestions on this, which webrtc client can i use to >register to kamailio and able to receive on the client and to make outbound >calls from the client. > >-- >Thanks and regards >Vinod.M.N >

1 0

CDRs
by David Villasmil 02 Jun '18

02 Jun '18

Hello guys, I setup the CDRs according to the instructions on https://kb.asipto.com/siremis:install40x:accounting I get CDRs nicely, but the cdrs, even though they have a src_ip, there's no dest_ip, I mean I need the IP where the call was forwarded to... What do i need to do to get that in the cdr? Thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

2 1

Upgrade issue from 4.4.7 to 5.1.3
by Kevin Greene 01 Jun '18

01 Jun '18

Hello List users: I am trying to upgrade to the current version of Kamailio from the older 4.4 version to the current version. The 4.4 is functional but there are some features of 5.1 that we wanted to test with. It seems that my version of Linux (RHEL 6.9 or Oracle 6.9) has an issue with a library file. Any help would be useful. Thank you. Kevin Notes: I used the "yum update kamailio" function. (edited for clarity - it showed a long list of modules to be updated) "--> Finished Dependency Resolution Error: Package: kamailio-http-utils-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms) Requires: kamailio = 4.4.7 Removing: kamailio-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms) kamailio = 4.4.7-1.1 Updated By: kamailio-5.1.3-1.el6.x86_64 (home_kamailio_v5.1.x-rpms) kamailio = 5.1.3-1.el6 Error: Package: kamailio-http-utils-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms) Requires: libkmi.so.1()(64bit) Removing: kamailio-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms) libkmi.so.1()(64bit) Updated By: kamailio-5.1.3-1.el6.x86_64 (home_kamailio_v5.1.x-rpms) Not found Error: Package: kamailio-xml-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms) Requires: kamailio = 4.4.7 Removing: kamailio-4.4.7-1.1.x86_64 (@home_kamailio_v4.4.x-rpms) kamailio = 4.4.7-1.1 Updated By: kamailio-5.1.3-1.el6.x86_64 (home_kamailio_v5.1.x-rpms) kamailio = 5.1.3-1.el6 You could try using --skip-broken to work around the problem You could try running: rpm -Va --nofiles --nodigest" However the "missing" library is there in /usr/lib64. [root@sipserver tmp]# find / -name libkmi.so.1 /usr/lib64/kamailio/libkmi.so.1 /usr/lib64/libkmi.so.1 /lib64/libkmi.so.1 To provide additional help, See: [root@sipserver tmp]# kamailio -version version: kamailio 4.4.7 (x86_64/linux) 35d8ab flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, 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: 35d8ab compiled on 15:38:14 Feb 26 2018 with gcc 4.4.7

2 1

configuring the websocket in kamailio 5.1
by vinod mn 01 Jun '18

01 Jun '18

Hi, I configured the websocket in kamailio and it is running properly, which is the best webrtc client (with proper code as I dont't want to spend more time in writing the codes based on the example snippets ) can I use for registering, and making/ receiving the calls on the webrtc-client . requirement: webrtc-client(??)->register-> kamailio(already configured to use websocket) webrtc-client -> makecall to kamailio webrtc-client receives call from another client(sip not webrtc-client) -- Thanks and regards Vinod.M.N

1 0

Thank You
by Wilkins, Steve 01 Jun '18

01 Jun '18

Hi All, I normally get on this board to ask questions but today I just wanted to take a minute and say - Thank you to all of you who have taken the time to help me. Some of you have spent a lot of time schooling me on Kamailio, its Interactions with a PBX and different clients. I have learned tons from this board and just want to say again, Thank You All! -Steve

2 1

Multiple CDR rate groups using Siremis sql procedures
by Sergiu Pojoga 01 Jun '18

01 Jun '18

Hi, The SQL procedure takes a param, rgroup, which is default (by default). That works fine. In a real case scenario, having multiple rate groups, I wonder what's the best approach to rate CDRs based on the rate group a client belongs to? Do you store the rate_group in the ACC table and then adjust 'kamailio_cdrs' and 'kamailio_rating' procedures to take into account the rate_group of a CDR? If so, does anybody have a working example? Thanks.

1 0

Anybody using 2 instances with topos communicating with each other?
by Daniel Tryba 01 Jun '18

01 Jun '18

Scenario: endpoint (bob) <-> kamailio proxy (5.1.3) <-> kamailio backend (5.1.2) Both kamailio instances are running the topos module (mysql backend). Call setup is fine, but forwarding BYEs fail. With a call from endpoint to kamailio, topos on the proxy scrambles Contact to kamailio backend. But in responses from backend contact remains untouched. In the ACK to the 200, topos on the proxy does nothing. If the caller hangsup, the BYE will loop internally till max hops is exceeded (483 which will be send to caller eventually). When the backend initiates the call to the endpoint, topos appears to work, but the BYE from the endpoint on hangup get a 404 Not here from the proxy (route[WITHINDLG]). Disabling topos on the proxy will result in a functioning proxy (config attached). Is there something wrong with my config? Or is this an unforseen scenario that just doesn't work with topos? * Endpoint to backend scenario endpoint:5060 -> proxy:5060 INVITE sip:callee@sip.pocos.nl;transport=tcp SIP/2.0 Via: SIP/2.0/UDP endpoint:5060;rport;branch=... Max-Forwards: 70 Route: <sip:proxy:5060;lr> Contact: <sip:bob@endpoint:5060;ob> proxy:5060 -> backend:5060 INVITE sip:callee@sip.itco.nl;transport=tcp SIP/2.0 Via: SIP/2.0/UDP proxy;branch=... CSeq: 29848 INVITE Route: <sip:proxy:5060;lr> Contact: <sip:btpsh-5af5972d-bf5e-2@proxy> backend:5060 -> proxy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy;branch=... Contact: <sip:atpsh-5af19bf0-9f94-7@backend> proxy:5060 -> endpoint:5060 SIP/2.0 200 OK Contact: <sip:atpsh-5af19bf0-9f94-7@backend> Via: SIP/2.0/UDP endpoint:5060;received=endpoint;rport=5060;branch=... Record-Route: <sip:proxy;lr> endpoint:5060 -> proxy:5060 ACK sip:atpsh-5af19bf0-9f94-7@backend SIP/2.0 Via: SIP/2.0/UDP endpoint:5060;rport;branch=... Route: <sip:proxy;lr> proxy:5060 -> backend:5060 ACK sip:atpsh-5af19bf0-9f94-7@backend SIP/2.0 Via: SIP/2.0/UDP proxy;branch=... Via: SIP/2.0/UDP endpoint:5060;received=endpoint;rport=5060;branch=... endpoint:5060 -> proxy:5060 BYE sip:atpsh-5af19bf0-9f94-7@backend SIP/2.0 Via: SIP/2.0/UDP endpoint:5060;rport;branch=... Max-Forwards: 70 Route: <sip:proxy;lr> proxy:5060 -> proxy:5060 BYE sip:atpsh-5af19bf0-9f94-7@backend SIP/2.0 Via: SIP/2.0/UDP proxy;branch=... Via: SIP/2.0/UDP endpoint:5060;received=endpoint;rport=5060;branch=... Max-Forwards: 69 Route: <sip:proxy;lr> proxy:5060 -> proxy:5060 BYE sip:atpsh-5af19bf0-9f94-7@backend SIP/2.0 Via: SIP/2.0/UDP proxy;branch=... Via: SIP/2.0/UDP proxy;rport=5060;branch=... Via: SIP/2.0/UDP endpoint:5060;received=endpoint;rport=5060;branch=... Max-Forwards: 68 Route: <sip:proxy;lr> [not going to mention the 50+ BYEs adding Vias and decrementing Max-Forwards. * Backend to endpoint scenario backend:5060 -> proxy:5060 INVITE sip:bob@endpoint:5060;ob SIP/2.0 Via: SIP/2.0/UDP backend;branch=... Route: <sip:loadbalancer@proxy;lr;received=sip:endpoint:5060> Contact: <sip:btpsh-5af07a27-72b-502c1@backend> proxy:5060 -> endpoint:5060 INVITE sip:bob@endpoint:5060;ob SIP/2.0 Via: SIP/2.0/UDP proxy;branch=... Contact: <sip:btpsh-5af5972d-bf5d-5@proxy> endpoint:5060 -> proxy:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP proxy;received=proxy;branch=... Contact: <sip:bob@endpoint:5060;ob> proxy:5060 -> backend:5060 SIP/2.0 200 OK Contact: <sip:bob@endpoint:5060;ob;alias=endpoint~5060~1> Via: SIP/2.0/UDP backend;rport=5060;branch=... Record-Route: <sip:proxy;lr> endpoint:5060 -> proxy:5060 BYE sip:btpsh-5af5972d-bf5d-5@proxy SIP/2.0 Via: SIP/2.0/UDP endpoint:5060;rport;branch=... proxy:5060 -> endpoint:5060 SIP/2.0 404 Not here Via: SIP/2.0/UDP endpoint:5060;rport=5060;branch=...

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Dispatcher module ERROR
by Aleksandar Sosic 01 Jun '18

01 Jun '18

Hi everyone, We have an error using the dispatcher module. In the logs: ```0(107) ERROR: dispatcher [dispatcher.c:815]: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled``` We are running 5.2.0~dev5+0~20180523010152.1118+xenial The configuration file parts regarding dispatcher are as follows: ``` loadmodule "dispatcher.so" ... # ----- dispatcher params ----- modparam("dispatcher", "list_file", "/tmp/dispatcher.list") # Dispatcher: Enable Failover-Support modparam("dispatcher", "flags", 2) # Dispatcher: Overwrite Destination address, if required. modparam("dispatcher", "force_dst", 1) # AVP's required for Fail-Over-Support: modparam("dispatcher", "dst_avp", "$avp(DISPATCHER_DST_AVP)") modparam("dispatcher", "grp_avp", "$avp(DISPATCHER_GRP_AVP)") modparam("dispatcher", "cnt_avp", "$avp(DISPATCHER_CNT_AVP)") # Try to recover disabled destinations every 15 seconds. modparam("dispatcher", "ds_ping_interval", 15) # Actively query the gateways: modparam("dispatcher", "ds_probing_mode",0)# ----- dispatcher params ----- modparam("dispatcher", "list_file", "/tmp/dispatcher.list") # Dispatcher: Enable Failover-Support modparam("dispatcher", "flags", 2) # Dispatcher: Overwrite Destination address, if required. modparam("dispatcher", "force_dst", 1) # AVP's required for Fail-Over-Support: modparam("dispatcher", "dst_avp", "$avp(DISPATCHER_DST_AVP)") modparam("dispatcher", "grp_avp", "$avp(DISPATCHER_GRP_AVP)") modparam("dispatcher", "cnt_avp", "$avp(DISPATCHER_CNT_AVP)") # Try to recover disabled destinations every 15 seconds. modparam("dispatcher", "ds_ping_interval", 15) # Actively query the gateways: modparam("dispatcher", "ds_probing_mode",0)``` What are we doing wrong? What are we missing? Thanks, -- Aleksandar Sosic linkedin.com/in/alexsosic/ mail: alex.sosic(a)evosip.cloud

2 2

Authenticate against outbound carrier
by Krzysztof Grobelak 01 Jun '18

01 Jun '18

Hello List, I have a working setup with a kamailio + freeswitch where i can register extension and make succesfull calls between them. I added now a "external" path for outbound call to which i can successfully route using the dialplan + dispatcher combo. My problem is that my outbound path requires authentication and i'm struggling to find a way to successfully authenticate with the remote end. I configured the UAC module to register against the remote end with a set of credentials but in the SIP trace i can see that the kamailio simply uses the local user credentials instead of the external ones. I believe that this is because the 401 Unauthorized from the carrier gets proxied to the extension which responds with its own credentials. Can anybody suggest how can i stop forwarding them in this case to extension and instead use the UAC to reply with different set of credentials? Regards, Krzysztof The config: request_route { ... if($rU=~"[0-9]{5,}"){ xlog("L_INFO","Looking up route for $rU\n"); dp_match("1", "$rU"); $var(dsid) = $(var(attrs){s.int}); if(!ds_select_dst("$var(dsid)", "4")) { send_reply("404", "No destination"); exit; } if(!ds_select_domain("$var(dsid)", "4")) { send_reply("404", "No destination"); exit; } xlog("L_INFO", "Looking up $fn in database\n"); $avp(s:auth_realm_avp) = "realm.com"; $avp(s:auth_username_avp) = "username"; $avp(s:auth_password_avp) = "password"; if(uac_reg_request_to("$fn", 1)) { xlog("L_INFO", "Found remote user [$rU] on [$rd] via [$du]"); t_on_failure("REMOTE_AUTH"); xlog("L_INFO", "going to <$ru> via <$du>\n"); route(RELAY); } exit; } failure_route[REMOTE_AUTH] { if ($T_reply_code == 401 or $T_reply_code == 407) { xlog("L_INFO", "Remote asked for authentication"); uac_auth(); } } -- PGP: 9E5D E635 5A03 485C 2E00 4530 EB83 920A 9E21 FBA8

1 0

Acc module and PostgreSQL
by Javohir Maxmarajabov 01 Jun '18

01 Jun '18

Is it possible to enable accounting module with PostgreSQL? All I found is documentation and examples that showing accounting module is used with MySQL module. -- Javokhir M.M. ________________________________________________ Disclaimer: The information in this e-mail is confidential. If you are not addressed recipient then please return and delete this e-mail from your system. Unauthorised use of or disclose the contents of this e-mail may be unlawful.

2 1

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