- sr-users - kamailio.org

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

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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

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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

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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

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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

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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

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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

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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.

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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

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