- sr-users - kamailio.org

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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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

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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

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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

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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/

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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

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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

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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

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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

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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=...

2 2

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

Strange behavior with $shtinc
by Brooks Bridges 01 Jun '18

01 Jun '18

Let me preface this with the statement that as best I can tell from docs and code, this function is supposed to be 100% atomic and this shouldn't be able to happen. I also have not found anything in the module's github commit logs that would indicate this behavior has changed from previous versions. That being said, I appear to have run into an oddity where intermittently I'm getting 2 slots being put in the same entry, which results in trying to parse the output for graphing and reporting not working quite right. The mechanism I'm using is a very simple counter mechanism to record statistics about calls. As an example, I am using the following mechanism: $var(z) = $shtinc(callstats=>$avp(realm),cps_exceeded); or $var(z) = $shtinc(callstats=>$avp(realm),sessions_exceeded); This typically results in something like this: { entry: 4335 size: 1 slot: { { name: realm1,total_requests value: 2365034 type: int } } } { entry: 4532 size: 1 slot: { { name: realm2,cps_exceeded value: 30 type: int } } } And so on... However, one or more of them sometimes end up like this: { entry: 4646 size: 2 slot: { { name: realm1,total_requests value: 15958026 type: int } { name: realm2,cps_exceeded value: 6432103 type: int } } } Thanks in advance for your help and/or suggestions! Brooks Bridges | Engineering Manager O1 Communications 4359 Town Center Boulevard, Suite 217 El Dorado Hills, California 95762 office: 916.235.2097 | main: 888.444.1111, Option 2 email: bbridges(a)o1.com<mailto:bbridges@o1.com> | web: www.o1.com<http://www.o1.com/>

3 5

Authentication against outbound carrier trunk
by Krzysztof Grobelak 31 May '18

31 May '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

error in tls configuration in kamailio 5.1.3
by Narasimhaiah, Vinod Maralenahalli 31 May '18

31 May '18

Hi, I upgraded kamailio 4.4.2 to kamailio5.1.3 and i am getting this error *"Unable to load certificate file '/etc/kamailio/kamailio-selfsigned.pem",* kamilio 4.4.2 does not need the ssl file, but kamailio 5.1.3 need ssl file, what changes required in kamailio 5.1.3 to run with tls. Please help me in resolving this issue. Thank you vinod.M.N -- To download the app, click: iOS <http://goo.gl/ZJx3y1> | Android <http://goo.gl/G1ehjZ> -- ***This transmission contains information that is confidential and proprietary of Kirusa Inc.,its affiliates and/or its partners and customers. This information is intended to be shared only with the recipient of this e-mail who is bound by confidentiality. If you received this e-mail in error, please delete the e-mail and notify the sender by return e-mail.***

4 4

Upgrade issue with old Kamailio 3.0.3 in table rls_presentity
by Kristijan Vrban 31 May '18

31 May '18

Hi, i try a upgrade from old 3.0.3 Kamailio mysql db. Got stuck with update the structure : ALTER TABLE rls_presentity MODIFY callid VARCHAR(255) NOT NULL;ALTER TABLE rls_presentity MODIFY contact VARCHAR(128) NOT NULL; found on: https://www.kamailio.org/wiki/install/upgrade/3.1.x-to-3.2.0 But this two table does not exist? See also table definition at: https://kamailio.org/docs/db-tables/kamailio-db-3.1.x.html#GEN-DB-RLS-PRESE… I reckon it is safe to skip this to tables? Kristijan

2 4

Register on kamailio.org wiki page not working
by Kristijan Vrban 31 May '18

31 May '18

Hi, https://www.kamailio.org/wiki/start?do=register Is not working. You get a empty page here. Kristijan

2 2

Sharing static variables in external library across child proceses
by User 721 31 May '18

31 May '18

Hi, I am adding a kamailio module that references an external C library that maintains a hashtable in global scope. I am wondering what practice I should follow so that: - This hashtable (which I don't control directly) is shared by all child processes - The hashtable is changed by one child process at a time Thanks in advance

2 1

PIKE module
by Robert Barrera 31 May '18

31 May '18

Good morning! We operate a VoIP platform and are interested in integrating the PIKE module. Is it possible for someone to assist us to discuss how this can be done and our options? Thank you logo_nop Robert Barrera 305-663-8989 305-600-4786 fax info(a)myrealmobile.com Complete Business PBX phone systems and Nationwide wireless! REAL Mobile does it!

2 1

sip_msg structure and memory free
by Konstantin Polyakov 31 May '18

31 May '18

Hello All, There is a structure sip_msg which is passed into a module and represents a received sip message. It contains number of pointers to hdr_field members. In its turn hdr_field contains member void* parsed In the sanity module we assign to parsed member a pointer to newly allocated memory and we don't free that memory at the end. Do we need to free explicitly or this memory will be deallocated later together with sip_msg ? In other words is it memory leak or not? Best regards, Konstantin

2 1

General Kamailio with Asterisk (or another PBX) Opinion
by Wilkins, Steve 30 May '18

30 May '18

HI All, I am curious, for those of you who use Kamailio with Asterisk or another PBX. Who forwards Registrations to Asterisk or PBX, and who lets Kamailio maintain Registrations? Thanks All!

4 3

send CANCEL to caller after hitting fr_inv_timeout
by Sergiu Pojoga 29 May '18

29 May '18

TM's noisy_ctimer=1 seems to take care of sending a CANCEL to the callee, but it eludes me which tm param makes a CANCEL to be sent to caller as well? Caller gets a '408 Request Timeout' only, which in my case doesn't effectively terminate a call for the calling party. Merci.

1 0

INFO: stun_parse_header: incomplete header of STUN message
by Pan Christensen 29 May '18

29 May '18

Hello! After proxying a REGISTER from a WebRTC client to our SIP backend, the backend starts sending keepalives. Every 30 seconds I get a log entry in Kamailio saying: "INFO: stun [kam_stun.c:169]: stun_parse_header(): INFO: stun_parse_header: incomplete header of STUN message". Here's one such packet: No. Time Source Destination Protocol Length Info 1 0.000000 xxx.xxx.xxx.xxx yyy.yyy.yyy.yyy UDP 60 5060 ? 15060 Len=4 Frame 1: 60 bytes on wire (480 bits), 60 bytes captured (480 bits) Ethernet II, Src: JuniperN_4f:3f:f0 (40:a6:77:4f:3f:f0), Dst: 02:81:64:58:cf:d1 (02:81:64:58:cf:d1) Internet Protocol Version 4, Src: xxx.xxx.xxx.xxx, Dst: yyy.yyy.yyy.yyy 0100 .... = Version: 4 .... 0101 = Header Length: 20 bytes (5) Differentiated Services Field: 0x00 (DSCP: CS0, ECN: Not-ECT) Total Length: 32 Identification: 0x5bf5 (23541) Flags: 0x4000, Don't fragment Time to live: 56 Protocol: UDP (17) Header checksum: 0x2fc3 [validation disabled] [Header checksum status: Unverified] Source: xxx.xxx.xxx.xxx Destination: yyy.yyy.yyy.yyy User Datagram Protocol, Src Port: 5060, Dst Port: 15060 Source Port: 5060 Destination Port: 15060 Length: 12 Checksum: 0xfa28 [unverified] [Checksum Status: Unverified] [Stream index: 0] Data (4 bytes) 0000 00 00 00 00 .... Data: 00000000 [Length: 4] I found this in kam_stun.c: static int stun_parse_header(struct stun_msg* req, USHORT_T* error_code) { if (sizeof(req->hdr) > req->msg.buf.len) { /* the received message does not contain whole header */ LOG(L_INFO, "INFO: stun_parse_header: incomplete header of STUN message\n"); /* Any better solution? IMHO it's not possible to send error response * because the transaction ID is not available. */ return FATAL_ERROR; } ... Could someone please explain what's wrong? Med vennlig hilsen Pan B. Christensen Utvikler Phonect AS Brugata 19, PB 9156 Grønland, N-0133 Oslo, Norway E-post: pan.christensen(a)phonect.no<mailto:pan.christensen@phonect.no> Mobil: 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…>

2 3

Dialplan module for load balancing
by igor.potjevlesch＠gmail.com 29 May '18

29 May '18

Hello, I'm maybe confused on how to use dialplan module. I have a dialplan setup for a long time now and I'd like to add a new rule with the same match_exp than an existing one but with a different repl_exp. If I set the same priority, the latest entry is used systematically. If I set priority +1, the new rule is never used. The goal is having two rules and alternatively reply with one or the other repl_exp. Is it possible? Regards, Igor.

1 0

Pkg memory stats
by Konstantin Polyakov 29 May '18

29 May '18

Hello, Could somebody explain what do numbers mean for command "kamcmd mod.stats all pkg"? Is there somewhere documentation about that? I can see the following information. " Module: xlog { mod_init(213): 4104 xlog_fixup_helper(533): 3176 xdbg_fixup_helper(499): 632 Total: 7912 } " I guess that numbers after colon are bytes. But what do numbers in (213) mean? When I use command "kamcmd pkg.stats", I got absolutely different numbers which don't match previous statistics? What is wrong? What is the best way to analyze pkg memory usage? Thank you Best regards Konstantin

1 0

Realtime Extensions SQLITE
by Benjamin Marty 29 May '18

29 May '18

I compiled a fresh Asterisk 13.18 certified with res_config_sqlite. I configured following: extconfig.conf [settings] sippeers => sqlite,asterisk,ast_config res_config_sqlite.conf [general] dbfile => /home/myuser/database.db On Bootup it gives the following error: [May 29 17:07:14] WARNING[32444]: res_config_sqlite.c:1206 realtime_multi_handler: no such table: ast_config Is the sqlite module deprecated or is my configuration wrong? Couldn't really find any documentation on that topic. The table ast_config is definitely present in the sqlite file.

2 1

Serial forking
by igor.potjevlesch＠gmail.com 29 May '18

29 May '18

Hello, I know that the default behaviour of two aliases in kamailio.aliases result in parallel forking. I'm wondering if its possible to do serial forking instead? For example, in case that the first INVITE has no 180/183, or neither 100 Trying, then send the INVITE to the second entry? Regards, Igor.

2 2

Switching form UDP to TCP causes authentication errors
by Wilkins, Steve 29 May '18

29 May '18

Hello All, When I switched from UDP to TCP I started getting Authentication Errors, Asterisk responds to an INVITE via Kamailio with a '401 Unauthorized', but Kamailio does nothing with it. Processing just stops near WITH_BLOCK401407. Shouldn't the 401 be relayed so a new INVITE can be sent? Thank you.

3 4

ims_isc: 3rd party Registration (S-CSCF) and P-Associated-Uri header
by Valentin Christoph 29 May '18

29 May '18

Hi, The function build_p_associated_uri() in module ims_isc filters for TEL URIs. Why are not ALL unbarred IMPUs added, but only TEL URIs? Is this specified in some 3GPP spec? Thanks, Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

1 0

jsonrpc http interface cant recognize method
by Yuriy Gorlichenko 29 May '18

29 May '18

Hi i trying to get htable.dump mytable via jsonrpc request {jsonrpc: "2.0", method: "htable.dump mytable"} { "jsonrpc": "2.0", "error": { "code": 500, "message": "Method Not Found" } } But via kamcmd it wroks perfect Aslo i see that if i will make request withont param (core.version for example) IT works fime How to make request with param?

1 1

Carrierroute Wholesale termination issue
by riaz hasan 29 May '18

29 May '18

Hello, I am facing an issue with kamailio carrierroute.I have a setup where multiple asterisk is behind a kamailio server.I am able to dispatch the load by using carrierroute. Its working perfectly for retail users.But problem is its not working for wholesale users.I am attaching the dump for wholesale client. here is a little descrption for the shark:- 192.151.149.102 is the wholesale client ip. 46.166.139.60:4848 is kamailio server. 46.166.139.60:3628 is asterisk server. when 192.151.149.102 sends invite to 46.166.139.60:4848 [Kamailio] we are getting 401. [dump2.pcap] when 192.151.149.102 sends invite to 46.166.139.60:3628 [Direct Asterisk ] we are getting 200. [dump.pcap] here is the code i am using for load balancing:- route[Carrer] { #!ifdef WITH_CARRIERROUTE if (!cr_route("default", "default", "$rU", "$rU", "call_id")) { xlog("L_ERR", "cr_route failed\n"); exit; } #!endif $avp(s:host)= $rd+":"+$rp; t_on_failure("1"); if (!t_relay()) { sl_reply_error(); }; } failure_route[1] { revert_uri(); #!ifdef WITH_CARRIERROUTE if (!cr_next_domain("default", "default", "$rU", "$avp(s:host)", "$T_reply_code", "$avp(s:domain)")) { xlog("L_ERR", "cr_next_domain failed\n"); exit; } if (!cr_route("default", "default", "$rU", "$rU", "call_id")) { xlog("L_ERR", "cr_route failed\n"); } t_on_failure("1"); route(NASIR); exit; #!endif $avp(s:host)= $rd+":"+$rp; if (!t_relay()) { xlog("L_ERR", "t_relay failed-custom\n"); exit; }; }

1 0

S-SCSF not forwarding SIP packets to AS
by John M 29 May '18

29 May '18

Hi All, My setup consists of Kamailio P-CSCF,I-SCSF , S-CSCF and OpenIMS HSS nodes and I have configured application server details and trigger point on HSS. I am using Boghe IMS client SIP validation. Whenever the client sends a REGISTER, messages are not reaching AS. In S-SCSF log files, I am seeing " isc_match_filter_reg(): No REGISTER match found in filter criteria" and " isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group" error message. Kmailio version: version: kamailio 5.1.3 (x86_64/linux) HSS XML The UserData XML document which is sent to the S-CSCF: <?xml version="1.0" encoding="UTF-8"?><IMSSubscription><PrivateID> alice(a)ims.mnc001.mcc001.3gppnetwork.org </PrivateID><ServiceProfile><PublicIdentity><Identity> sip:alice@ims.mnc001.mcc001.3gppnetwork.org </Identity><Extension><IdentityType>0</IdentityType></Extension></PublicIdentity><InitialFilterCriteria><Priority>0</Priority><TriggerPoint><ConditionTypeCNF>0</ConditionTypeCNF><SPT><ConditionNegated>0</ConditionNegated><Group>0</Group><Method>REGISTER</Method><Extension><RegistrationType>0</RegistrationType></Extension></SPT></TriggerPoint><ApplicationServer><ServerName>sip:192.168.20.10:5060 ;transport=TCP</ServerName><DefaultHandling>0</DefaultHandling></ApplicationServer></InitialFilterCriteria></ServiceProfile></IMSSubscription> SCSF Log-1 0(1541) DEBUG: ims_isc [ims_isc_mod.c:428]: isc_match_filter_reg(): Checking triggers 0(1541) DEBUG: ims_isc [ims_isc_mod.c:432]: isc_match_filter_reg(): Starting triggering 0(1541) DEBUG: ims_isc [ims_isc_mod.c:446]: isc_match_filter_reg(): Orig User before clean: <sip:alice@ims.mnc001.mcc001.3gppnetwork.org> [1] 0(1541) DEBUG: ims_isc [ims_isc_mod.c:448]: isc_match_filter_reg(): Orig User after clean: <sip:alice@ims.mnc001.mcc001.3gppnetwork.org> [1] 0(1541) DEBUG: ims_isc [checker.c:416]: isc_checker_find(): isc_checker_find: starting search 0(1541) DEBUG: ims_isc [checker.c:438]: isc_checker_find(): isc_checker_find(): got a r_public for the user sip:alice@ims.mnc001.mcc001.3gppnetwork.org 0(1541) DEBUG: ims_isc [checker.c:454]: isc_checker_find(): About to try p->s->service_profiles_cnt!! #profiles is 1 0(1541) DEBUG: ims_isc [checker.c:456]: isc_checker_find(): About to try p->s->service_profiles[a].filter_criterai_cnt 0(1541) DEBUG: ims_isc [checker.c:467]: isc_checker_find(): DEBUG ISC: SECOND TIME About to try p->s->service_profiles_cnt!! 0(1541) DEBUG: ims_isc [checker.c:471]: isc_checker_find(): DEBUG ISC : About to try p->s->service_profiles 0(1541) DEBUG: ims_isc [checker.c:474]: isc_checker_find(): DEBUG ISC : About to try public identities 0(1541) DEBUG: ims_isc [checker.c:477]: isc_checker_find(): DEBUG ISC : About to try WPSI 0(1541) DEBUG: ims_isc [checker.c:319]: isc_check_filter_criteria(): ifc_checker_trigger: Starting expression check: 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == PUBLISH ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method REGISTER 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == PUBLISH ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method REGISTER 0(1541) DEBUG: ims_isc [checker.c:333]: isc_check_filter_criteria(): ifc_checker_trigger: - group 1 => 0. 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == SUBSCRIBE ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method REGISTER 0(1541) DEBUG: ims_isc [checker.c:333]: isc_check_filter_criteria(): ifc_checker_trigger: - group 2 => 0. 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == SUBSCRIBE ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method REGISTER 0(1541) DEBUG: ims_isc [checker.c:333]: isc_check_filter_criteria(): ifc_checker_trigger: - group 3 => 0. 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:348]: isc_check_filter_criteria(): ifc_checker_trigger: Check finished => 0 0(1541) DEBUG: ims_isc [ims_isc_mod.c:460]: isc_match_filter_reg(): No REGISTER match found in filter criteria SCSF Log-2 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == PUBLISH ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method SUBSCRIBE 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == PUBLISH ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method SUBSCRIBE 0(1541) DEBUG: ims_isc [checker.c:333]: isc_check_filter_criteria(): ifc_checker_trigger: - group 1 => 0. 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == SUBSCRIBE ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method SUBSCRIBE 0(1541) DEBUG: ims_isc [checker.c:333]: isc_check_filter_criteria(): ifc_checker_trigger: - group 2 => 1. 0(1541) DEBUG: ims_isc [checker.c:252]: isc_check_spt(): ifc_check_spt: SPT type 3 -> Header[Event] %= .*presence.* ? 0(1541) DEBUG: ims_isc [checker.c:88]: isc_check_headers(): isc_check_headers: Looking for Header[Event(22)] .*presence.* 0(1541) DEBUG: ims_isc [checker.c:99]: isc_check_headers(): isc_check_headers: Found Header[Event(22)] message-summary 0(1541) DEBUG: ims_isc [checker.c:99]: isc_check_headers(): isc_check_headers: Found Header[Allow-Events(39)] refer, presence, presence.winfo, xcap-diff, conference 0(1541) DEBUG: ims_isc [checker.c:261]: isc_check_spt(): ifc_check_spt: SPT type 4 -> Session Case == 1 ? 0(1541) DEBUG: ims_isc [checker.c:263]: isc_check_spt(): ifc_check_spt: Found session_case 0 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:239]: isc_check_spt(): ifc_check_spt: SPT type 2 -> Method == SUBSCRIBE ? 0(1541) DEBUG: ims_isc [checker.c:241]: isc_check_spt(): ifc_check_spt: Found method SUBSCRIBE 0(1541) DEBUG: ims_isc [checker.c:333]: isc_check_filter_criteria(): ifc_checker_trigger: - group 3 => 1. 0(1541) DEBUG: ims_isc [checker.c:252]: isc_check_spt(): ifc_check_spt: SPT type 3 -> Header[Event] %= .*presence.* ? 0(1541) DEBUG: ims_isc [checker.c:88]: isc_check_headers(): isc_check_headers: Looking for Header[Event(22)] .*presence.* 0(1541) DEBUG: ims_isc [checker.c:99]: isc_check_headers(): isc_check_headers: Found Header[Event(22)] message-summary 0(1541) DEBUG: ims_isc [checker.c:99]: isc_check_headers(): isc_check_headers: Found Header[Allow-Events(39)] refer, presence, presence.winfo, xcap-diff, conference 0(1541) DEBUG: ims_isc [checker.c:261]: isc_check_spt(): ifc_check_spt: SPT type 4 -> Session Case == 2 ? 0(1541) DEBUG: ims_isc [checker.c:263]: isc_check_spt(): ifc_check_spt: Found session_case 0 0(1541) DEBUG: ims_isc [checker.c:340]: isc_check_filter_criteria(): ifc_checker_trigger: - group compromised, skipping to next group 0(1541) DEBUG: ims_isc [checker.c:348]: isc_check_filter_criteria(): ifc_checker_trigger: Check finished => 0 Regards, Uts

1 0

Kamailio 5.1.x - Dialog event route not fired
by Loic Chabert 28 May '18

28 May '18

Hi list, I'm facing a strange behaviour with dialog module and stop event route. When a BYE request has been receved by Kamailio, no stop event route has been fired but dialog state go to 5. My config is following this guide: http://kb.asipto.com/kamailio:presence:k43-blf Using dialog_manage on RELAY route, as describe on this guide. If i set more debug information, i can see dialog has been found and dialog status incremented : 1(7) DEBUG: <core> [core/udp_server.c:491]: udp_rcv_loop(): received on udp socket: (106/100/818) [[BYE sip:172.16.74.201:5060;prxroute=1 SIP/2.0 0D 0A Record-Route: <sip:172.16.74.24;r2=on;lr=on;ftag=1;ng]] 1(7) DEBUG: <core> [core/parser/msg_parser.c:604]: parse_msg(): SIP Request: 1(7) DEBUG: <core> [core/parser/msg_parser.c:606]: parse_msg(): method: <BYE> 1(7) DEBUG: <core> [core/parser/msg_parser.c:608]: parse_msg(): uri: <sip:172.16.74.201:5060;prxroute=1> 1(7) DEBUG: <core> [core/parser/msg_parser.c:610]: parse_msg(): version: <SIP/2.0> 1(7) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bKc2c.939fa953ea648059586a7f684fd56b5a.0>; state=16 1(7) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 1(7) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=2 1(7) DEBUG: <core> [core/parser/msg_parser.c:494]: parse_headers(): this is the first via 1(7) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 235, <rport> = <5060>; state=6 1(7) DEBUG: <core> [core/parser/parse_via.c:1303]: parse_via_param(): Found param type 232, <branch> = <z9hG4bK-69-1-10>; state=16 1(7) DEBUG: <core> [core/parser/parse_via.c:2639]: parse_via(): end of header reached, state=5 1(7) DEBUG: <core> [core/parser/msg_parser.c:492]: parse_headers(): Via found, flags=78 1(7) DEBUG: <core> [core/parser/msg_parser.c:505]: parse_headers(): this is the second via 1(7) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=f651e13a-0aac-40d5-b877-1c24497ff87c 1(7) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 1(7) DEBUG: <core> [core/parser/msg_parser.c:171]: get_hdr_field(): <To> [87]; uri=[sip:+33975181257@sip.testinfratrunk.vox.fr] 1(7) DEBUG: <core> [core/parser/msg_parser.c:172]: get_hdr_field(): to body [<sip:+33975181257@sip.testinfratrunk.vox.fr>] 1(7) DEBUG: <core> [core/parser/msg_parser.c:152]: get_hdr_field(): cseq <CSeq>: <3> <BYE> 1(7) DEBUG: <core> [core/receive.c:232]: receive_msg(): --- received sip message - request - call-id: [1-69(a)60.0.0.30] - cseq: [3 BYE] 1(7) DEBUG: <core> [core/receive.c:287]: receive_msg(): preparing to run routing scripts... 1(7) DEBUG: maxfwd [mf_funcs.c:74]: is_maxfwd_present(): value = 69 1(7) DEBUG: <core> [core/parser/parse_addr_spec.c:185]: parse_to_param(): add param: tag=1 1(7) DEBUG: <core> [core/parser/parse_addr_spec.c:864]: parse_addr_spec(): end of header reached, state=29 1(7) DEBUG: <core> [core/parser/msg_parser.c:183]: get_hdr_field(): content_length=0 1(7) DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header 1(7) DEBUG: sanity [sanity_mod.c:256]: w_sanity_check(): sanity checks result: 1 1(7) NOTICE: <script>: {1 3 BYE 1-69(a)60.0.0.30} Initialisation des variables. 1(7) DEBUG: tm [t_lookup.c:1018]: t_check_msg(): msg (0x7ffa420727d8) id=2 global id=1 T start=(nil) 1(7) DEBUG: tm [t_lookup.c:476]: t_lookup_request(): start searching: hash=3116, isACK=0 1(7) DEBUG: tm [t_lookup.c:435]: matching_3261(): RFC3261 transaction matching failed 1(7) DEBUG: tm [t_lookup.c:659]: t_lookup_request(): no transaction found 1(7) DEBUG: tm [t_lookup.c:1087]: t_check_msg(): msg (0x7ffa420727d8) id=2 global id=2 T end=(nil) 1(7) DEBUG: siputils [checks.c:123]: has_totag(): totag found 1(7) DEBUG: rr [loose.c:113]: find_first_route(): No Route headers found 1(7) DEBUG: rr [loose.c:960]: loose_route(): There is no Route HF 1(7) DEBUG: tm [t_lookup.c:1018]: t_check_msg(): msg (0x7ffa420727d8) id=2 global id=2 T start=(nil) 1(7) DEBUG: tm [t_lookup.c:1094]: t_check_msg(): T previously sought and not found for msg (0x7ffa420727d8) 1(7) DEBUG: dialog [dlg_hash.c:768]: internal_get_dlg(): ref dlg 0x7ffa327dd588 with 1 -> 3 1(7) DEBUG: dialog [dlg_hash.c:770]: internal_get_dlg(): dialog callid=' 1-69(a)60.0.0.30' found on entry 3063, dir=1 1(7) DEBUG: dialog [dlg_profile.c:536]: set_current_dialog(): setting current dialog [3063:9173] 1(7) DEBUG: dialog [dlg_hash.c:1151]: next_state_dlg(): dialog 0x7ffa327dd588 changed from state 4 to state 5, due event 7 (ref 3) 1(7) DEBUG: dialog [dlg_hash.c:932]: dlg_ref_helper(): ref op on 0x7ffa327dd588 with 1 from dlg_handlers.c:1389 1(7) DEBUG: dialog [dlg_hash.c:936]: dlg_ref_helper(): ref dlg 0x7ffa327dd588 with 1 -> 4 1(7) DEBUG: dialog [dlg_handlers.c:1400]: dlg_onroute(): BYE successfully processed Do you have any hints concerning this issue ? Is it a bug or a config part is missing ? Thanks in advance, Best regards.

2 1

Loop Detect - 482 Merged Request
by Amar Tinawi 28 May '18

28 May '18

Hello All i'm facing strange behavior with Zoiper Client only in an IMS Kamailio based system. in IMS call scenario (ims to ims) a lot of Loop Detect Msgs Appeared in the sip log. while other clients like Acrobits did not occurred i'm using TCP protocol and Wireshark shows a lot of (TCP Re-transmission and TCP ACKed unseen Segment) Any Suggestion Thanks !

1 0

Link problem between kamailio and MariaDB
by Ginhoux, Patrick 28 May '18

28 May '18

Hi, At the moment, I'm trying to replace MySQL by MariaDB on my Kamailio server (5.0.1) . In previous post I have been told that Kamailio supports MariaDB and there is no special module to load. This work is done on a new server running RHEL 7.4 and MariaDB 10.2.14 that is installed under /opt/mysql/na/10.2.14 folder. I encounter different problem: - Without changing anything in the kamailio script, kamailio failed to find the libmysqlclient.so.18 library. - So I thought that I could get rid of the above error by changing the MODULEPATH in the Kamailio-local.cfg as follows : MODULEPATH="/usr/lib64/kamailio/modules/:/opt/mysql/na/10.2.14/lib64/" But with this, I get 2 problem: The kamailio.log that was written in a /opt/log/Kamailio folder is no longer updated. I can only find the errors in the /var/log/messages file: May 23 11:19:19 vm-siprouter74 systemd: Starting SYSV: Kamailio is a fast, reliable and flexible SIP Server.... May 23 11:19:19 vm-siprouter74 kamailio: Not starting : invalid configuration file! May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.lex:1822]: pp_define_get(): ### returning define ID [DEFINE_LOG_FACILITY] value [LOG_LOCAL0] May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.lex:1822]: pp_define_get(): ### returning define ID [LOCAL_LISTEN] value [udp:129.227.82.109:5060] May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.lex:1822]: pp_define_get(): ### returning define ID [LOCAL_LISTEN_2] value [udp:127.0.0.1:5060] May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.lex:1822]: pp_define_get(): ### returning define ID [MODULEPATH] value ["/usr/lib64/kamailio/modules/:/opt/mysql/na/10.2.14/lib64/"] May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.y:1632]: yyparse(): loading modules under /usr/lib64/kamailio/modules/:/opt/mysql/na/10.2.14/lib64/ May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.y:1613]: yyparse(): loading module db_mysql.so May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/sr_module.c:575]: load_module(): trying to load </usr/lib64/kamailio/modules/db_mysql.so> May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) ERROR: <core> [core/sr_module.c:582]: load_module(): could not open module </usr/lib64/kamailio/modules/db_mysql.so>: libmysqlclient.so.18: cannot open shared object file: No such file or directory May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) CRITICAL: <core> [core/cfg.y:3401]: yyerror_at(): parse error in config file //etc/kamailio/kamailio.cfg, line 191, column 12-24: failed to load module May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/cfg.y:1613]: yyparse(): loading module jsonrpcs.so May 23 11:19:19 vm-siprouter74 kamailio: 0(3328) DEBUG: <core> [core/sr_module.c:575]: load_module(): trying to load </usr/lib64/kamailio/modules/jsonrpcs.so> To work around the problem we have created 2 symbol links: <<< ln -s /opt/mysql/na/10.2.14/lib64/libmysqlclient.so.18.0.0 /usr/lib64/libmysqlclient.so.18 >>> the libmysqlclient error <<< ln -s /myqdata/mysiprdb/data/my.cnf /etc/my.cnf >>> to fix the issue to open the db It seems that kamailio is always looking for the my.cnf in the /etc/ directory. My concern are to share this job with people who have much more experience with Kamailio. I would like to know - If the different workaround (ln) makes sense to you or is there better change to do - How to explain the logging issue when changing the modulepath variable - Is it true that kamailio is always looking for the my.cnf in the /etc/ directory? Any thoughts are welcome. Thanks in advance. Cordialement. Patrick GINHOUX

2 2

Developer Guide
by Igor Olhovskiy 27 May '18

27 May '18

Hi! Is http://www.asipto.com/pub/kamailio-devel-guide/ still holds actual info? Cause it’s mostly for 3.3 version, and now is 5.1 And where to read, except source code of course, about how to develop modules for Kamailio? Regards, Igor

2 1

Compact Headers
by Duarte Rocha 26 May '18

26 May '18

Greetings, Is there any method on Kamailio that automatically rewrites the headers in its compact form ? I know how i can do it by re-writing each of the eligible headers, but i was wondering if there is any way to do it automatically. Thanks

2 1

UAC Module Registration Table
by Greg Adams 25 May '18

25 May '18

Hey there, I am currently using the UAC module to maintain remote registrations. I am just curious if it is possible to register new users immediately rather than waiting for reg_timer_interval? I am not seeing any functionality like that but just wondering if I am missing something or if it would be possible to add in. For example an RPC command that takes a key and instantly registers that user. Thanks, Greg ________________________________

1 0

kamailio.org server maintenance work
by Oriol Capsada 25 May '18

25 May '18

Hello, A kamailio.org server maintenance work will be done today at 18:00 CEST. There will be some minutes of downtime (should be less than 5') of the following main services: website (including the wiki) and mailing lists. Regards, Oriol.

1 0

Kamailio does not receive anything after exit route
by Andrey Deykunov 25 May '18

25 May '18

Hi there, I've added a simple black list that rejects incoming requests (REGISTER and INVITE) containing certain text in the message body. Source IP addresses of the rejected requests are going to be banned and added to the ban list: route > { > ... > > > $var(ban_list_key) = $si + ":" + $sp; > if ($sht(ip_ban_list=>$var(ban_list_key)) != $null) { > xlog("L_INFO", "$var(log_fmt_tag) The incoming $rm request > from: $fU, si: $si, sp: $sp was rejected. The source host is temporarily > banned."); > sl_send_reply("400", "Bad request"); > exit; > } > > ... > > if (method=="REGISTER" || method=="INVITE") { > sht_iterator_start("i1", "black_list"); > while(sht_iterator_next("i1")) { > $var(term_is_in_wl) = 0; > $var(collision_idx) = 0; > while($sht(white_list=>$fU[$var(collision_idx)]) != $null > ) { > if($shtitval(i1) == > $sht(white_list=>$fU[$var(collision_idx)])) { > $var(term_is_in_wl) = 1; > break; > } > $var(collision_idx) = $var(collision_idx) + 1; > } > if ($var(term_is_in_wl) == 0) { > if ( $mb =~ $shtitval(i1) ) { > # found a term from the black list. access denied. > xlog("L_INFO", "$var(log_fmt_tag) The incoming $rm > request from: $fU, si: $si, sp: $sp was rejected. SIP message body contains > a term from the black list - $shtitval(i1)"); > # add device to banlist > $var(ban_info) = "" + $Ts + "_" + $shtitval(i1); > $sht(ip_ban_list=>$var(ban_list_key)) = > $var(ban_info); > xlog("L_INFO", "$var(log_fmt_tag) The source host > $si:$sp was banned for 1 hour."); > sht_iterator_end("i1"); > sl_send_reply("400", "Bad request"); > exit; > } > } > } > sht_iterator_end("i1"); > } > ... > } All works fine with UPD, but using TCP, kamailio ceases to accept any requests. Please take a look, what do I wrong. Thanks, Andrey

2 1

Performance Testing Kamailio 5.1 as SIP300 LCR on AWS - Recommendation on Tools and Methods
by KamDev Essa 24 May '18

24 May '18

I am currently setup with Kamailio 5.1 on an AWS t2.micro as a SIP300 LCR solution with a http_query call to an external resource (c# .NetCore app for LRN dip and MySQL Route choices) to get the contacts string, tag it to the Contact header an send 300. Its been a while since I have performance tested on SIP and It was SIPp. - Is http_query a bad choice for performance. Is KEMI better? - Are there any better free apps than SIPp? - Has anyone got cps stats for a SIP300 only Kamailio instance on AWS - Is AWS a bad choice for prime time LCR. Any gory stories? KD

2 3

SIP Server Hardware or VM ??
by Pinter, Gerd. 24 May '18

24 May '18

Hallo there, we are a Broadcasting Network with about 50 Stations and try to establish our own SIP Network for high quality Audio over IP with Bitrates of 192-256 kbit for transmitting an accurate stereo signal. Since we encountered problems with connections via mobile Networks we set up our SIP Server as RTP Proxy which resolved all those multiple NAT issues. But on the other hand we pay for that with a higher jitter rate. The question is, if anyone has experience if this issue maybe can be solved by running the Sip Server on a standalone hardware plugged into network Port on our Hardware Firewall, instead of a virtual Machine running on VM with virtual switches and so on. The current SIP Server has a Public IP Address, but all the Traffic runs thru Firewall, VPN, Hardware Switches and Virtual Switches. What you think? Thanks a Lot Nice regards Gerhard Pinter

4 5

count of registered users or count of tls connections
by Karthik Srinivasan 24 May '18

24 May '18

Hello, Via the routing file i am trying to find access two items:

2 5

Core Value : myself
by Duarte Rocha 23 May '18

23 May '18

Greetings, I'm doing some work with the AUTH routes and modules, and sometimes i get a "403 Not relaying" answer when the INVITE has a parameter in the R-URI. That parameter was not included in the REGISTER. I think it's this part of the code that throws this reply : # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { sl_send_reply("403","Not relaying"); exit; } Does a different or new parameter in the R-URI trigger the "uri!=myself" ? if so, can I set it up to check R-URI but not check the parameters? Thanks

2 1

Any modules with LRN dipping capacity?
by KamDev Essa 22 May '18

22 May '18

Are there any modules with LRN dipping capacity?. If not, whats the best way we can implement it as a bolt on. KD

3 14

Best way to implement custom LCR
by KamDev Essa 22 May '18

22 May '18

I apologize for asking unit questions today and decided to just present the big picture and address what I am trying to get @ holistically. I want to implement a custom LCR only solution on kamailio. INVITE in 300 out. Nothing else. I expect awesome cps stats that makes we lean towards kamailio rather than throwing up a home grown SIP server and worrying about performance. I have the guts of the LCR logic down pat, but todays its manual. In my process I do the following : - Check LRN of the requested user. I use dig doing a ENUM call. - Use the requesting user and LRN of requested user to mediate call type. Local or Interstate using LRN check via ENUM call (above) and LCAD data lookup. I use sql query. - Mediate call type Intrastate or International if not Local or Interstate using LERG lookup. SQL query too. - dip into database with requesting user and call type and get N lists by costs ascending. SQL query. - Construct Contact info in db or kamailio. - Send 300 reply will contact having the various routes. I can automate 2 through 6 using Kamailio. Basically xquery with supplied data and deal with the Contact header with data received. 404 or default route if no data received. I cannot handle the ENUM in kamailio or mysql. The provider does not provide SIP redirect. What are my options, besides changing LRN providers? KD

3 11

300 Multiple Choices response engine
by KamDev Essa 22 May '18

22 May '18

Whats the best route to implement a basic 300 Multiple Choices response engine. INVITE in 300 Multiple Choices out with choices in the response. I have a logic to generate my list choices, so a basic process would be better. H

2 10

2 pull requests: issue #1168 and another small issue
by Valentin Christoph 22 May '18

22 May '18

Hi all, Issued two pull requests. Any comment welcome, in particular, whether it will be taken for master branch...... https://github.com/kamailio/kamailio/pull/1538 https://github.com/kamailio/kamailio/pull/1539 Kr Christoph The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.

1 0

I think I found issue to Kamailio not forwarding ACK sent to it by Asterisk
by Wilkins, Steve 22 May '18

22 May '18

Hello All, When Kamailio sends a '200 OK' to Asterisk, it is putting its Public IP into the Record-Routes. In Asterisk 14.6, it would send the 'ACK' back to the Private IP Address of Kamailio, but Asterisk 15.x is using the Public IP Address that Kamailio placed in the Record-Routes so... Is there a way to force Kamailio to set the Record-Route to its Private IP address of the first Record-Route in messages forwarded to Asterisk? Thank you?

2 8

rtpp_test() in kamailio log
by Giovanni Tommasini - evosip 22 May '18

22 May '18

Hi guys, I've a question on using rtpengine with kamailio *scenario*: kamailio istance with rtpengine enabled and configured in localhost (but with a remote rtpengine is the same) modparam("rtpengine", "rtpengine_sock","udp:localhost:22222") kamailio version 4.4.6 + rtpengine version 6.3.0.0 running on Ubuntu 16.04 when I start kamailio I see a lot of INFO log rows like that May 22 12:13:12 my-dev /usr/sbin/kamailio[3388]: INFO: rtpengine [rtpengine.c:2546]: rtpp_test(): rtp proxy <udp:localhost:22222> found, support for it enabled May 22 12:13:12 my-dev /usr/sbin/kamailio[3397]: INFO: rtpengine [rtpengine.c:2546]: rtpp_test(): rtp proxy <udp:localhost:22222> found, support for it enabled ... more to the point. I see always 24 rows for each rtpengine_sock configured i n rtpengine.c I find the rtpp_tes function and that's fine, but why 24 times? thanks a lot -- Giovanni Tommasini | evosip web: evosip.cloud

2 2

Kamailio sends SIP messages to old websocket after reconnection during call
by Idris AVCI 22 May '18

22 May '18

Hi, We use kamailio as a backend for JSSIP based WebRTC UAs. We use SIP over Weboscket as transport and everythings work fine generally. Some websocket connections to kamailio drop during an active call and JSSIP reconnects immediately. New websocket connection has a different port than the dropped one but kamailio tries to send to old websocket port which is not active anymore. We use GRUU and configured all the related parameters in registrar, dialog, tm, websocket, tls, xhttp and userloc modules both for versions 4.4 and 5.1 but couldn't find a solution yet. For a typical call below are the ordered events: 1. UA register to kamailio over wss (Code: WS1) 2. Kamailio send an INVITE to the UA (Code: Call1) 3. UA answer and media starts 4. UA drops WS1 to kamailio and reconnect (Code: WS2) 5. UA has an active websocket (WS2) to kamailio 6. Kamailio tries to send BYE message of Call1 to WS1 which is no longer active and complains that websocket could not be found 7. Kamailio send an INVTE to the UA (Code: Call2) over WS2 8. UA answer and media starts 9. Kamailio send BYE message of Call2 to UA over WS2 and everything work as expected (No websocket drop) Error log: WARNING: <core> [msg_translator.c:2761]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found I think we need a mechanism that will update the dialog information of Call1 to use WS2 after reconnection. Any help greatly appreciated. Thanks.

2 2

htable - unable to update entry when event_route[htable:expired:<table>] is executed
by David Heath 22 May '18

22 May '18

Hi, Looking for some help/ advice on updating the expiration timer/ values in a hash table when when [htable:expired:<table>] occurs. On startup my script successfully reads in a set of entries from a DB; I also know from testing/ tracing that I can add new entries to the hash table in request_route. My issue occurs when the expiration timer kicks in - my goal was to re-read from the DB to ensure either the entry still exists or hasn’t changed and either delete, update or refresh the value until the next time the expiration event occurs. This way entries in the DB could be modified overtime. At the moment all that’s occurring is that in the event_route[htable:expired:cli] block the entries are being drained one-by-one. I re-read the row from the from the DB and I reset the value using $sht(cli=>$shtrecord(key)) = $xavp(cli=>MAPPED_CLI); I’ve also tried updating the expiration timer against the key. I’ve tried both $shtex(cli=>$shtrecord(key)) = 60 and $shtex(cli=>$shtrecord(key)) = $TS+60 [the latter based on the values shown from sht_print()] but the entries still get deleted. I’ve tried this against both 5.1.0 & 5.1.2. Any help appreciated. Thanks David Heath

1 0

Is it possible to change a positive (2xx) response to a negative one (4xx) during kamailio processing of a reply (g711 fallback drama)?
by George Diamantopoulos 21 May '18

21 May '18

Hello all, I've been hitting a wall in getting T.38 fallback to G.711 to work in a specific case, and I was wondering if I could use kamailio to the rescue, since all my other endeavours have failed miserably. In short, this is this the scenario where I need to have this manipulation happen: - During an established dialog between a SIP endpoint and asterisk (with kamailio in the middle acting as proxy), the latter (asterisk) sends a re-INVITE with a single offer for m=image with T.38 in the SDP. - The other (eccentric) endpoint (yes, it's a Cisco) replies with 200 OK, but will reject the T.38 offer in the SDP body instead by including a port number of '0' - Instead of falling back to G.711 as configured, asterisk in that case will do nothing, and the dialog will eventually end with a BYE (don't remember which of the two endpoints will send that, but it doesn't matter much) The thing here is that asterisk *WILL* fallback to G711 with a second re-INVITE, if the other endpoint replies to the first T.38 re-INVITE with a negative response (488) instead of a positive one (200) declining the media in the SDP body. I've looked at using freeswitch for a fax gateway instead, but apparently FS doesn't support fallback at all even in the 488 reply case, much less in the inactive media stream in 200 OK corner case I'm investigating. So, I thought I'd do that in kamailio and convert those 200 OKs to 488s with change_reply_status() from the textopsx module. Of course, the universe hates me, and this will fail spectacularly: > ERROR: textopsx [textopsx.c:301]: ki_change_reply_status(): the class of provisional or positive final replies cannot be changed So you can't change a reply's "class" with change_reply_status(), only the response code within that class can be manipulated. I guess my question is now whether there is any way to do this sort of transformation with kamailio, perhaps with some other function or method? If anyone else has had to deal with this problem and come up with a different solution I'd be glad to hear it, as neither my head nor the surrounding walls can take any more mutual banging at this point. Thanks! BR, George. P.S. I hate fax

2 4

Charging support/protocol information in Kamailio
by amit Kumar 21 May '18

21 May '18

Hi All, Could you please provide the charging related Information in Kamailio. ie list of Charging protocol/standard Kamailio Support ??. As per the kamailio modules(ims modules) , Kamailio support Diameter.Is there any other standard/interface Kamailio support.?? Thanks, amit

2 1

Cannot get rid of following errors, and wondering if this is causing some of my issues
by Wilkins, Steve 21 May '18

21 May '18

ERROR: <core> [core/pvapi.c:1452]: pv_printf(): no more space for spec value ERROR: <core> [core/pvapi.c:1461]: pv_printf(): buffer overflow -- increase the buffer size... I have pv_buffer_size=16384; tcp_rd_buf_size=16384; Any ideas? Thank you,

1 0

Supper confused on Kamailio with Asterisk Call
by Wilkins, Steve 21 May '18

21 May '18

Hi All, I am using Kamailio 5.2 and Asterisk (14.6 & 15.x) and I am having a very strange issue. If I use Asterisk 14.6, the call (WebRTC<=>WebRTC) works perfectly. However, if I use Asterisk 15.x, the call drops in 30 seconds. In comparing tcpdump files, the first place I see a difference is in the '200 OK' response to Kamailio' s 'INVITE to Asterisk'; in Asterisk 15.3 the first record Record-Route is Kamailio Public IP Address, however, in the Asterisk 14.6 (the working call) it is Kamailio' s Private IP Address. Asterisk Developers told me it is Kamailio doing this...and I am confused. Is this true, and if so why would it be different based on the Asterisk version? Thank you!

1 0

Sudden database drop
by Amar Tinawi 20 May '18

20 May '18

Hello list recently i faced strange behaviour when dealing with ng-voice version of kamailio ims specially in P-CSCF node, the database "pcscf" dropped suddenly (deleted) without any command or action from user this happened many times and unfortunately there is nothing in the syslog as error massages Any advice please ? Thanks Cheers, Ammar

2 3

WebRTC to WebRTC call Kamailio ACK Problem
by Wilkins, Steve 20 May '18

20 May '18

Hello, I am using Kamailio 5.2 and Asterisk 15.3 and while doing a WebRTC (client1) to WebRTC (client2) call, Asterisks sends a re-invite and Kamailio responds with a '200 OK', but when Asterisk sends back the "ACK" to Kamailio, it cannot be relayed. Errors => WARNING: <core> [core/msg_translator.c:2767]: via_builder(): TCP/TLS connection (id: 0) for WebSocket could not be found ERROR: <core> [core/msg_translator.c:1982]: build_req_buf_from_sip_req(): could not create Via header ERROR: <core> [core/forward.c:549]: forward_request(): building failed ERROR: sl [sl_funcs.c:362]: sl_reply_error(): stateless error reply used: I'm terribly sorry, server error occurred (1/SL) As a side note, I still occasionally see the Errors => ERROR: <core> [core/pvapi.c:1452]: pv_printf(): no more space for spec value ERROR: <core> [core/pvapi.c:1461]: pv_printf(): buffer overflow -- increase the buffer size... No matter how large I make pv_buffer_size, and tcp_rd_buf_size; pv_buffer_size=16384, and tcp_rd_buf_size=16384; Thank you all!

1 0

deb.kamailio.org bionic
by Kelvin Chua 19 May '18

19 May '18

any plans of adding support for bionic? Kelvin Chua

3 3

Transport issue thought
by Wilkins, Steve 18 May '18

18 May '18

It appears that the bottom line of my TCP transport not working is that Kamailio is randomly assigning large port numbers to send the TCP traffic out on. I am not able to randomly open high ports for this purpose. Is there a way to tell Kamailio to only use specific ports for this. I have tried using force_send_socket() with Kamailio' s IP, and the port I want to send out on, but this did not work either. Does this sound like I am on the right track? Thank you, -

3 11

Transport question
by Wilkins, Steve 18 May '18

18 May '18

Hi All, I was attempting to switch from UDP to TCP for relaying. I used t_relay_to_tcp() instead of t_relay() => if (!t_relay_to_tcp()) { sl_reply_error(); } The problem is that I am still seeing the SIP messages as UDP on the Server that the messages are relayed to. Any Ideas? Thank you, Steve

6 17

Hunting - Dynamic logistics
by KamDev Essa 18 May '18

18 May '18

So I finally got to the docs on hunting under the TM Modules and there was an awesome write up and it is making a lot more sense now. Basic logic is to append_branch with your extensions and and manage serial and/or parallel forking by setting the q values. The script is static. Is there a module that automatically loads registered contacts from a grouped list of extensions or is the best way to dip into the db and DIY? I would like to know how some of you have implemented hunting, Also is there a way to set the fr_inv_timer of each branch. Thanks in advance. KD

2 1

Alcatel-Lucent Enterprise acquires majority stake in Sipwise
by Andreas Granig 18 May '18

18 May '18

Dear community, I'd like to inform you about Alcatel-Lucent Enterprise (ALE) acquiring a majority stake in Sipwise. Over the last 10 years, we at Sipwise were committed to contributing significant efforts to the open source telecom environment, resulting amongst others in the free and open source Sipwise CE soft-switch appliance, the rtpengine media relay, as well as major contributions to the Kamailio SIP proxy. Under the ALE roof, Sipwise is continuing to operate as its own brand, and our commitment to open source and our community components will be as strong as ever. Please find the official announcement here: https://www.sipwise.com/ale-sipwise/ Best regards, Andreas

3 2

Re: [SR-Users] rtpproxy not forwarding rtp packets.
by Giovanni Tommasini - evosip 18 May '18

18 May '18

Hi Younas, when you have a call if you make a trace in your sip server with sngrep or tshark can you see the RTP packet ? you says "I have kamailio server behind nat with rptproxy.", so is it possible that the router in front your SIP server blocks the traffic? could you have a trace there? I mean the packets exit the rtpproxy but don't arrive to callee or just don't come out of the rtpproxy? Giovanni Tommasini | *evosip* <http://evosip.cloud> 2018-05-18 1:36 GMT+02:00 Aqs Younas <aqsyounas(a)gmail.com>: > Thanks for replying. Callee is actually a sip provider. I see no firewall > or iptables rules on server to prevent rtp following towards callee. > > Is there anything i can do to see which thing is blocking rtps. > > Best > > On Fri, 18 May 2018 at 3:59 AM, Mack Hendricks <ap(a)goflyball.com> wrote: > >> Is there something blocking RTP traffic from reaching the callee? Is the >> callee a carrier or the actually endpoint endpoint (aka SIP Phone)? >> >> *Mack Hendricks / Head of Support / dOpenSource* >> web: http://dopensource.com >> support: +888-907-2085 >> dSIPRouter <http://dsiprouter.org> - GUI focused on implementing >> Kamailio to provide SIP Trunking and PBX Hosting Services >> >> On May 18, 2018, at 12:50 AM, Aqs Younas <aqsyounas(a)gmail.com> wrote: >> >> Greetings list, >> >> >> I have kamailio server behind nat with rptproxy. But i am getting no >> voice on the call. After taking trace i could see that rtpproxy was getting >> rtp packets but not packets was being forwarded towards callee side. >> >> Though i see in rtpproxy logs, packets being relayed from caller side. >> >> Does rtpproxy wait to receive a single rtp frame from callee before >> fowarding that to callee? >> >> Since, i am getting no rtp from callee. So, might be rptrpoxy is waiting >> to learn source address of callee side rtp, that is why it is not >> forwarding rtp packets from caller towards callee. >> >> >> Any pointer/suggestion is much appreciated. >> >> Best Regards, >> >> Aqs Younas >> >> _______________________________________________ >> 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 > >

4 4

Help on how to debug cfg_get variable having unexpected value?
by Malcolm O'Hare 18 May '18

18 May '18

I'm trying to debug an issue I've encountered where a variable that I've set using kamcmd cfg.set has an unexpected value when used in kamailio. The variable should be 0 or 1, but it looks like its getting set accidently somehow. 2018-05-11T22:02:58.344260+00:00 ip-172-31-129-45 kamailio[5253]: DEBUG: <core> [select.c:263]: resolve_select(): 'with_dynamodb_user_data' 2018-05-11T22:02:58.344264+00:00 ip-172-31-129-45 kamailio[5253]: DEBUG: <core> [cfg/cfg_select.c:174]: select_cfg_var(): DEBUG: select_cfg_var(): select fixup is postponed: features.with_dynamodb_user_data 2018-05-11T22:05:03.467110+00:00 ip-172-31-129-45 kamailio[5551]: INFO: <core> [cfg/cfg_ctx.c:608]: cfg_set_now(): INFO: cfg_set_now(): features.with_dynamodb_user_data has been changed to 1 2018-05-11T22:15:51.695118+00:00 ip-172-31-129-45 kamailio[5574]: [2c03c333-1705-41b1-aa26-6f10492a5a6f] DEBUG: with_dynamodb_user_data is 1157627905 I've tried to debug using gdb using breaks and set a watch on the memory address of the var, but it looks like it keeps using different memory locations. The value in the address pointed at by group.handle seems to change for each invocation of set_cfg_now, so when I try and put a watch on the address of *group.handle + var.offset it never gets hit. (gdb) print *group $36 = {num = 25, mapping = 0x7f56ee4474f0, vars = 0x0, add_var = 0x0, size = 164, meta_offset = 968, var_offset = 984, handle = 0x7f56edf92fd0, orig_handle = 0x0, dynamic = 1 '\001', next = 0x0, name_len = 8, name = "f"} (gdb) print *var $62 = {def = 0x7f56ee471668, name_len = 23, pos = 24, offset = 164, flag = 0} Any ideas? Malcolm

2 2

Auth module with shared "secret"
by Asgaroth 18 May '18

18 May '18

Hi All, I am testing a scenario where we have 3 proxies and I'm trying to ensure that if a messages comes in to 1 proxy, which challenges the ua, is able to be auth'd successfully on one of the remaining 2 proxies if, for some reason the ua decides to send the response to one of the remaining two proxies. I am testing this using a seperate kamailio proxy which loadbalances messages across the 3 proxies that I am testing with. I have 1 ua (zoiper) which I tell to subscribe for mwi, the initial subscribe is sent to proxy 1, which sends a 407 back, the new subscribe with auth, from the ua, is sent to proxy 2, which deems the nonce as expired and rechallenges, this continues in a loop across the 3 proxies and never authenticates. I think I'm missing something in my settings that I just cannot put my finger on, and/or, I'm mis-understanding the documentation and how these settings interact with each other. My auth module settings are as follows: # # ----- auth params ----- # modparam("auth", "qop", "auth") modparam("auth", "nonce_count", 1) modparam("auth", "one_time_nonce", 1) modparam("auth", "nid_pool_no", 4) modparam("auth", "nc_array_size", 4194304) modparam("auth", "otn_in_flight_no", 8388608) modparam("auth", "auth_checks_no_dlg", 9) modparam("auth", "auth_checks_in_dlg", 15) modparam("auth", "auth_checks_register", 11) modparam("auth", "secret", "secret_32_char_str") I tried with one_time_nonce enabled and disabled but the result is the same, continous rechallenge. The documentation mentions that if nonce_count and one_time_nonce are enable, and a ua sends a response with nonce and qop in the auth header then one_time_nonce will not be used, one_time nonce will be used as a fallback in the case the ua does not support qop. My auth block in the routing logic looks like so: route[AUTH_CHECK]{ xlog("L_DBG", "route[AUTH_CHECK] : $rm : Performing authentication checks for '$var(creds_key)'"); $var(challenge_flags) = 1; if ( ! pv_auth_check("$fd", "$sht(creds=>$var(creds_key)::Password)", "20", "1") ) { switch($retcode) { case -1: xlog("L_DBG", "route[AUTH_CHECK] : $rm : Generic error occurred, no reply sent out."); break; case -2: xlog("L_DBG", "route[AUTH_CHECK] : $rm : Invalid password supplied, re-challenging client and removing from hash table."); sht_rm_name_re("creds=>$var(creds_key)"); break; case -4: xlog("L_DBG", "route[AUTH_CHECK] : $rm : Nonce has expired, re-challenging client."); $var(challenge_flags) = $var(challenge_flags) + 16; break; case -5: xlog("L_DBG", "route[AUTH_CHECK] : $rm : No credentials supplied, challenging client."); break; case -6: xlog("L_DBG", "route[AUTH_CHECK] : $rm : Nonce has already been used to auth a previous request, challenging client."); break; case -8: xlog("L_DBG", "route[AUTH_CHECK] : $rm : Auth user is different to From/To user, challenging client."); break; } xlog("L_INFO", "route[AUTH_CHECK] : $rm : auth_challenge will be called with '$fd' and '$var(challenge_flags)'"); auth_challenge("$fd", "$var(challenge_flags)"); exit; } consume_credentials(); } When I look at the logs I always see the proxies seeing the nonce as expired: oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' oUBGp0n6HRWu_mvPPxJQqg..: INFO: <script>: route[AUTH_CHECK] : SUBSCRIBE : auth_challenge will be called with 'example.com' and '17' I have checked: [1] that the proxies are all syncing time, and they are not out of sync, [2] The source address of the client doesnt change (in case auth_checks_no_dlg is failing), [3] The r-uri of the subscription dialog is the same (in case auth_checks_no_dlg is failing) I am using kamailio version 5.0.6: kamailio -V version: kamailio 5.0.6 (x86_64/linux) flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, 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: unknown compiled on 15:57:47 Mar 5 2018 with gcc 4.8.5 Any thoughts on what I can look at or if I'm missing something fundamental here? All suggestions/criticisms are appreciated. Thanks for reading.

3 6

rtpproxy not forwarding rtp packets.
by Aqs Younas 17 May '18

17 May '18

Greetings list, I have kamailio server behind nat with rptproxy. But i am getting no voice on the call. After taking trace i could see that rtpproxy was getting rtp packets but not packets was being forwarded towards callee side. Though i see in rtpproxy logs, packets being relayed from caller side. Does rtpproxy wait to receive a single rtp frame from callee before fowarding that to callee? Since, i am getting no rtp from callee. So, might be rptrpoxy is waiting to learn source address of callee side rtp, that is why it is not forwarding rtp packets from caller towards callee. Any pointer/suggestion is much appreciated. Best Regards, Aqs Younas

2 2

Further question on transport issue.
by Wilkins, Steve 17 May '18

17 May '18

Hello, I am still struggling with getting TCP relay to work. Kamailio is now trying to send TCP but it is using the incorrect port. Kamailio is relaying to Asterisk on xx.xx.xx.xx:some-large-port but Asterisk is listening on xx.xx.xx.xx:5060. I'm not sure why this is happening. I would appreciate any help. Thank you,

3 4

cfg script 101
by KamDev Essa 17 May '18

17 May '18

I have decided to do Hunting DIY. I needed some help with the cfg script part and using a database dip. Say I do the db dip sql_xquery("ca", "call usp_GetHuntBranches( '$rU', '$rd');", "ra");that will return the uri/q pairs and I will add using append_branch dynamically. How do I do the looping in the script? I have 2 ways to deliver the list via the sp. - Rows with uri and q values. easiest - Single row with variable paired fields like say ur1,q1,ur2,q2 etc etc Questions : - Can the script handle sql_xquery with multiple rows. If so, can I get a code snippet or example that show me how to iterate through. Is there a "while" loop capability in the script? - Is there a "substring" or "split" function in the script that could allow me to get the data as a single text with uri and q separated by a \ or |. - Or is there a document that explains the various generic functions I can use the cfg script area. KD

2 2

NatHelper's Keep-alive timeout behind proxy
by Asgaroth 17 May '18

17 May '18

Hi All, We are looking to enable nathelpers keepalive timeout parameter where it will remove an aor's contact if it doesn't receive a reply from the ua. The problem we have is that our registrar's are seperated from our proxies, so this means nathelper is sending options messages via the proxy to the ua. In this case, if the us has lost connectivity, the kamailio proxy sends a 408 Request Timeout back to the registrar and the nathelper module sees that as a response to the initial options message it sends, so it never actually removes the contact from the location table. Is there a way I can stop the proxy from sending any responses for these specific nat ping/options messages, or, is there a way I can tell the nathelper modules that a 408 response should be treated as the ua being down and then after the timeout interval remove the contact from the location table? I've had a read of the nathelper, tm and sl modules but I didnt see anything that jumps out at me that I could achieve one or the other possible solutions I mentioned above. Does anyone have any suggestions or pointers on how we can achieve this with the registrars being seperate from the edge proxies. Any pointrs/tips etc would be greatly appreciated. Thanks

2 2

Updated NAT traversal article
by Alex Balashov 17 May '18

17 May '18

Hi, I have updated this article with some other topics: http://blog.csrpswitch.com/server-side-nat-traversal-with-kamailio-the-defi… "The definitive guide" is of course an ambitious and moving target, so there may be more evolution in the future. -- 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/

5 7

t_load_contacts() and t_next_contacts() guts
by KamDev Essa 16 May '18

16 May '18

I am tackling Hunting and see that there are two functions recommended to accomplish this. Before I jump into using them I wanted to know a bit about each regard to the call flow. >From where does t_load_contacts() load contacts and how does it order then in case of a serial fork flow, Is there a generic way of doing it say with a loop in the script. Docs on these two functions are a bit cryptic for me H

2 7

Contact Header
by KamDev Essa 16 May '18

16 May '18

My scenario is UA <====> Kamailio <====> Freeswitch UA calls a number. Kam delivers to FS and FS plays a wav file. My problem is that Kam relays the 200 OK from FS back to UA without change and UA ACKs back to FS because the FS IP is in the Contact Header. - Why does Kam not change the contact header since the ACK should come back to Kam - If this is normal what do I need to do to change the contact header for the 200K to the UA to make this work. KD

3 4

Whats is wrong in this call
by Sergey Safarov 15 May '18

15 May '18

Hello I have a call with not passed ACK by Kamailio. http://prntscr.com/jhtobo Same behaviour i looked before when SIP endpoint contact string is wrong In this call look as contact string correct is correct (packets 7 and 8) and i not understand why ACK (packet 9) not passed to callee. Could you help me understand this. PCAP file is create using homer tracer and TCP transport packets is stored as UDP. Sergey

3 4

Last negative 5XX reply not getting relayed
by Sergiu Pojoga 15 May '18

15 May '18

Not sure if it's a default Kamailio behavior or something I'm doing wrong... When receiving a negative reply for an INVITE (more like in the 5xx range), proxy doesn't relay the message to the originator but instead issues it's own "500 Service Unavailable". That doesn't happen, from observations so far, to "603 Declined" or "486 Busy Here" or "404 Not Found" or "200 OK" replies. *For example, here's the negative reply proxy receives from the callee:* 2018/05/14 10:52:37.716627 65.XX.XX.166:5060 -> 65.XX.XX.167:5060 SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP 65.xx.xx.167;branch=z9hG4bK9223.00b4f3143d245a895c4bb6a107add258.0;received=65.xx.xx.167;rport=5060 Via: SIP/2.0/UDP 205.xx.xx.221;received=205.xx.xx.221;rport=5060;branch=z9hG4bKQ4XpXga0vgZ8B From: "514XXXXXXX" <sip: 514XXXXXXX @205.xx.xx.221>;tag=tXj0NHrje9jNS To: <sip:450XXXXXXX@65.xx.xx.167>;tag=as18e452b7 Call-ID: 48eac296-d229-1236-a685-005056a149a4 CSeq: 122818714 INVITE Server: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas X-Asterisk-HangupCause: Subscriber absent X-Asterisk-HangupCauseCode: 20 Content-Length: 0 *Here's what the originator gets in the final message:* 2018/05/14 10:52:37.718143 65.xx.xx.167:5060 -> 205.xx.xx.221:5060 SIP/2.0 500 Service Unavailable Via: SIP/2.0/UDP 205.xx.xx.221;rport=5060;branch=z9hG4bKQ4XpXga0vgZ8B;received=205.xx.xx.221 From: "514XXXXXXX" <sip:514XXXXXXX@205.xx.xx.221>;tag=tXj0NHrje9jNS To: <sip:450XXXXXXX@65.xx.xx.167>;tag=31a2c69b5d434ef4a60796e26f568bf0-9343 Call-ID: 48eac296-d229-1236-a685-005056a149a4 CSeq: 122818714 INVITE Server: Kamailio 5.1 Content-Length: 0 I'd like any reply to be relayed back as is, unless I do some t_check_status() => t_reply() for specific reasons (which I have none at this point). Thanks Sergiu

3 5

ACC_CDRS table with src and dst data are inverted
by Emanuele Gambaro 15 May '18

15 May '18

Hi to all, I have some troubles with acc_db module and acc_cdrs table. First scenario: Alice calls Bob and after a while, when a call is established, Alice hangs up, sending a BYE. The acc_cdrs table is filled up with the right data: * src_user, src_domain, src_ip are from Alice account (the caller) * dst_user etc etc are from Bob account (the callee) In this scenario all works as expected: the src data are from who starts the call, and the dst data are from who receives the call. In the following scenario src and dst data are inverted. Alice calls Bob and after a while, when a call is established, Bob hangs up, sending a BYE. The acc_cdrs table now is: * src_user, src_domain, src_ip are from Bob account (the callee that sends the BYE message) * dst_user etc etc are from Alice account (the caller that receives the BYE) This is the intended behavior or I’m missing something? How I can record the CDR data with the right CALLER and CALLEE data ? Emanuele

2 1

UAC Module Registration Table
by Greg Adams 15 May '18

15 May '18

Hey there, I am currently using the UAC module to maintain remote registrations. I am just curious if it is possible to register new users immediately rather than waiting for reg_timer_interval? I am not seeing any functionality like that but just wondering if I am missing something or if it would be possible to add in. For example an RPC command that takes a key and instantly registers that user. Thanks, Greg

2 1

Parallel Forking - 1 Ring on extension to notify that Call Forward is in place
by KamDev Essa 14 May '18

14 May '18

I am trying to simulate a Broadsoft feature that blips the extension with a single ring that basically heralds the user that their extension has CF turned on. I know parallel forking can do that. But how do I set the fr timer on the call to the extension @ just 1 ring (5 seconds) and let the other one the default timeout set in fr_inv_timer. KD

2 1

Tagging a call with custom pseudo variables - cfg 101
by KamDev Essa 14 May '18

14 May '18

Being spoiled by Freeswitch profiles and dialplans, the stock 1 piece Kamailio cfg file is cool @ first to get certain call flows knocked out but quite a maze when you get in deeper. Right now, using the 1 piece stock cfg file, I am faced with the fact that my carrier outbound calls started failing after I enabled VM. Basically WITH_VOICEMAIL in route LOCATION started sending the carrier calls to TOVOICEMAIL and it basically dies there. I can see the route(TOVOICEMAIL) line and can condition it out with an if but I really don't now how to differentiate between a ext to ext call or a outbound call in the cfg file. However given SQL I could. Is there a way for me to place custom pseudo variables in the call right @ the start and use them in the script as needed. For example I could mark the call as ext to ext or carrier bound right @ the start and dodge VM in the LOCATION route. KD

2 5

Question about using Kamailio and Asterisk and flow of an "INVITE"
by Wilkins, Steve 13 May '18

13 May '18

Hello All, I am trying to resolve, in my mind, the flow of a WebRTC<=>WebRTC call using Kamailio and Asterisk. Each WebRTC client is registered in Kamailio and when I call WebTRC Client1 from WebRTC Client2 what I see is -> The Invite is sent from Kamailio to Asterisk and then Asterisk is sending the Invite back to Kamailio. Also depending on The version of Asterisk, the INVITE will then get forwarded to the AOR that is registered in Kamailio for the called number. Does this seem correct? It seems like there is an extra hop in there. The reason I am now very curious now is because everything works fine if using Kamailio 5.0 and Asterisk 14.6, but I switch to Asterisk 15.3 I get the extra hop and call is dropped after 30 seconds. I would appreciate any thoughts on this. Thank you in advance.

3 6

"Sleeping Giant - No Snore" Kam?
by KamDev Essa 11 May '18

11 May '18

We have noticed pesky SIP hackers already knocking on our POC servers. Kam is holding the fort by sending out the proverbial 401/407 (snore). I can SQLOPS to do a dip to check destination domain or Source IP and nix the snore for alien requests, but wanted to know if there was a native way to keep silent when we have a request from foes. KD

3 6

Understanding port changing
by Social Boh 11 May '18

11 May '18

Hello list, I'm looking for to understand a problem have with Kamailio. The configuration is: user (behind NAT) -> Kamilio (Public IP) -> user (behind NAT) When I make a call from user A to user B (both registered) , User B receive the INVITE correctly but in the ringing SIP message change the port present on the Contact Header and same think on 200OK so when Kamailio try to send it ACK SIP message to this port fail because the Softphone listen on the port where has received the INVITE from Kamailio. This happens with 3 different Softphone: BRIA, Xlite, Linphone on the same computer. Thank you Regards -- --- I'm SoCIaL, MayBe

2 1

WebRTC to SIP gateway
by Pan Christensen 11 May '18

11 May '18

Hello! It's been several years since I've used Kamailio. My current employer wants to implement WebRTC, which is currently not supported in our SIP backend, and asked if I could set up a Kamailio server as a gateway. I've been able to make calls in all directions between SIP and WebRTC clients registered locally on Kamailio. When I tried to connect the server to the SIP backend, I ran into an issue. I'm able to register SIP clients in the backend via the gateway and make calls everywhere. However, the WebRTC client fails to register. Here are the messages between the Kamailio gateway and the SIP backend: U 2018/05/09 10:12:58.316643 GATEWAY:15060 -> DOMAIN:5060 REGISTER sip:DOMAIN SIP/2.0. Via: SIP/2.0/UDP GATEWAY:15060;branch=z9hG4bK4fc6.04d1730be5b78d595c69a3aa137987c1.0. Via: SIP/2.0/AUTO lr2l9s72ehhc.invalid;rport=61353;received=CLIENT;branch=z9hG4bK5927151. Max-Forwards: 68. To: <sip:4777519304@DOMAIN>. From: <sip:4777519304@DOMAIN>;tag=9qhqhrnj3s. Call-ID: j5h7830ivr5dfc2mn5sov1. CSeq: 4 REGISTER. Contact: <sip:c3qkm4fv@lr2l9s72ehhc.invalid;transport=ws>;+sip.ice;reg-id=1;+sip.instance="<urn:uuid:0ef2deac-2d56-465a-840b-543b9fd01af8>";expires=600. Expires: 600. Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO. Supported: path,gruu,outbound. User-Agent: JsSIP 3.2.9. Content-Length: 0. Path: <sip:GATEWAY:15060;lr>. . U 2018/05/09 10:12:58.368409 DOMAIN:5060 -> GATEWAY:15060 SIP/2.0 400 Wrong transport. Provided transport either invalid or not supported.. Via: SIP/2.0/UDP GATEWAY:15060;branch=z9hG4bK4fc6.04d1730be5b78d595c69a3aa137987c1.0. Via: SIP/2.0/AUTO lr2l9s72ehhc.invalid;rport=61353;received=CLIENT;branch=z9hG4bK5927151. To: <sip:4777519304@DOMAIN>;tag=91334f57. From: <sip:4777519304@DOMAIN>;tag=9qhqhrnj3s. Call-ID: j5h7830ivr5dfc2mn5sov1. CSeq: 4 REGISTER. Content-Length: 0. I believe that this error message is caused by ';transport=ws' in the Contact header. I'm not allowed to modify this header. In the backend database, I found that some other clients have ';transport=UDP' in their path headers, so I tried to add that. (Why can I not add parameters in path module without adding username?) I still got the same error. How do I best proceed? For your information: We have outsourced the development of the WebRTC client, so we are able to change it. We also have the option of paying the supplier of the backend for development there. 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: +47 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…>

2 2

"No Answer" time out after number of rings
by KamDev Essa 11 May '18

11 May '18

Does Kamailio have any way to control how long it tries to connect per call? KD

2 5

Is Media Release (signaling only) possible in Kamailio?
by KamDev Essa 10 May '18

10 May '18

What cfg files changes do I need to make to get Kamailio to be a signally only server, yet manipulate the SDP part of the INVITE message to allow remote parties to send media to each other? In Freeswitch terms "bypassmedia". KD

2 14

Build Kamailio with MariaDB failed.
by Ye, Charlie (NSB - CN/Qingdao) 10 May '18

10 May '18

Hi all, Does everyone use MariaDB in Kamailio? When I built Kamailio with MariaDB, the below error happened. Would anyone give a help? Thanks in advance! [root@clcmos /usr/local/src/kamailio-devel/kamailio]$ # make all make -C src/ all make[1]: Entering directory `/usr/local/src/kamailio-devel/kamailio/src' make[3]: `libsrdb2.so.1.0' is up to date. make[3]: `libsrdb1.so.1.0' is up to date. LD (gcc) [M db_mysql.so] db_mysql.so /usr/bin/ld: cannot find -lmariadb collect2: error: ld returned 1 exit status make[2]: *** [db_mysql.so] Error 1 make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/local/src/kamailio-devel/kamailio/src' make: *** [all] Error 2 Thanks and Best Regards, Charlie

4 6

Parse presece xml and put stats into asterisk database
by Aqs Younas 09 May '18

09 May '18

Greetings list, I need to put presence info of a device into asterisk database through sqlops module. Currently, i opt to use a lua script to parse xml body of PUBLISH packet issued by pua_dialoginfo and insert trying/ringing stats into asterisk database. Is there any better way to accomplish this? Best Regards, Aqs Younas

1 0

P-CSCF error: pv_parse_spec2(): error searching pvar "uac_req"
by John M 09 May '18

09 May '18

Hello, I upgraded Kamailio 4.4 to 5.2.0-dev5 and when I tried to start the service(p-cscf), I am getting below error. However I am able to start the service with 4.4 p-cscf example cfg. Also after upgrading I-CSCF and S-CSCF nodes are running without any issue. Any help would be greatly appreciated! [root@8bd3221bd31e kamailio]# kamailio -f /usr/local/src/kamailio-5.1/kamailio/misc/examples/ims/pcscf/kamailio.cfg loading modules under config path: /usr/local/lib64/kamailio/modules/ 0(2464) ERROR: <core> [core/pvapi.c:903]: pv_parse_spec2(): error searching pvar "uac_req" 0(2464) ERROR: <core> [core/pvapi.c:1107]: pv_parse_spec2(): wrong char [m/109] in [$uac_req(method)] at [9 (5)] 0(2464) CRITICAL: <core> [core/cfg.y:3489]: yyerror_at(): parse error in config file /usr/local/src/kamailio-5.1/kamailio/misc/examples/ims/pcscf/kamailio.cfg, line 831, column 3-18: Can't get from cache: $uac_req(method) ERROR: bad config file (1 errors) 0(2464) INFO: <core> [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized Regards, Uts

2 1

Use GStreamer instead of FreeSwitch or Asterisk?
by Chandramouli P 09 May '18

09 May '18

Hello, I am using Kamailio as registration server and FreeSwitch for signalling (RTP packet handling). Can I use GStreamer instead of Freeswitch or Asterisk? Thank you. Regards, CM

2 1

Connecting UAs behind Firewall/CgNat with Kamailio without using a Stun Server
by Pinter, Gerd. 09 May '18

09 May '18

Hi there, unfortunately I am personally not a complete IT geek, I am more in to Pro Audio. Anyway, I have to maintain a SIP Server for high quality Audio transmission with special UA's. Also we use Software UA's running as Apps (Luci Live) on mobile devices such as I Phones and I pads which are by nature situated in mobile networks. Due to "lack of IP address issues" most providers use private ranges separated by CGNAT from the real Internet. This causes a lot of trouble. In some cases stun server entries on the clients helps, in other cases don't! But these days I read in announcement of a german company (Mayah Communications), which is in the ProAudio biz since decades, that they offer a SIP Service where no stun is recommended whether the clients are behind NAT or not. After I acknowledged that IT business is not witchcraft (almost :-) ) , I thought that there might exist a solution for my Kamailio SIP server to make it work like the "Mayah" Sip Server. What settings have to be done that Kamailio uses IP Header Information for SIP signalling instead of the content of SIP packages themselves. Without stun you find the Local Address in the SIP packages, and with Stun the external Address but mostly with the wrong portnumber, but in the IP header you'll find the correct IP:port. I already read a lot of the Kamailio documentation but due to the fact that I am not too deep in IT I am not sure if I got it all. If someone could give me a useful hint on that, that would make my day. Hope this description is unterstandable. Nice Regards Gerhard Pinter

4 8

Kamailio + RTPEngine + RTPEngine Recording daemon forward-to option
by Balázs Radák 08 May '18

08 May '18

Dear all, I would need some help setting up Kamailio with RTP engine. I'm trying to make use of RTPEngine recording daemon's forward-to option, to have the calls streams sent to 3rd party application. Have someone managed to have this forward-to option work? If I understood well, when RTPEngine's forward-to option is turned on, the call metainfo + streams can be sent to a socket server? Thanks in advance! Regards, Balazs

2 1

Copy $xavp
by Duarte Rocha 08 May '18

08 May '18

Greetings, I'm getting a query result in a $xavp. I'm trying to copy the entire structure to a new $xavp like this : $xavp(Valid_Acd_Results) = $xavp(ACD_Query_Result[$var(iterator)]); However, i can't retrieve the value with : xnotice(" Mode = $xavp(Valid_Acd_Results=>TBK_AcdMode)"); since it returns NULL. Am i doing something wrong? Thanks

1 0

Record-route and proxy_challenge
by Евгений Голей 08 May '18

08 May '18

Hi! Tell me why when using the save ('') function in the response generated by the server, there are the Record-route headers that were in the request. And when using the following functions: - proxy_challenge (), - www_challenge (), - send_reply (), - sl_send_reply there is no Record-route header. How can you add them? -- Evgeniy

2 2

Garbage in to field
by Kjeld Flarup 08 May '18

08 May '18

I'm having problems with my To field. I'm doing an async_route, and afterwards sends the call with auth to a SIP server. My problem is that in the first invite I see the expected To field sip:004520202020@isp.com But when the Auto is send the To field is changed to the, To field for the PBX sip:1234@pbx.local I tried to fix this with uac_replace_to("$ru"), but that ends up in this To field: sip:004520202020@isp.comsip:004520202020@isp.com It seems like uac_replace_to sometimes appends rather that replaces. If I run it twice, then it for sure appends. Is this expected behaviour. -- -------------------- 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 1

How to change Calling Line ID in cr_route
by KamDev Essa 08 May '18

08 May '18

I have cr_route working well. Only problem is that the username is sent off to the carrier and the 3 or 4 digit username appears as the callerid. Is there a module or scripts that can replace the username with a db table defined callerid. Say 202 maps to a 10 digit TN. KD

1 0

STUN
by Wilkins, Steve 08 May '18

08 May '18

Hello all, Does anyone know the current status of how to enable STUN. I am using Kamailio Ver 5.x. Thank you, -Steve

5 6

Kamailio v5.1.3 Released
by Daniel-Constantin Mierla 07 May '18

07 May '18

Hello, Kamailio SIP Server v5.1.3 stable release is out. This is a maintenance release of the latest stable branch, 5.1, that includes fixes since the release of v5.1.2. 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 v5.1.3. For more details about version 5.1.3 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2018/04/kamailio-v5-1-3-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 - May 14-16, 2018 - www.kamailioworld.com

3 3

Determining next hop for any SIP message
by George Diamantopoulos 07 May '18

07 May '18

Hello all, I've been trying to figure out a cleaner way to determine the next hop for a SIP message, mainly for use within the NATMANAGE route for multi-homed kamailio instances (with three or more network interfaces on the kamailio host). So far I have achieved this with a series of nested ifs, depending on whether the message is a request or a response and by calculating the next hop based on the various headers (R-URI, Route) and variables ($T_req($Ri), $dd, ) involved in SIP routing. A simpler way to do it, of course, would be to use the onsend_route, but that would most likely introduce an unnecessary overhead for all routed messages. I recently noticed there a pseudovariable called $nh(key), and I believe I can use $nh(d) to the same effect. I understand however, that this works for requests only. Also, the description of this PV in the documentation reads as follows: Return attributes of next hop for the SIP request. Address is taken from > dst_uri, if set, if not from new r-uri or original r-uri. > What is not clear to me is if this covers in-dialog requests with the Route header set as well. Does the inclusion of a route header set the dst_uri PV? And if yes, is it safe to rely on dst_uri during request processing or is it set only after completion of script processing? Lastly, is there an analogue for SIP responses? If not, is it safe to rely on first Via header to determine next hop for responses, or are there any other corner cases I need to heed? Thanks in advance, BR, George

3 6

Server-side NAT traversal with Kamailio
by Alex Balashov 07 May '18

07 May '18

Recent repeated questions on this topic have inspired a blog post: http://blog.csrpswitch.com/server-side-nat-traversal-with-kamailio-the-defi… This is a rough cut, and may be edited or expanded later. -- 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/

3 5

Route header in local reply
by Евгений Голей 07 May '18

07 May '18

Hi. Colleagues write how to configure the local responses to requests so that the response added the header of the Route in accordance with the record-route headers that were in the request ? Example SBC --> Kamailio REGISTER sip:domain.com SIP/2.0 Record-Route: <sip:AAA.AAA.AAA.AAA:6060;lr> Via: SIP/2.0/UDP CCC.CCC.CCC.CCC;branch=z9hG4bKae7f.474bd538e793b8bc912ce7d0c8e0bae8.0 Via: SIP/2.0/UDP S.s.s.s;branch=z9hG4bKae7f.d81efe54000000000000000000000000.0 To: <sip:000143949@domain.com> From: <sip:000143949@domain.com>;tag=d55866b28e60059575f1fd1facc53fdd-516b CSeq: 10 REGISTER Call-ID: f74322050fd19c19f5d0ce763cc7dbdb Max-Forwards: 70 Content-Length: 0 User-Agent: SBC Expires: 0 Kamailio --> SBC SIP/2.0 200 OK Record-Route: <sip:AAA.AAA.AAA.AAA:6060;lr> Via: SIP/2.0/UDP CCC.CCC.CCC.CCC;branch=z9hG4bKae7f.474bd538e793b8bc912ce7d0c8e0bae8.0 Via: SIP/2.0/UDP S.s.s.s;branch=z9hG4bKae7f.d81efe54000000000000000000000000.0 To: <sip:000143949@domain.com>;tag=550bc1454fc5070bdadb084c686d99a2.ac18 From: <sip:000143949@domain.com>;tag=d55866b28e60059575f1fd1facc53fdd-516b CSeq: 10 REGISTER Call-ID: f74322050fd19c19f5d0ce763cc7dbdb Server: kamailio (5.1.0 (x86_64/linux)) Content-Length: 0 -- Evgeniy

1 1

How long time does a push take?
by Kjeld Flarup 07 May '18

07 May '18

Hello I have a question to those of You who uses push via Google or Apple to initiate calls. How long time does it usually take, before the phone gets the push and wakes up? Our experience is, that it can take from 2 to 10 seconds, and that not impressing when we are talking about telephony. -- -------------------- 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

4 5

Pleasing all kinds of UACs for RTP/SRTP with rtpengine
by Sergiu Pojoga 07 May '18

07 May '18

In continuation of previous <https://marc.info/?l=sr-users&m=150730146302404&w=2>thread about none/optional/compulsory SRTP handling... I used the suggested approach of *t_reuse_branch()* in a *tm:branch-failure* detected by t_check_status("(415)|(488)") replies, with the only difference of me doing it vice-versa from the suggested mantra of offer SRTP first then fallback to RTP. It works... well, for some UACs, not so with others. The 'good' UACs happily accept an updated branch, like for e.g. Yealink, will reply with "*488 Not Acceptable Here*" at first followed by a *200 OK*. The 'bad' UACs, like for e.g. Bria || Zoiper, will issue a "*415 Unsupported Media Type*" at first, and then a "*482 Merged Request*" after receiving and updated branch with a different RTP profile. Rightfully so, they decided that the updated branch is a retransmission, since both branches have the same CSeq, Call-ID and From tag. The branch ID however differs, so does the RTP profile obviously. *Any suggestion on how to convince all UACs that the 2nd attempt is different than the 1st one?* According to RFC 3261, a request is a merged request if ... "8.2.2.2 Merged Requests If the request has no tag in the To header field, the UAS core MUST check the request against ongoing transactions. If the From tag, Call-ID, and CSeq exactly match those associated with an ongoing transaction, but the request does not match that transaction (based on the matching rules in Section 17.2.3), the UAS core SHOULD generate a 482 (Loop Detected) response and pass it to the server transaction."

2 2

TOPOS (5.1.[23]) breaks PRACK
by Daniel Tryba 07 May '18

07 May '18

Enabling topos breaks PRACK in my simple (near default) proxy setup (attached kamailio.conf). asterisk 13 (10.0.3.147) <-> kamailio (10.0.3.87) <-> asterisk 13 (10.0.3.154) With the event route topos:msg-outgoing dropping all. The flow of messages is as expected. INVITE -> <- Ringing PRACK -> <- 200 OK With topos enabled, the PRACK doesn't get past route[WITHINGDLG] and is answered with the sl_send_reply("404","Not here"); INVITE -> <- Ringing PRACK -> <- 404 Not here These specific tests were done on kamailio 5.1.2, but with 5.1.3 I see the exact same messages.

2 1

Need help with NAT Traversal - Changing INVITE OK contact to IP port to Contact in Location
by KamDev Essa 06 May '18

06 May '18

The stock cfg states definng WITH_NAT and alls good. I did exactly that. However boing with stock cfg and WITH_NAT, the Registration info in location table is perfect with the UAs external IP address and port. This is not the case with the OK that in response to an INVITE. In the Contact of the OK the private IP shows up and Kamailio just sends it off to the Carrier who obviously cannot send a ack to a private IP. Is there a step by step guide for NAT Traversal? H

1 0

log CDRs to text file
by eyas barhouk 06 May '18

06 May '18

Hello Dears I'm trying to get Kamailio to log its CDRs (collected By ACC module ) to text file instead of a database so i used modparam("acc", "db_url", "flatstore:/var/log/acc") and in each time kamailio trying to write on the text file i get the error : " db_flatstore flat_new_connection() : not a directory" so could any one guide me about how i have to create the database text file "acc"? and what is the cause of the error i got ?

2 2

Carrier IP on softphone call history !!!
by KamDev Essa 04 May '18

04 May '18

I have a simple call flow for Inbound call going to an extendion carrier call ===> WITH_ALIASDB ===> extension (on Xlite) Works perfectly getting the call. Issue : - On Xlite I see the call as <TN>@<carrierIP>. I dont want my customers to know what carrier I use and also I want to be in the loop for CDR. - Hanging on either phone will not send a BYE to my server. How do I get Kamailio to be in the SIP ladder for this call? KD

2 7

Private IP in Contact Header of 200 OK back to Carrier, so no ACK - Call drop
by KamDev Essa 04 May '18

04 May '18

Scenario Carrier ===> Kamailio ===> Ext on domainINVITE ===> ====> 200 OK <== <=== Repeated because carrier pcap shows private 192.168 IP in Contact header. cfg has WITH_NAT setup. I assumed that all I need to do. What more is required to get Kamilio IP and port in 200K to Carrier? K

2 2

Where do I get rpc documentation for Kamailio
by KamDev Essa 04 May '18

04 May '18

I would like to perform subscriber and domain adds and reloads programmatically. Where can I find info on kamailio APIs I could use? KD

2 2

Kamilio Hunting 101
by KamDev Essa 04 May '18

04 May '18

With alias behind me I want to tackle a simple sequential and simulring (parallel). I see docs on using lookup() for simul and t_load_contacts but I don't see the basics on what table needs to be populated to get this working. Test set is basic : 1. 1234567890 from carrier2. Hunt to 201(a)mytestdomain.com, 202(a)mytestdomain.com. These are both rgistered. What tables do I populate or what CLI tools do I use to get this working. Thanks KD

1 0

How does one populate the dbaliases table and how do we initiate a reload.
by KamDev Essa 04 May '18

04 May '18

How do I add entries into the dbaliases table and how do I reload without restating Kamailio. KD

2 3

MSILO: Message limit on SIP stored offline MESSAGE delivery
by Abdul Basit 04 May '18

04 May '18

Hi Team, I have setup IM server with kamailio using MSILO module. Its working fine now. I have a scenario where lot of offline messages are being stored in mysql database for a user B. As soon as destination (B) REGISTER with kamailio, these messages will deliver to him. Now, kamailio will deliver all of these messages that will cause user (B) device (softphone on mobile) stuck or freezs for the moment messages are being delivered like bombardment from kamailio. Is there any way, where I can define batch size of messages to be delivered after provided interval? like batch_size=10 messages and batch_interval= 5sec, etc. I want to avoid applying static limit to mysql query in module source code. Please advise. -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445

1 1

kamailio 5.1 keepalive module + usrloc
by Aidar Kamalov 04 May '18

04 May '18

Hi, can I with keepalive module ping with is_alive function endpoints connected to kamailio? Or this module can use only static destionations via modparam("keepalive", "destination", "sip.provider.com")? -- Aydar A. Kamalov

1 0

Relay Latency
by Daniel Greenwald 03 May '18

03 May '18

We are using kamailio 5.0 as an inbound dispatcher. Sometimes in testing we see it take over half a second to relay a 200 OK. The logs show the new dialog being creating for the 200 OK within .0001 seconds of receiving the inbound 200 but it takes another .5 seconds for the relayed 200 to actually hit the wire. Here is a screenshot showing the timing of the packets. The box is an EC2 m3.medium under almost zero load (1-2 calls). Is this normal? What should I be looking at? Screenshot below: 10.0.23.34 is the kamailio box. Carrier IP obfuscated. You can see the carrier resends the 200 OK because of the delay.

3 6

Kamailio with dispatcher not send BYE to Asterisk.
by Jeferson Oliveira 03 May '18

03 May '18

Hello everyone, I have an error that I have not yet been able to solve and would like the help of colleagues to indicate a correct path. The problem that is occurring is that when the client disconnects the call kamailio is not sending the BYE forward until arriving at the asterisk. Both in the test scenario and in the production scenario the problem is the same and the message I see in the capture is 404 Not here, msg this coming from kamailio. Production scenario. PSTN <----------> Dialer --------->kamailio -----------> asterisk1 -----------> asterisk2 Test scenario. sipp generated calls ------> kamailio -------> asterisk1 -------> asterisk2 When this occurs, the calls that are disconnected by the client are in a "zombie" state in asterisk, and end up being terminated by timeout with the following message in the asterisk CLI: /[Apr 25 17:49:59] WARNING[2121]: chan_sip.c:4072 retrans_pkt: Retransmission timeout reached on transmission 22-6073(a)10.110.7.242 for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions// //Packet timed out after 31999ms with no response/ In the sipp panel I see in the Retransmission column several incrementing counters, as per the attachment. If I take the kamailio from the move and point the sipp to only one of the asterisk, the retransmissions do not happen and BYE follows normally. My kamailio.cfg configuration file can be downloaded from this url: https://drive.google.com/file/d/1bBj4GEZSPrp1iJXSLWQ4Z6g59tEFjnNT/view?usp=… Thank you very much. --

4 8

Record-Route and received/rport parameters
by Asgaroth 03 May '18

03 May '18

Hi All, I've come accros a scenario where a proxy is sitting on a private address range and inserts a record-route specifying the the private address. This causes issues whereby the BYE to an INVITE attempts to relay to the private address defined in record-route. I was wondering, if we were inject a received and rport parameter into the record-route header of the original invite, whould kamailio relay the response (BYE) to the receive/rport destination instead of the uri defined in the record-route header. As an example, initial invite comes in with a record-route as follows: Record-Route: <sip:172.17.0.2:5062;lr;ftag=b4551d29> If we injected received and rport as follows: Record-Route: <sip:172.17.0.2:5062;lr;ftag=b4551d29;rport=33429;received=212.172.2.212> and relayed the message to the B2B. Then, I assume, when the B2B creates its BYE message, the Route header should look like this: Route: <sip:172.17.0.2:5062;lr;ftag=b4551d29;rport=33429;received=212.172.2.212> Once this hits the kamailio instance to relay to the last route header as mentioned above, would it set $du to received:rport like it does with Via headers? Thanks

2 4

DMQ and/or Database for dialogs, registrations, etc..
by Joel Serrano 03 May '18

03 May '18

Hi all, Just wanted to know what your opinions were on using DMQ modules over database for things like dialog replication, registrations, etc... Is DMQ the "new way to go"? I know that there lots of ways of doing things with each having pros/cons... But I was wondering... What does the community think on this topic? Are you guys taking advantage of the DMQ modules or are you still relying on database as much as possible? Maybe a combination of both? Cheers, Joel.

8 19

Seeing Large Waits
by Mack Hendricks 03 May '18

03 May '18

Hey All, We are load testing Kamailio 4.2.3 and we are seeing the waiting within the transaction manager increasing. I’ve increased the children from 8 to 16 to 24, but we are still seeing a large amount of waiting. I don’t see an errors in the logs. Can anyone give me a clue on where to look. 05-02-18_03:07:38 PM tm.stats { current: 390 waiting: 226 total: 2361 total_local: 0 replied_locally: 1406 6xx: 0 5xx: 0 4xx: 88 3xx: 0 2xx: 3180 created: 2361 freed: 1971 delayed_free: 0 } Mack Hendricks / Head of Support / dOpenSource web: http://dopensource.com <http://dopensource.com/> support: +888-907-2085 dSIPRouter <http://dsiprouter.org/> - GUI focused on implementing Kamailio to provide SIP Trunking and PBX Hosting Services

4 6

BLF+Asterisk+NAT
by John Bob 03 May '18

03 May '18

Phone sends Subscribe Kamailio Responds 202 OK Kamailio Responds with NOTIFY Via: SIP/2.0/TCP 66.66.66.66:5051;branch=z9hG4bKfe3e.e4fedb84000000000000000000000000.0 To: <sip:30041@66.66.66.66 <mailto:30041@66.66.66.66>>;tag=48ae253ab7 From: <sip:47701@66.66.66.66 <mailto:47701@66.66.66.66>>;tag=00c39aabf419ddeab4250cd65af8ef90-6fb2 CSeq: 2 NOTIFY Call-ID: 1493ef52c41b6c1a Content-Length: 0 User-Agent: kamailio (5.1.2 (x86_64/linux)) Max-Forwards: 70 Event: dialog Contact: <sip:192.168.1.30;transport=udp> Subscription-State: terminated;reason=timeout In that packet I see Subscription-state: terminated;reason=timeout Im not sure what I am missing or misconfigured. Any help would be great. I’ve attached my config. Thanks.

2 7

Seeing Large Waits
by Mack Hendricks 02 May '18

02 May '18

Hey All, We are load testing Kamailio 4.2.3 and we are seeing the waiting within the transaction manager increasing. I’ve increased the children from 8 to 16 to 24, but we are still seeing a large amount of waiting. I don’t see an errors in the logs. Can anyone give me a clue on where to look. 05-02-18_03:07:38 PM tm.stats { current: 390 waiting: 226 total: 2361 total_local: 0 replied_locally: 1406 6xx: 0 5xx: 0 4xx: 88 3xx: 0 2xx: 3180 created: 2361 freed: 1971 delayed_free: 0 } Mack Hendricks / Head of Support / dOpenSource web: http://dopensource.com <http://dopensource.com/> support: +888-907-2085 dSIPRouter <http://dsiprouter.org/> - GUI focused on implementing Kamailio to provide SIP Trunking and PBX Hosting Services

1 0

how to disable sip authentication
by BIPIN KUMAR Das 02 May '18

02 May '18

Hi, Is there any way in Kamailio where I can suppress sip registration authentication. I want, Kamailio can send 200 OK for sip registration without going for 401 Unauthorized. Please tell me if you have such configuration in Kamailio. thanks Bipin

2 1

Kamailio config file String Concatenation
by George Diamantopoulos 02 May '18

02 May '18

Hello, I'm a a bit confused as to when string concatenation will work in kamailio config file. So far I have identified the following cases: Working: As part of the right hand side of an assignment: $var(x) = "string1" + "string2"; String argument of some functions: xlog("L_ERR", "string1" + "string2" + DEFINED_STRING + '\n'); Conditional Statements: if ( ru =~ "$" + "sip:" + $var(user) ) { Not Working: In string transformations (the fact that quotes is not part of the argument should give a hint though: $var(x) = $(var(y){s.replace,match,"repl1" + "repl2"}); String argument of some other functions: append_hf_value("P-Asserted-Identity", "<" + "$var(new_pai)" + ">"); lookup("location", "sip:$var(username)@" + SIP_REALM_DEFINITION)); In module parameter definitions: modparam("nathelper", "sipping_from", "sip:keepalive@" + SIP_REALM_DEFINITION) So far, I have found this to be fairly consistent, except for use within functions. For example, string concatenation works for the second argument of xlog, but not for the string arguments of append_hf_value() or lookup(). Is there anyway to know this for each function beforehand, or is my only choice to go through trial and error here? Thanks! BR, George

3 4

Routing 10 Digit Inbound call to Registered User or VM server
by KamDev Essa 02 May '18

02 May '18

I have Kamailio 5.1 installed and running with 2 extensions, 201 and 202 registered. Both extensions can call each other perfectly. Next I am workin on inbound calls. I have an inbound call from a carrier that arrives perfectly but ends up in this area of the cfg case -3: send_reply("404", "Not Found"); exit; and sends a 404 not found. How do I : - Associate the 10 digit DID to the 201 or 202? Custom table I make or is there something native to Kamailio. - Relay the call to the 201 or 202. Thank in advance KamDev

1 0

Routing inbound call from carrier to user or voice mail
by KamDev Essa 02 May '18

02 May '18

I have Kamailio 5.1 installed and running with 2 extensions, 201 and 202 registered. Both extensions can call each other perfectly. I have an inbound call from a carrier that arrives perfectly but ends up in this area of the cfg case -3: send_reply("404", "Not Found"); exit; and sends a 404 not found. How do I : - Associate the 10 digit DID to the 201 or 202? - In the cfg, check if user is registered if - yes, relay call to user - no relay call to VM server. KamDev

2 1

Auto 100 Reply
by Duarte Rocha 02 May '18

02 May '18

Greetings, I have set the auto_inv_100() parameter from the TM module so i can get automatic "100" replies to the INVITE. However, in some cases i want to reply to the INVITE with provisional responses (181 for example) and i would like that "100" could be sent before the "181". In order to achieve this, on transactions that are going to be replied with 181 i'm turning off the parameter with t_set_auto_inv(0) and the sl_send_reply("100","Trying"). Is this the best way to do this? Thanks

2 1

I am not able to make a call from kam1 SIP Registered user to kam2 SIP Registered user.
by Suresh Talasaniya 02 May '18

02 May '18

Hello, Is it possible to configure two Kamailio servers with shared registration database? Any SIP UA can be registered on any server but it must transparently call another SIP UA which can be registered on the same or on another Kamailio server. *I configured usrloc module with below params.* ## ---------- usrloc params ------------------- #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) *modparam("usrloc", "db_mode", 3)* modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "timer_interval", 30) modparam("usrloc", "matching_mode", 1) modparam("usrloc", "cseq_delay", 10) modparam("usrloc", "fetch_rows", 3000) modparam("usrloc", "hash_size", 12) modparam("usrloc", "timer_procs", 4) modparam("usrloc", "handle_lost_tcp", 1) modparam("usrloc", "close_expired_tcp", 1) modparam("usrloc", "db_timer_clean", 1) modparam("usrloc", "skip_remote_socket", 0) modparam("usrloc", "server_id_filter", 0) modparam("usrloc", "preload", "location") #!endif I am not able to make a call from kam1 SIP Registered user to kam2 SIP Registered user. it would be appreciable if anyone helps me to resolve the issue. Thank you! -- Regards, Suresh Talasaniya. Contact : +91-9724264776 Skype : suresh.talsaniya

2 1

Nat_UAC
by Nicolas Breuer 02 May '18

02 May '18

Hello The list, What do you think of the purpose of this function Nat_Uac_test('16') 16 - Test if the source port is different from the port in the "Via" header Correct me if I'm wrong but compare the source port of the message with the source port of the via is only valid on the initial invite ? All subsequent messages or replies don't list the UAC anywhere in the via. Thanks for your replies,

2 1

Working with users using external server
by Arik Halperin 02 May '18

02 May '18

Hello, I have a project for which I need to get user data(username and password) from an external server with REST APIs. What is the best way to achieve this? Thanks, Arik Halperin

2 1

Send UPDATE to inform changes in SDP
by Alfonso Pinto 30 Apr '18

30 Apr '18

Hi guys, I'm trying to setup a Call Forward No Reply with Kamailio and I've run into this problem (that was expected): When A INVITEs B, B replies with 183 and some SDP. B doesn't answer so Kamailio triggers the timeout and sends the INVITE TO C. C replies with 200 OK and some SDP, but A side drops the call as the media session doesn't match with the expected one. To solve it, I'm trying to send to A an UPDATE before relaying the 200 OK from C. I tried to use UAC module (uac_req_send + $uac_req) but I don't see how I can UPDATE the existing dialog. uac_req_send seems to try to setup a new dialog always. Is there a way to indicate to uac_req_send that I wan't to use the existing dialog? Or maybe there is another mechanism I can use to send the UPDATE inside the dialog? Regards, Alfonso.

4 5

Issue while using jsonrpc-s module in kamailio 4.4.
by Deepen Patel 30 Apr '18

30 Apr '18

Hi All, I am configuring Kamailio version 4.4 with jsonrpc-s and xhttp module. My aim is to use Kamailio commands using RPC from a remote server. I need help to solve error. After send rpc command, I got execution error in response. I am attaching code in kamailio.cfg file for jsonrp. event_route[xhttp:request] { if(src_ip!=127.0.0.1){ xhttp_reply("403", "Forbidden", "text/html", "<html><body>Not allowed from $si</body></html>" ); exit; } if ($hu =~ "^/RPC"){ xlog("json rpc dispatch"); # xlog($jsonrpl(method)); jsonrpc_exec('{"jsonrpc": "2.0", "method": "dispatcher.reload", "id": 1}'); jsonrpc_dispatch(); # xlog($jsonrpl(code)); # xlog($jsonrpl(text)); # xlog($jsonrpl(body)); xhttp_reply("200", "OK", "text/html", "<html><body> $jsonrpl(text), $jsonrpl(body)</body></html>"); } else { xhttp_reply("200", "OK", "text/html", "<html><body>Wrong URL $hu</body></html>"); } return; } Also, I am attaching logs. which could help you. If you can dream it, we will deliver it. <https://www.facebook.com/linteltechnologies/> <https://www.linkedin.com/company/7045679/> <http://blog.lintel.in/> Deepen Patel Python Developer A 507, Ashwamegh Elegance-2, Beside Central Mall, Ahmedabad - 380015 Gujarat, India P +1(281) 205-9247 M +91-898-077-3553 E deepen(a)lintelindia.com Skype deepenpatel.lintel W <http://lintel.in>http://lintel.in The science of today is Technology of tomorrow.

1 0

nathelper issues
by Konstantin Polyakov 29 Apr '18

29 Apr '18

Hello All, Unfortunately we found two more bugs in the nathelper module in my favorite Kamailio! Oh my god! 1. nathelper doesn't send pings when db_mode=3 and db is mongodb. I will provide fix soon. The root cause is incorrect date format in get_all_db_ucontacts function in usrloc. We need to use datatime type instead string. 2. nathelper doesn't send pings if ping_nated_only=0 (default value). It looks like incorrect condition in nh_timer in nathelper. Definitely lack of testing. It is interesting how testing is organized in the Kamailio community? Best regards. Konstantin ping_nated_only

1 0

User agent is registered but location table doesn't contain its contact.
by Konstantin Polyakov 28 Apr '18

28 Apr '18

Hello All, Hello Daniel! We have met a new issue. We use db_mode=1 ( Write-Through scheme. All changes to usrloc are immediately reflected in database too ). timer_interval has default value 60 seconds. Problem description. Some user agent is registered successfully in Kamailio registrar. Its contact exists inside internal cache (ul.lookup command shows it). But location table doesn't contain this contact. After several minutes (5-6 min) it becomes available in the location table too. The issue is reproduced periodically. Most of the time everything is ok. Is that expected behavior when internal cache is used? Did somebody else observed such situation? Is there some workaround? Best regards Konstantin

2 2

SIP Registartion Sharing between two kamailio
by Suresh Talasaniya 27 Apr '18

27 Apr '18

Hello All, I have two Kamailio servers and shared the database between two Kamailio. Below are my usrloc configuration in kamailio.cfg # ----- usrloc params ----- modparam("usrloc", "db_url", DBURL) *modparam("usrloc", "db_mode", 3)* modparam("usrloc", "use_domain", MULTIDOMAIN) modparam("usrloc", "timer_interval", 30) modparam("usrloc", "matching_mode", 1) modparam("usrloc", "cseq_delay", 10) modparam("usrloc", "fetch_rows", 3000) modparam("usrloc", "hash_size", 12) modparam("usrloc", "timer_procs", 4) modparam("usrloc", "handle_lost_tcp", 1) modparam("usrloc", "close_expired_tcp", 1) modparam("usrloc", "db_timer_clean", 1) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) modparam("registrar", "gruu_enabled", 0) modparam("registrar", "received_avp", "$avp(RECEIVED)") modparam("registrar", "default_expires", 3600) modparam("registrar", "min_expires", 1200) modparam("registrar", "max_expires", 3600) modparam("registrar", "default_q", 1000) modparam("registrar", "max_contacts", 10) modparam("registrar", "retry_after", 10) modparam("registrar", "xavp_cfg", "reg") modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 0) modparam("registrar", "path_use_received", 1) modparam("registrar", "contact_max_size", 1024) modparam("registrar", "path_check_local", 1) modparam("registrar", "sock_flag", 18) modparam("registrar", "sock_hdr_name", "Sock-Info") #------------------------------------------------------------ I have two users in same domain. 1. 1000(a)dev.domain.com --> Registered on kamailio 1 2. 1001(a)dev.domain.com --> Registered on kamailio 2 I am not able to make a call from 1000(a)dev.domain.com to 1001(a)dev.domain.com and vice versa. It will be appreciable if anyone help to resolve the issue. Thank you in advance! -- Regards, Suresh Talasaniya. Contact : +91-9724264776 Skype : suresh.talsaniya

2 1

Sending Notify With Custom Event Header
by strain17 27 Apr '18

27 Apr '18

Hi All, I'm trying to send a custom NOTIFY message to a list of users that are registered with Kamailio. The procedure would be invoked via HTTP (jsonrpc/xmlrpc over HTTP). 1. Should I be looking at the t_uac_start/t_uac_dlg MI commands to accomplish this? Is this the best way to do it if the NOTIFY sent to the phones are unsolicited? 2. I tried to use the xhttp_rpc module for testing out tm.t_uac_start but kept running into 400 invalid headers. I'm not sure if there is a syntax issue. The NOTIFY example is NOTIFY sip:123@192.168.5.5 . . From:333@1.1..1\r\nTo:test@2.2.2.2 - Request URL: http://x.x.x.x/http_rpc/tm/tm.t_uac_start?arg=NOTIFY+sip%3A123%40192.168.5.… - Request Method: GET - Status Code: 400 Invalid headers Thanks for any feedback!

2 1

I need some documentation :)
by Facchini Antoine 26 Apr '18

26 Apr '18

hello here ! I'm very newbie ... I search documentation (because I guess u can't explain that here like this) for create proxy. I want redirect all SIP traffic to an Asterisk instance. I search again and again but i don't find anything...

2 1

rare crash at kamailio 5.0.1
by David Escartin 26 Apr '18

26 Apr '18

Hello all last March 26th we experienced a crash on a 5.0.1 kamailio server and a core was generated. Seems that the call involved in the core generated was doing several retransmisions of a request (expected) and when it finally forwarded the request upstream, seems fter receiving a 481 from a-side, it crashed. here you have the backtrace <154>Mar 26 19:54:03 mad-proxy-inout-1 /usr/local/kamailio/sbin/kamailio[24992]: : <core> [core/mem/q_malloc.c:469]: qm_free(): BUG: qm_free: bad pointer 0x7f247acb5878 (out of memory block!) called from tm: h_table.c: free_cell_helper(187) - aborting (gdb) bt #0 0x00000036e8a32925 in raise () from /lib64/libc.so.6 #1 0x00000036e8a34105 in abort () from /lib64/libc.so.6 #2 0x0000000000654c24 in qm_free (qmp=0x7f23b9cba000, p=0x7f247acb5878, file=0x7f2479e3cd86 "tm: h_table.c", func=0x7f2479e3d070 "free_cell_helper", line=187, mname=0x7f2479e3c780 "tm") at core/mem/q_malloc.c:471 #3 0x00007f2479d6bf16 in free_cell_helper (dead_cell=0x7f23c54fea50, silent=0, fname=0x7f2479e4da80 "timer.c", fline=651) at h_table.c:187 #4 0x00007f2479db8148 in wait_handler (ti=1503818844, wait_tl=0x7f23c54fead0, data=0x7f23c54fea50) at timer.c:651 #5 0x0000000000629578 in timer_list_expire (t=1503818844, h=0x7f23b9d38dc0, slow_l=0x7f23b9d3b0c8, slow_mark=43533) at core/timer.c:874 #6 0x00000000006299d4 in timer_handler () at core/timer.c:939 #7 0x0000000000629e73 in timer_main () at core/timer.c:978 #8 0x0000000000422d4e in main_loop () at main.c:1690 #9 0x0000000000429732 in main (argc=13, argv=0x7fff2b9f8088) at main.c:2639 (gdb) frame 4 #4 0x00007f2479db8148 in wait_handler (ti=1503818844, wait_tl=0x7f23c54fead0, data=0x7f23c54fea50) at timer.c:651 651 UNREF_FREE(p_cell); (gdb) info locals p_cell = 0x7f23c54fea50 ret = 1 (gdb) p *p_cell $1 = {next_c = 0x0, prev_c = 0x0, hash_index = 28196, label = 341555146, flags = 4132, nr_of_outgoings = 2, ref_count = {val = 0}, from = { s = 0x7f23c87c79ed "From: <sip:932687220@100.100.100.177;user=phone;noa=national>;tag=50aa4a46-co11146-INS002\r\nCall-ID: b8997821-2814-46d3-b8f7-cf690f50f640\r\nCSeq: 1114601 BYE\r\nUser-Agent: ENSR3.0.67.10-IS2-RMRG1119-RG2040"..., len = 89}, callid = { s = 0x7f23c87c7a46 "Call-ID: b8997821-2814-46d3-b8f7-cf690f50f640\r\nCSeq: 1114601 BYE\r\nUser-Agent: ENSR3.0.67.10-IS2-RMRG1119-RG2040-CPI1-CPO2612\r\nContent-Length: 0\r\n\r\n", len = 47}, cseq_n = { s = 0x7f23c87c7a75 "CSeq: 1114601 BYE\r\nUser-Agent: ENSR3.0.67.10-IS2-RMRG1119-RG2040-CPI1-CPO2612\r\nContent-Length: 0\r\n\r\n", len = 13}, to = { s = 0x7f23c87c799e "To: \"34676893111\"<sip:34676893111@100.100.100.134;user=phone>;tag=64jNUpSZK8mSp\r\nFrom: <sip:932687220@100.100.100.177;user=phone;noa=national>;tag=50aa4a46-co11146-INS002\r\nCall-ID: b8997821-2814-46d3-b8f7"..., len = 79}, method = { s = 0x7f23c87c78c8 "BYE sip:34676893111@100.100.100.134:5080 SIP/2.0\r\nVia: SIP/2.0/UDP 200.200.200.78:5060;branch=z9hG4bK2d5c0cb4e11bb1503BYE50aa466c2b8a\r\nMax-Forwards: 34\r\nRoute: <sip:100.100.100.177;lr;ftag=64jNUpSZK8mSp;d"..., len = 3}, tmcb_hl = {first = 0x7f23bd9f7818, reg_types = 131714}, wait_timer = { next = 0x0, prev = 0x0, expire = 1503818844, initial_timeout = 80, data = 0x7f23c54fea50, f = 0x7f2479db8097 <wait_handler>, flags = 513, slow_idx = 0}, uas = {request = 0x7f23c87c71d8, end_request = 0x7f23c87c8288 "\300\300\300\300", response = {activ_type = 481, flags = 128, t_active = 0 '\000', branch = 0, buffer_len = 526, buffer = 0x7f23bef07b18 "SIP/2.0 481 Call Does Not Exist\r\nVia: SIP/2.0/UDP 200.200.200.78:5060;branch=z9hG4bK2d5c0cb4e11bb1503BYE50aa466c2b8a\r\nFrom: <sip:932687220@100.100.100.177;user=phone;noa=national>;tag=50aa4a46-co11146-I"..., my_T = 0x7f23c54fea50, timer = {next = 0x0, prev = 0x0, expire = 0, initial_timeout = 0, data = 0x0, f = 0x7f2479db7b4b <retr_buf_handler>, flags = 0, slow_idx = 0}, dst = {send_sock = 0x7f247a95be78, to = {s = { sa_family = 2, sa_data = "\023\304\325\300\313N\000\000\000\000\000\000\000"}, sin = {sin_family = 2, sin_port = 50195, sin_addr = { s_addr = 1321976021}, sin_zero = "\000\000\000\000\000\000\000"}, sin6 = {sin6_family = 2, sin6_port = 50195, sin6_flowinfo = 1321976021, sin6_addr = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>, __u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {0, 0, 0, 0}}}, sin6_scope_id = 0}}, id = 0, proto = 1 '\001', send_flags = {f = 0 '\000', blst_imask = 0 '\000'}}, retr_expire = 0, fr_expire = 0}, local_totag = { s = 0x0, len = 0}, cancel_reas = 0x0, status = 481}, uac = 0x7f23c54fec58, async_backup = {backup_route = 1, backup_branch = 4294967295, blind_uac = 0, ruri_new = 1}, fwded_totags = 0x0, uri_avps_from = 0x0, uri_avps_to = 0x0, user_avps_from = 0x0, user_avps_to = 0x0, domain_avps_from = 0x0, domain_avps_to = 0x0, xavps_list = 0x0, reply_mutex = {val = 0}, reply_locker_pid = {val = 0}, reply_rec_lock_level = 0, fr_timeout = 480, fr_inv_timeout = 1920, rt_t1_timeout_ms = 650, rt_t2_timeout_ms = 4000, end_of_life = 1503819200, relayed_reply_branch = 1, on_failure = 0, on_branch_failure = 0, on_reply = 0, on_branch = 0, on_branch_delayed = 0, md5 = 0x7f23c54fea50 ""} (gdb) frame 3 #3 0x00007f2479d6bf16 in free_cell_helper (dead_cell=0x7f23c54fea50, silent=0, fname=0x7f2479e4da80 "timer.c", fline=651) at h_table.c:187 187 sip_msg_free_unsafe(rpl); (gdb) info locals b = 0x0 i = 1 rpl = 0x7f247acb4fc0 tt = 0x0 foo = 0x7f23b9e2bd90 cbs = 0x0 cbs_tmp = 0x7f23c60c16b0 __FUNCTION__ = "free_cell_helper" (gdb) p *rpl $3 = {id = 2060145040, pid = 32548, tval = {tv_sec = 72, tv_usec = 139794652378712}, fwd_send_flags = {f = 1 '\001', blst_imask = 0 '\000'}, rpl_send_flags = { f = 0 '\000', blst_imask = 0 '\000'}, first_line = {type = -347, flags = 30263, len = 32548, u = {request = {method = {s = 0x2 <Address 0x2 out of bounds>, len = 1983381159}, uri = {s = 0x5 <Address 0x5 out of bounds>, len = 2}, version = {s = 0x2b <Address 0x2b out of bounds>, len = 2057435688}, method_value = 0}, reply = {version = {s = 0x2 <Address 0x2 out of bounds>, len = 1983381159}, status = {s = 0x5 <Address 0x5 out of bounds>, len = 2}, reason = {s = 0x2b <Address 0x2b out of bounds>, len = 2057435688}, statuscode = 0}}}, via1 = 0x0, via2 = 0x7f247acb5038, headers = 0x7fff2b9f3e50, last_header = 0x4, parsed_flag = 139794576763810, h_via1 = 0x1, h_via2 = 0x7f247616eba0, callid = 0x1, to = 0x270000000b, cseq = 0x1, from = 0x7f247aa1fa28, contact = 0x7f247acb5258, maxforwards = 0x7f247acb54d8, route = 0x0, record_route = 0x7fff2b9f3e50, content_type = 0x7f247acb5318, content_length = 0x0, authorization = 0x0, expires = 0x0, proxy_auth = 0x0, supported = 0x0, require = 0x0, proxy_require = 0x0, unsupported = 0x7f247acb5258, allow = 0x0, event = 0x7f247acb53d8, accept = 0x0, accept_language = 0x0, organization = 0x0, priority = 0x0, subject = 0x0, user_agent = 0x0, server = 0x0, content_disposition = 0x0, diversion = 0x0, rpid = 0x0, refer_to = 0x0, session_expires = 0x0, min_se = 0x0, sipifmatch = 0x0, subscription_state = 0x0, date = 0x0, identity = 0x0, identity_info = 0x0, pai = 0x0, ppi = 0x0, path = 0x7f247a98b6d0, privacy = 0x0, min_expires = 0x0, body = 0x7f247acb5258, eoh = 0x7fff2b9f3e50 "@f\313z$\177", unparsed = 0x0, rcv = {src_ip = {af = 0, len = 0, u = {addrl = {0, 0}, addr32 = {0, 0, 0, 0}, addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, addr = '\000' <repeats 15 times>}}, dst_ip = {af = 3233857728, len = 0, u = {addrl = {2882400237, 0}, addr32 = {2882400237, 0, 0, 0}, addr16 = {61421, 43981, 0, 0, 0, 0, 0, 0}, addr = "\355\357ͫ", '\000' <repeats 11 times>}}, src_port = 0, dst_port = 0, proto_reserved1 = 0, proto_reserved2 = 536, src_su = {s = {sa_family = 0, sa_data = "\000\000 \266\230z$\177\000\000\030\002\000"}, sin = {sin_family = 0, sin_port = 0, sin_addr = {s_addr = 2056828448}, sin_zero = "$\177\000\000\030\002\000"}, sin6 = {sin6_family = 0, sin6_port = 0, sin6_flowinfo = 2056828448, sin6_addr = {__in6_u = {__u6_addr8 = "$\177\000\000\030\002\000\000\000\000\000\000\210T\313z", __u6_addr16 = {32548, 0, 536, 0, 0, 0, 21640, 31435}, __u6_addr32 = {32548, 536, 0, 2060145800}}}, sin6_scope_id = 32548}}, bind_address = 0x7f247616ebb0, proto = 32 ' '}, buf = 0x7f247616eba6 "rtpengine", len = 139, new_uri = {s = 0xf0f0f0f0 <Address 0xf0f0f0f0 out of bounds>, len = 2060145720}, dst_uri = { s = 0x20 <Address 0x20 out of bounds>, len = 2060144576}, parsed_uri_ok = 1, parsed_uri = {user = { s = 0x7f247637fead "7:packetsi32e5:bytesi1896e6:errorsi0eeee5:flagsl11:initialized4:send4:recveeeee6:totalsd3:RTPd7:packetsi4951e5:bytesi158432e6:errorsi0ee4:RTCPd7:packetsi32e5:bytesi1896e6:errorsi0eee6:result2:oke", len = 2}, passwd = { s = 0x7f247637feaf "packetsi32e5:bytesi1896e6:errorsi0eeee5:flagsl11:initialized4:send4:recveeeee6:totalsd3:RTPd7:packetsi4951e5:bytesi158432e6:errorsi0ee4:RTCPd7:packetsi32e5:bytesi1896e6:errorsi0eee6:result2:oke", len = 7}, host = {s = 0x900000002 <Address 0x900000002 out of bounds>, len = 1}, port = { s = 0x7f247acb5038 "\004", len = 0}, params = {s = 0x0, len = 2060145336}, sip_params = {s = 0x7fff2b9f3e50 "@f\313z$\177", len = 2}, headers = { s = 0x7f247637feb6 "i32e5:bytesi1896e6:errorsi0eeee5:flagsl11:initialized4:send4:recveeeee6:totalsd3:RTPd7:packetsi4951e5:bytesi158432e6:errorsi0ee4:RTCPd7:packetsi32e5:bytesi1896e6:errorsi0eee6:result2:oke", len = 4}, port_no = 0, proto = 0, type = ERROR_URI_T, flags = 0, transport = { s = 0x400000001 <Address 0x400000001 out of bounds>, len = 32}, ttl = {s = 0x7f247acb5038 "\004", len = 0}, user_param = {s = 0x0, len = 2060145432}, maddr = {s = 0x7fff2b9f3e50 "@f\313z$\177", len = 1}, method = { s = 0x7f247637feba "5:bytesi1896e6:errorsi0eeee5:flagsl11:initialized4:send4:recveeeee6:totalsd3:RTPd7:packetsi4951e5:bytesi158432e6:errorsi0ee4:RTCPd7:packetsi32e5:bytesi1896e6:errorsi0eee6:result2:oke", len = 2}, lr = { s = 0x7f247637febc "bytesi1896e6:errorsi0eeee5:flagsl11:initialized4:send4:recveeeee6:totalsd3:RTPd7:packetsi4951e5:bytesi158432e6:errorsi0ee4:RTCPd7:packetsi32e5:bytesi1896e6:errorsi0eee6:result2:oke", len = 5}, r2 = {s = 0x700000002 <Address 0x700000002 out of bounds>, len = 1522079939}, gr = { s = 0x7f247acb5038 "\004", len = 0}, transport_val = {s = 0x0, len = 2060145528}, ttl_val = {s = 0x7fff2b9f3e50 "@f\313z$\177", len = 2}, user_param_val = { s = 0x7f247637fec1 "i1896e6:errorsi0eeee5:flagsl11:initialized4:send4:recveeeee6:totalsd3:RTPd7:packetsi4951e5:bytesi158432e6:errorsi0ee4:RTCPd7:packetsi32e5:bytesi1896e6:errorsi0eee6:result2:oke", len = 6}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x600000001 <Address 0x600000001 out of bounds>, len = 1896}, lr_val = {s = 0x7f247acb5038 "\004", len = 0}, r2_val = {s = 0x0, len = 2060145624}, gr_val = {s = 0x7fff2b9f3e50 "@f\313z$\177", len = 1}}, parsed_orig_ruri_ok = 1983381191, parsed_orig_ruri = {user = {s = 0x2 <Address 0x2 out of bounds>, len = 1983381193}, passwd = { s = 0x6 <Address 0x6 out of bounds>, len = 2}, host = {s = 0x0, len = 2060144696}, port = {s = 0x0, len = 0}, params = {s = 0x7f247acb54d8 "\002", ---Type <return> to continue, or q <return> to quit--- len = 731856464}, sip_params = {s = 0x0, len = 0}, headers = {s = 0x0, len = 0}, port_no = 49344, proto = 49344, type = ERROR_URI_T, flags = 2882400237, transport = {s = 0x0, len = 0}, ttl = {s = 0x218 <Address 0x218 out of bounds>, len = 2060144520}, user_param = {s = 0x218 <Address 0x218 out of bounds>, len = 2060146440}, maddr = {s = 0x7f247616ebb0 "rtpengine: bencode.c", len = 1981214752}, method = {s = 0x7f247616eba6 "rtpengine", len = 139}, lr = { s = 0xf0f0f0f0 <Address 0xf0f0f0f0 out of bounds>, len = 2060146360}, r2 = {s = 0x20 <Address 0x20 out of bounds>, len = 2060145216}, gr = { s = 0x2 <Address 0x2 out of bounds>, len = 1983381199}, transport_val = {s = 0x3 <Address 0x3 out of bounds>, len = 0}, ttl_val = {s = 0x0, len = 1}, user_param_val = {s = 0x0, len = 2060144696}, maddr_val = {s = 0x0, len = 0}, method_val = {s = 0x0, len = 731856464}, lr_val = { s = 0x1 <Address 0x1 out of bounds>, len = 1983381205}, r2_val = {s = 0x2 <Address 0x2 out of bounds>, len = 1983381207}, gr_val = { s = 0x5 <Address 0x5 out of bounds>, len = 2}}, add_rm = 0x7f247acb5878, body_lumps = 0x7f247aa43300, reply_lump = 0x0, add_to_branch_s = "\000\000\000\000\000\000\000\000\230U\313z$\177\000\000P>\237+\377\177\000\000\003\000\000\000\000\000\000\000\244\353\026v$\177\000\000\001\000\000\000\000\000\000\000\240\353\026v$\177\000\000\001", add_to_branch_len = 0, hash_index = 8, msg_flags = 28, flags = 2060162240, set_global_address = { s = 0x7f247aa43300 "\004", len = 2060146168}, set_global_port = {s = 0x7f247acb5758 "\001", len = 0}, force_send_socket = 0x7fff2b9f3e50, path_vec = { s = 0x1 <Address 0x1 out of bounds>, len = 1983381213}, instance = {s = 0x3 <Address 0x3 out of bounds>, len = 1983381216}, reg_id = 11, ruid = { s = 0xe00000002 <Address 0xe00000002 out of bounds>, len = 0}, location_ua = {s = 0x7f247acb5598 "\003", len = 0}, ldv = {flow = {decoded = 0, rcv = { src_ip = {af = 2060146264, len = 32548, u = {addrl = {140733925244496, 30064771073}, addr32 = {731856464, 32767, 1, 7}, addr16 = {15952, 11167, 32767, 0, 1, 0, 7, 0}, addr = "P>\237+\377\177\000\000\001\000\000\000\a\000\000"}}, dst_ip = {af = 1983381227, len = 32548, u = {addrl = {2, 139794578931437}, addr32 = {2, 0, 1983381229, 32548}, addr16 = {2, 0, 0, 0, 65261, 30263, 32548, 0}, addr = "\002\000\000\000\000\000\000\000\355\376\067v$\177\000"}}, src_port = 4, dst_port = 0, proto_reserved1 = 0, proto_reserved2 = 2, src_su = { s = {sa_family = 6, sa_data = "\000\000P>\237+\377\177\000\000\230U\313z"}, sin = {sin_family = 6, sin_port = 0, sin_addr = {s_addr = 731856464}, sin_zero = "\377\177\000\000\230U\313z"}, sin6 = {sin6_family = 6, sin6_port = 0, sin6_flowinfo = 731856464, sin6_addr = {__in6_u = { __u6_addr8 = "\377\177\000\000\230U\313z$\177\000\000\000\000\000", __u6_addr16 = {32767, 0, 21912, 31435, 32548, 0, 0, 0}, __u6_addr32 = { 32767, 2060146072, 32548, 0}}}, sin6_scope_id = 0}}, bind_address = 0x0, proto = 88 'X'}}}} (gdb) (gdb) bt full #0 0x00000036e8a32925 in raise () from /lib64/libc.so.6 No symbol table info available. #1 0x00000036e8a34105 in abort () from /lib64/libc.so.6 No symbol table info available. #2 0x0000000000654c24 in qm_free (qmp=0x7f23b9cba000, p=0x7f247acb5878, file=0x7f2479e3cd86 "tm: h_table.c", func=0x7f2479e3d070 "free_cell_helper", line=187, mname=0x7f2479e3c780 "tm") at core/mem/q_malloc.c:471 qm = 0x7f23b9cba000 f = 0x7f23bc87a2c8 size = 528 next = 0x400 prev = 0x7fff2b9f7920 __FUNCTION__ = "qm_free" #3 0x00007f2479d6bf16 in free_cell_helper (dead_cell=0x7f23c54fea50, silent=0, fname=0x7f2479e4da80 "timer.c", fline=651) at h_table.c:187 b = 0x0 i = 1 rpl = 0x7f247acb4fc0 tt = 0x0 foo = 0x7f23b9e2bd90 cbs = 0x0 cbs_tmp = 0x7f23c60c16b0 __FUNCTION__ = "free_cell_helper" #4 0x00007f2479db8148 in wait_handler (ti=1503818844, wait_tl=0x7f23c54fead0, data=0x7f23c54fea50) at timer.c:651 p_cell = 0x7f23c54fea50 ret = 1 #5 0x0000000000629578 in timer_list_expire (t=1503818844, h=0x7f23b9d38dc0, slow_l=0x7f23b9d3b0c8, slow_mark=43533) at core/timer.c:874 tl = 0x7f23c54fead0 ret = 0 #6 0x00000000006299d4 in timer_handler () at core/timer.c:939 saved_ticks = 1503818844 run_slow_timer = 0 i = 525 __FUNCTION__ = "timer_handler" #7 0x0000000000629e73 in timer_main () at core/timer.c:978 No locals. #8 0x0000000000422d4e in main_loop () at main.c:1690 i = 1 pid = 0 si = 0x0 si_desc = "udp receiver child=0 sock=100.100.100.169:5060", '\000' <repeats 16 times>, "\001\000\000\000\340|\237+\377\177\000\000W\327e\000\000\000\000\000\200\335u\000\000\000\000\000\250\065\313zo\000\000\000\316\357u\000\000\000\000\000\205\335u\000\000\000\000\000\004\000\000\000\000\000\000\000\000\240˹#\177---Type <return> to continue, or q <return> to quit--- \000" nrprocs = 1 woneinit = 1 __FUNCTION__ = "main_loop" #9 0x0000000000429732 in main (argc=13, argv=0x7fff2b9f8088) at main.c:2639 cfg_stream = 0x1f8f010 c = -1 r = 0 tmp = 0x7fff2b9f8742 "" tmp_len = 54 port = -398323320 proto = 54 options = 0x732e78 ":f:cm:M:dVIhEeb:l:L:n:vKrRDTN:W:w:t:u:g:P:G:SQ:O:a:A:x:X:Y:" ret = -1 seed = 3083103015 rfd = 4 debug_save = 0 debug_flag = 0 dont_fork_cnt = 0 n_lst = 0x36e8a90687 p = 0x7fff2b9f7f88 "\200\200\237+\377\177" st = {st_dev = 2307, st_ino = 9183277, st_nlink = 3, st_mode = 16832, st_uid = 522, st_gid = 501, __pad0 = 0, st_rdev = 0, st_size = 4096, st_blksize = 4096, st_blocks = 8, st_atim = {tv_sec = 1496231473, tv_nsec = 0}, st_mtim = {tv_sec = 1520405673, tv_nsec = 0}, st_ctim = { tv_sec = 1520405673, tv_nsec = 0}, __unused = {0, 0, 0}} __FUNCTION__ = "main" (gdb) i dont know if you can help us to know the reason of the crash, because the call trace seemed quite normal, and we don't know if the cause was from the rtpengine responses of the bye requests, or was caused by the request retransmisions, which would be rare because we sometimes have those type of calls. thanks a lot and regards david

2 5

ASN.1 HI2- IRI-Parameters
by Mojtaba 26 Apr '18

26 Apr '18

Hello. I need to use HI2Operations for transfering IRI-Parameters in kamailio.Dose anybody have experience in this regards? Thanks. Mojtaba

2 1

Possibility of kemi lua to keep lua global vars
by Yuriy Gorlichenko 26 Apr '18

26 Apr '18

Hi For now as I see: every request that calls lua script reevables my global lua variables For example I trying to connect to DB via luasql driver and checking if connection already existis to avoid new connection I have function init() that calls in ksr_request_route() Inside i have ome check: if not CON then CON = assert(MYSQL:connect(DBdata.dbname,DBdata.username,DBdata.password,DBdata.host,DBdata.port)) if not CON then KSR.log("info","|init|log| DB NOT connected\n") KSR.sl.sl_send_reply(503,"Internal server error") KSR.x.exit() else KSR.log("info","|init|log| DB connected\n") end end So in this case if CON was already created it should not try to connect But for now it tries for every request. So means kemi not stores global lua variables between sessions (in one kamailio process) Also as I know INIT routes initiates before lua machine will be started so it not allowed to init some lua global variables in the init routes So my question is there is a way to keep global vars per kamailio process instead per kamailio request?

2 1

Sha2 self-signed certs config in Kamilio
by Kiran Gaddam 25 Apr '18

25 Apr '18

Hello All, I want to enable certificate validation on the server. I am only using self-signed certs. I have the same cert/key in the client and server and want to only allow connection from clients with this cert/key. I have turned on the following in tls.cfg and done all the steps required in kamailio.cfg file. But it's failed to verify certs and allowing the clients which doesn't have same certs. Please help to configure the cert/key in right way. [server:default] method = TLSv1 verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/selfsigned.key certificate = /usr/local/etc/kamailio/selfsigned.pem ca_list = /usr/local/etc/sip-router/cacert.pem [client:default] verify_certificate = yes require_certificate = yes In advance Thank you. Thanks, Kiran

2 2

TLS support and certificate validation
by Jignesh Gandhi 24 Apr '18

24 Apr '18

I want to enable certificate validation on the server. I am only using self-signed certs. I have the same certificate/key in the client and server and want to only allow connection from clients with this cert/key. I have turned on the following in tls.cfg and done all the steps required in kamailio.cfg file. [server:default] method = TLSv1 verify_certificate = yes require_certificate = yes private_key = /usr/local/etc/kamailio/selfsigned.key certificate = /usr/local/etc/kamailio/selfsigned.pem ca_list = /usr/local/etc/sip-router/cacert.pem [client:default] verify_certificate = yes require_certificate = yes Any help is appreciated. Thanks, --Jignesh

2 1

TCP packet relay delayed (Kamailio 4.1.5)
by Andrew Chen 24 Apr '18

24 Apr '18

Hi Kamailio community, I need your urgent help regarding a situation where the Kamailio TCP packet delayed in relaying SIP packets. They delay between the time Kamailio logs the relay to the time it left the system was ~20 seconds difference, which is huge in SIP world. Any ideas, thoughts or suggestions on how I can narrow down to the problem? We have about 30 cases of this today Thanks in advance. -- Andy Chen Sr. Telephony Lead Engineer achen@ <achen(a)thinkingphones.com>fuze.com -- *Confidentiality Notice: The information contained in this e-mail and any attachments may be confidential. If you are not an intended recipient, you are hereby notified that any dissemination, distribution or copying of this e-mail is strictly prohibited. If you have received this e-mail in error, please notify the sender and permanently delete the e-mail and any attachments immediately. You should not retain, copy or use this e-mail or any attachment for any purpose, nor disclose all or any part of the contents to any other person. Thank you.*

3 7

Diversion Headers
by Duarte Rocha 24 Apr '18

24 Apr '18

Greetings, Does the function "add_diversion() present in the "Diversion" module allows me to introduce other parameters (privacy, counter, etc) or is it only for the "diversion-reason"? If it doesn't let me do that, how can I add the Diversion with insert_hf() and make sure the headers are added in the right order? Cheers

2 1

Any ideas on how to implement prepaid calling function which is persistent after restart
by Kit Lim 24 Apr '18

24 Apr '18

Hi Everyone, My project involves keeping track of the allowable duration for the caller to talk, and keeping track of the status of the call with the back end database. Currently i have implemented using the following modules (aside from the default modules in the sample config script). * http_async_client.so - to communicate with the backend DB * dialog.so - mainly use to detect the 200 ok answer in relation to invite, and the end of the call * cnxcc.so - to release the call when the allowable talk time is reached The flow is as below:- Invite -- > http to backend DB to ask for talk time <-- DB returns value 200 ok (answer) ---> http backend DB to update that the call has answered <-- DB returns ok dialog end or Cnxcc terminate --> http backend DB to update that the call has hangup My problem is that if there is a need to restart the kamailio config there is no 'graceful' way to hangup all the calls and inform the backend, or deny any new calls from coming in. If kamailio is restarted then cnxcc would have forgotten that there is a maximum talk time allowed to be called or recover from the restart. How to I make the system resilience to failures/restarts. Appreciate any good suggestions. regards CK

2 1

Re: [SR-Users] Kamailio doesn't start with db_mongodb and usrloc.
by Konstantin Polyakov 24 Apr '18

24 Apr '18

Dear All, It looks like there is a serious error inside db_mongodb module in the latest versions of Kamailio 4.4.7 and 5.1.2. During query operations string fields inside result structure db1_res_t could be corrupted because of incorrect usage of bson memory. The issue is located in the function db_mongodb_convert_bson() around line 614. This issue is reproduced when we need to read more than 150 documents from the database. So that is why this issue still exists. Firstly bson_iter_utf8 function returns pointer to memory which is allocated inside bson memory and which is used multiple times. So the data must be copied into Kamailio memory instead storing that pointer. The pointer becomes invalid after some time. Secondly, this pointer must not be freed by the caller of bson_iter_utf8 . At the current version bson allocated pointer is freed by Kamailio function pkg_free like it is was allocated by pkg_malloc . So I am going to create ticket in the github and propose the fix. It looks like nobody uses mongodb with Kamailio in db_mode=1 :) Best regards Konstantin >---------------------------------------------------------------------- > >Message: 1 >Date: Sat, 21 Apr 2018 23:37:26 +0300 >From: Konstantin Polyakov < piligrim_pk(a)mail.ru > >To: sr-users(a)lists.kamailio.org >Subject: [SR-Users] Kamailio doesn't start with db_mongodb and usrloc. >Message-ID: < 1524343046.238272820(a)f356.i.mail.ru > >Content-Type: text/plain; charset="utf-8" > > >Hello! > >Similar issue but with different symptoms. > >kamailio 4.4.7 (tried to use db_mongodb module from 5.1.2) >usrloc is configured to use db_mode=1 and db_insert_null=1. > >MongoDB 3.6.3 >mongo-c-driver was tested 1.9.2 and 1.9.4. > >location table contains 300 records. > >The issue was not reproduced since February when we started a new feature. Now it is reproduced constantly, but on some machines. > >Logs: > >[0;39;49m[0;39;49m 1(13958) DEBUG: usrloc [udomain.c:313]: dbrow2info(): non-local socket <all.com>...ignoring >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:256]: dbrow2info(): bad callid >[0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:484]: preload_udomain(): skipping record for agent in table location >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:231]: dbrow2info(): bad contact >[0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:484]: preload_udomain(): skipping record for cflags in table location >[0;39;49m[0;39;49m 1(13958) DEBUG: usrloc [udomain.c:313]: dbrow2info(): non-local socket <��keepalive>...ignoring >[0;39;49m[0;35;49m 1(13958) WARNING: usrloc [../../ut.h:717]: shm_str_dup(): shm_str_dup fallback; dup called for src->s == NULL or src->len < 0 >[0;39;49m[0;35;49m 1(13958) WARNING: usrloc [../../ut.h:731]: shm_str_dup(): shm_str_dup fallback; skip memcpy for src->s == NULL >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:231]: dbrow2info(): bad contact >[0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:484]: preload_udomain(): skipping record for cseq in table location >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:231]: dbrow2info(): bad contact >[0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:484]: preload_udomain(): skipping record for 5.17.xxx in table location >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:231]: dbrow2info(): bad contact >[0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:484]: preload_udomain(): skipping record for .17.x in table location >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping >[0;39;49m[0;39;49m 1(13958) DEBUG: usrloc [udomain.c:313]: dbrow2info(): non-local socket <nce>...ignoring >[0;39;49m[0;35;49m 1(13958) WARNING: usrloc [../../ut.h:717]: shm_str_dup(): shm_str_dup fallback; dup called for src->s == NULL or src->len < 0 >[0;39;49m[0;35;49m 1(13958) WARNING: usrloc [../../ut.h:731]: shm_str_dup(): shm_str_dup fallback; skip memcpy for src->s == NULL >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping >[0;39;49m[0;39;49m 1(13958) DEBUG: usrloc [udomain.c:313]: dbrow2info(): non-local socket <c-89-5ada654b-35a2-42>...ignoring >[0;39;49m[0;39;49m 1(13958) DEBUG: usrloc [udomain.c:313]: dbrow2info(): non-local socket <>...ignoring >[0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:231]: dbrow2info(): bad contact >[0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:484]: preload_udomain(): skipping record for 38.146:5360 in table location >[0;39;49m[0;39;49m 1(13958) DEBUG: usrloc [udomain.c:313]: dbrow2info(): non-local socket <ified>...ignoring >[0;39;49m[0;31;49m 1(13958) ERROR: usrloc [ucontact.c:94]: new_ucontact(): no ruid for aor: lQ.com@5.17.60@8/Ղusername >[0;39;49m[0;31;49m 1(13958) ERROR: usrloc [urecord.c:149]: mem_insert_ucontact(): failed to create new contact >[0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:517]: preload_udomain(): inserting contact failed >[0;39;49m[0;31;49m 1(13958) ERROR: usrloc [ul_mod.c:457]: child_init(): child(1): failed to preload domain 'location' >[0;39;49m[0;31;49m 1(13958) ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module usrloc (/usr/ecsp/lib64/kamailio/modules/usrloc.so) >[0;39;49m[0;31;49m 1(13958) ERROR: <core> [pt.c:329]: fork_process(): init_child failed for process 1, pid 13958, "udp receiver child=0 sock=5.17.138.137:5060 (5.17.138.137:5060)" >[0;39;49m[0;33;49m 1(13958) CRITICAL: <core> [main.c:1615]: main_loop(): Cannot fork >[0;39;49m[1;31;49m 0(13956) ALERT: <core> [main.c:740]: handle_sigs(): child process 13958 exited normally, status=255 >[0;39;49m[0;36;49m 0(13956) INFO: <core> [main.c:768]: handle_sigs(): terminating due to SIGCHLD >[0;39;49m[0;39;49m 0(13956) DEBUG: <core> [main.c:770]: handle_sigs(): terminating due to SIGCHLD > > >Could you please help us to resolve the issue? >Thank you in advance. >Best regards >Konstantin >

1 0

UAC with HA1
by Ben Hood 24 Apr '18

24 Apr '18

Hi, Does the UAC module support ha1 passwords for the remote registrar? The DDL for the uacreg table indicates the might be the case, but it’s unclear which one the UAC module would use by default. Is there a parameter to set this? Cheers, Ben

2 1

Planning the release of Kamailio v5.1.3
by Daniel-Constantin Mierla 24 Apr '18

24 Apr '18

Hello, I am considering to package a new release from branch 5.1, respectively v5.1.3 sometime next week, likely on Tuesday or Wednesday (April 24 or 25). I any issue is not yet reported on github tracker, do it as soon as possible to have a chance to be investigated. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 1

Use of MariaDB to replace MySQL
by Ginhoux, Patrick 23 Apr '18

23 Apr '18

Hi, Currently, my Kamailio proxy uses a MySql DB to get configuration data. We have a plan to migrate MySQL to MariaDB. Is there MariaDB modules with Kamailio ? Thanks in advance for your answer. P. Ginhoux

2 2

Kamailio not listening to proper ipv6 address
by selak Dev 23 Apr '18

23 Apr '18

Hi, I need to enable Kamailio to work with ipv6 also, and I enabled auto_bind_ipv6=1 and when restarted I can see that it is listening to ipv6 ip addresses also. The problem is that it stripped the ip address at the end and it is listening to the wrong one. When I try to register I get timeout, since it is not listening the correct address. ip address of the server is 2604:a880:400:d0::yyyy:xxxx and when I execute "sudo netstat -plnt" too see which app is listening to which address/port I can see: .... tcp6 0 0 2604:a880:400:d0:::5085 :::* LISTEN 17748/kamailio tcp6 0 0 2604:a880:400:d0:::5061 :::* LISTEN 17748/kamailio It somehow stripped the last part of the address (yyyy:xxxx). I dont know if it is related to :: which would be all zeros, but it doesn't work even if i explicitly specify which address to listen with listen=[2604:a880:400:d0:0000:0000:yyyy:xxxx] or listen=tcp:[2604:a880:400:d0:0000:0000:yyyy:xxxx]:5061 or listen=tcp:[2604:a880:400:d0::yyyy:xxxx]:5061 I tried all combinations and the results are always the same... Have anybody had this problem before? Do you know how to fix this? Best regards, Marko Selakovic

3 3

select outgoing interface
by Laurent Schweizer 23 Apr '18

23 Apr '18

Dear all, I have a config with multiple interface( multiple IP in same subnet) and want to know how I can select the interface used to send a request (UDP). The best is the possibility to choose for each request the outgoing interface but if it's not possible I think I can also work with a solution where the outgoing interface is defined globally ? I need this because one provider request one IP for inbound call and one IP for outbound call . Thanks Laurent

2 2

Variables in database
by Nelson Migliaro 22 Apr '18

22 Apr '18

Hello, I am palnning to set variables to limit the number of channels but I would like to set up them in database in order to upload them without needing to restart the service. I also would like to have that db information to be loaded in memory. So far I only thought in dialplan module but i am looking for a more "fancy" way. Can someone recommend me a module? thank you

5 5

Kamailio relays ACK itself
by Volkan Oransoy 22 Apr '18

22 Apr '18

Hi all, I have a Kamailio box which I use as a uac_client. We register a bunch of accounts to remote providers. The box relays incoming calls to a FreeSwitch box. We encountered a strange problem on one account. Kamailio relays the call to my FreeSwitch box successfully but routes the ACK itself which is the reply of 200Ok. I believe this is a provider related issue but I need a technical explanation. Does anyone have any idea? Here is the packet capture: https://pastebin.com/PA8StiWY Thanks, -- Volkan Oransoy

3 2

TCP core parameters and TLS
by Alex Balashov 22 Apr '18

22 Apr '18

Hi, Is it reasonable to assume that the various tcp_* core parameters (keepalive behaviour, connection limits, buffer sizes, etc.) also regulate the behaviour of TLS connections, since TLS is encapsulated in TCP? -- 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

CLID blacklist
by Nelson Migliaro 21 Apr '18

21 Apr '18

Hello, I am looking for a module to block number based on CLID. If the module allows REG expresions it would really make my day. I need to block some callers and I would like to use a module and database to avoid restarting kamailio service everytime a new number appears. Thank you Nelson.-

2 2

Kamailio doesn't start with db_mongodb and usrloc.
by Konstantin Polyakov 21 Apr '18

21 Apr '18

Hello! Similar issue but with different symptoms. kamailio 4.4.7 (tried to use db_mongodb module from 5.1.2) usrloc is configured to use db_mode=1 and db_insert_null=1. MongoDB 3.6.3 mongo-c-driver was tested 1.9.2 and 1.9.4. location table contains 300 records. The issue was not reproduced since February when we started a new feature. Now it is reproduced constantly, but on some machines. Logs: [0;39;49m[0;39;49m 1(13958) DEBUG: usrloc [udomain.c:313]: dbrow2info(): non-local socket <all.com>...ignoring [0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping [0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:256]: dbrow2info(): bad callid [0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:484]: preload_udomain(): skipping record for agent in table location [0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping [0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:231]: dbrow2info(): bad contact [0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:484]: preload_udomain(): skipping record for cflags in table location [0;39;49m[0;39;49m 1(13958) DEBUG: usrloc [udomain.c:313]: dbrow2info(): non-local socket <��keepalive>...ignoring [0;39;49m[0;35;49m 1(13958) WARNING: usrloc [../../ut.h:717]: shm_str_dup(): shm_str_dup fallback; dup called for src->s == NULL or src->len < 0 [0;39;49m[0;35;49m 1(13958) WARNING: usrloc [../../ut.h:731]: shm_str_dup(): shm_str_dup fallback; skip memcpy for src->s == NULL [0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping [0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:231]: dbrow2info(): bad contact [0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:484]: preload_udomain(): skipping record for cseq in table location [0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:231]: dbrow2info(): bad contact [0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:484]: preload_udomain(): skipping record for 5.17.xxx in table location [0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:231]: dbrow2info(): bad contact [0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:484]: preload_udomain(): skipping record for .17.x in table location [0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping [0;39;49m[0;39;49m 1(13958) DEBUG: usrloc [udomain.c:313]: dbrow2info(): non-local socket <nce>...ignoring [0;39;49m[0;35;49m 1(13958) WARNING: usrloc [../../ut.h:717]: shm_str_dup(): shm_str_dup fallback; dup called for src->s == NULL or src->len < 0 [0;39;49m[0;35;49m 1(13958) WARNING: usrloc [../../ut.h:731]: shm_str_dup(): shm_str_dup fallback; skip memcpy for src->s == NULL [0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping [0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping [0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:476]: preload_udomain(): empty username record in table location...skipping [0;39;49m[0;39;49m 1(13958) DEBUG: usrloc [udomain.c:313]: dbrow2info(): non-local socket <c-89-5ada654b-35a2-42>...ignoring [0;39;49m[0;39;49m 1(13958) DEBUG: usrloc [udomain.c:313]: dbrow2info(): non-local socket <>...ignoring [0;39;49m[0;33;49m 1(13958) CRITICAL: usrloc [udomain.c:231]: dbrow2info(): bad contact [0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:484]: preload_udomain(): skipping record for 38.146:5360 in table location [0;39;49m[0;39;49m 1(13958) DEBUG: usrloc [udomain.c:313]: dbrow2info(): non-local socket <ified>...ignoring [0;39;49m[0;31;49m 1(13958) ERROR: usrloc [ucontact.c:94]: new_ucontact(): no ruid for aor: lQ.com@5.17.60@8/Ղusername [0;39;49m[0;31;49m 1(13958) ERROR: usrloc [urecord.c:149]: mem_insert_ucontact(): failed to create new contact [0;39;49m[0;31;49m 1(13958) ERROR: usrloc [udomain.c:517]: preload_udomain(): inserting contact failed [0;39;49m[0;31;49m 1(13958) ERROR: usrloc [ul_mod.c:457]: child_init(): child(1): failed to preload domain 'location' [0;39;49m[0;31;49m 1(13958) ERROR: <core> [sr_module.c:923]: init_mod_child(): Error while initializing module usrloc (/usr/ecsp/lib64/kamailio/modules/usrloc.so) [0;39;49m[0;31;49m 1(13958) ERROR: <core> [pt.c:329]: fork_process(): init_child failed for process 1, pid 13958, "udp receiver child=0 sock=5.17.138.137:5060 (5.17.138.137:5060)" [0;39;49m[0;33;49m 1(13958) CRITICAL: <core> [main.c:1615]: main_loop(): Cannot fork [0;39;49m[1;31;49m 0(13956) ALERT: <core> [main.c:740]: handle_sigs(): child process 13958 exited normally, status=255 [0;39;49m[0;36;49m 0(13956) INFO: <core> [main.c:768]: handle_sigs(): terminating due to SIGCHLD [0;39;49m[0;39;49m 0(13956) DEBUG: <core> [main.c:770]: handle_sigs(): terminating due to SIGCHLD Could you please help us to resolve the issue? Thank you in advance. Best regards Konstantin

1 0

question on TM transaction matching
by Kelvin Chua 19 Apr '18

19 Apr '18

i am monitoring an issue with 5.1.2 where a cancel is not being propagated after a 183 is received the packet dump doesn't show anything anomalous. some notes: 1. the invite went out through t_relay(), transaction is tracked 2. cancel is handled this way if (is_method("CANCEL")) { xlog("CANCEL received"); if (t_check_trans()) { route(RELAY); } exit; } 3. if the UAC who is sending the call to kamailio is configured to send invite with RURI without a port i.e. 14087654321(a)w.x.y.z instead of 14087654321@w.x.y.z:6050, then this does not happen 4. can be argued that this is not an issue, and that the client should just be configured properly with outbound proxy and ruri without port. but i am just being difficult here and wanted to understand what is going on with the matching algo. TM debug, in case it helps 2018-04-18T07:29:50.984652+00:00 pinkman /usr/sbin/kamailio[17670]: ERROR: <script>: 100 Trying received 2018-04-18T07:29:50.984746+00:00 pinkman /usr/sbin/kamailio[17670]: DEBUG: tm [t_reply.c:1263]: t_should_relay_response(): ->>>>>>>>> T_code=100, new_code=100 2018-04-18T07:29:50.984752+00:00 pinkman /usr/sbin/kamailio[17670]: DEBUG: tm [t_reply.c:1786]: relay_reply(): branch=1, save=0, relay=-1 icode=0 2018-04-18T07:29:53.564979+00:00 pinkman /usr/sbin/kamailio[17678]: DEBUG: tm [h_table.c:129]: free_cell_helper(): freeing transaction 0x7f820b6b8a60 from timer.c:651 2018-04-18T07:29:53.752387+00:00 pinkman /usr/sbin/kamailio[17678]: DEBUG: tm [h_table.c:129]: free_cell_helper(): freeing transaction 0x7f820b6bbcd8 from timer.c:651 2018-04-18T07:29:54.002501+00:00 pinkman /usr/sbin/kamailio[17678]: DEBUG: tm [h_table.c:129]: free_cell_helper(): freeing transaction 0x7f820b6bf130 from timer.c:651 2018-04-18T07:29:54.184438+00:00 pinkman /usr/sbin/kamailio[17671]: DEBUG: tm [t_lookup.c:1019]: t_check_msg(): msg (0x7f8212d77e28) id=3 global id=2 T start=0xffffffffffffffff 2018-04-18T07:29:54.184454+00:00 pinkman /usr/sbin/kamailio[17671]: DEBUG: tm [t_lookup.c:897]: t_reply_matching(): t_reply_matching: hash 6057 label 0 branch 1 2018-04-18T07:29:54.184468+00:00 pinkman /usr/sbin/kamailio[17671]: DEBUG: tm [t_lookup.c:951]: t_reply_matching(): reply (0x7f8212d77e28) matched an active transaction (T=0x7f820b6cc988)! 2018-04-18T07:29:54.184474+00:00 pinkman /usr/sbin/kamailio[17671]: DEBUG: tm [t_lookup.c:1088]: t_check_msg(): msg (0x7f8212d77e28) id=3 global id=3 T end=0x7f820b6cc988 2018-04-18T07:29:54.184485+00:00 pinkman /usr/sbin/kamailio[17671]: DEBUG: tm [t_reply.c:2207]: reply_received(): org. status uas=100, uac[1]=100 local=0 is_invite=1) 2018-04-18T07:29:54.184544+00:00 pinkman /usr/sbin/kamailio[17671]: ERROR: <script>: 183 Session Progress received 2018-04-18T07:29:54.184825+00:00 pinkman rtpengine[1660]: INFO: [EqRkPDiOFOOhFg_3018V8A..]: Received command 'answer' from 127.0.0.1:41084 2018-04-18T07:29:54.184918+00:00 pinkman rtpengine[1660]: INFO: [EqRkPDiOFOOhFg_3018V8A..]: answer time = 0.000074 sec 2018-04-18T07:29:54.184931+00:00 pinkman rtpengine[1660]: INFO: [EqRkPDiOFOOhFg_3018V8A..]: Replying to 'answer' from 127.0.0.1:41084 2018-04-18T07:29:54.185049+00:00 pinkman /usr/sbin/kamailio[17671]: DEBUG: tm [t_reply.c:1263]: t_should_relay_response(): ->>>>>>>>> T_code=100, new_code=183 2018-04-18T07:29:54.185066+00:00 pinkman /usr/sbin/kamailio[17671]: DEBUG: tm [t_reply.c:1786]: relay_reply(): branch=1, save=0, relay=1 icode=0 2018-04-18T07:29:54.252450+00:00 pinkman /usr/sbin/kamailio[17678]: DEBUG: tm [h_table.c:129]: free_cell_helper(): freeing transaction 0x7f820b6c2540 from timer.c:651 2018-04-18T07:29:54.377492+00:00 pinkman rtpengine[1660]: WARNING: [EqRkPDiOFOOhFg_3018V8A.. port 18598]: RTP packet with unknown payload type 95 received 2018-04-18T07:29:54.564924+00:00 pinkman /usr/sbin/kamailio[17678]: DEBUG: tm [h_table.c:129]: free_cell_helper(): freeing transaction 0x7f820b6c57a8 from timer.c:651 2018-04-18T07:29:55.627429+00:00 pinkman /usr/sbin/kamailio[17678]: DEBUG: tm [h_table.c:129]: free_cell_helper(): freeing transaction 0x7f820b6c8bf8 from timer.c:651 2018-04-18T07:29:55.939949+00:00 pinkman /usr/sbin/kamailio[17678]: DEBUG: tm [h_table.c:129]: free_cell_helper(): freeing transaction 0x7f820b6cc988 from timer.c:651 2018-04-18T07:29:56.404385+00:00 pinkman /usr/sbin/kamailio[17673]: ERROR: <script>: CANCEL received 2018-04-18T07:29:56.404411+00:00 pinkman /usr/sbin/kamailio[17673]: DEBUG: tm [t_lookup.c:711]: t_lookupOriginalT(): searching on hash entry 6057 2018-04-18T07:29:56.404424+00:00 pinkman /usr/sbin/kamailio[17673]: DEBUG: tm [t_lookup.c:435]: matching_3261(): RFC3261 transaction matching failed 2018-04-18T07:29:56.404429+00:00 pinkman /usr/sbin/kamailio[17673]: DEBUG: tm [t_lookup.c:801]: t_lookupOriginalT(): no CANCEL matching found! 2018-04-18T07:29:56.404434+00:00 pinkman /usr/sbin/kamailio[17673]: DEBUG: tm [t_lookup.c:803]: t_lookupOriginalT(): lookup completed 2018-04-18T07:29:56.404441+00:00 pinkman /usr/sbin/kamailio[17673]: DEBUG: tm [tm.c:1096]: ki_t_lookup_cancel_flags(): lookup_original: t_lookupOriginalT returned: (nil) Kelvin Chua

3 9

Info: Kamailio Unit Testing Framework using Docker
by Daniel-Constantin Mierla 19 Apr '18

19 Apr '18

Hello, a while ago, I started to put together a unit testing framework for Kamailio. It was more or less inspired from the old shell-based testing framework added by Henning Westerholt many years ago (which still resides in source tree path test/unit/), but now leveraging docker to have an isolated environment for running the tests. The new testing framework is available on github at: * https://github.com/kamailio/kamailio-tests Giacomo Vacca helped a bit so far, he is actually going to present more about it at Kamailio World 2018 (https://www.kamailioworld.com/k06/schedule/). But we want now to get more feedback from community, eventually to adjust the framework to become something easy to use and even more important, to be easy to contribute to. One of the main benefits from my point of view it would be to have the reports of the issues that can be reproduced submitted along with a unit test. It would make it easier to troubleshoot and then after fixing it, would be tested always before releases in order to avoid regressions. Right now, a good help from community would represent converting the old unit tests to the new framework, afterwards we can decommission test/unit. The conversion should be rather easy, as we still rely on shell and the sip tools like sipp/sipsak ... If you want to help here by you need clarifications or get stuck somewhere, just write to mailing list and I would be more than happy to assist. Such contributions should be submitted as pull requests in order to be easy to review: * https://github.com/kamailio/kamailio-tests/pulls Looking forward to your feedback, comments or contributions! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com

1 0

Help needed to compile and install kamailio 5.1.2
by vinay kumar 18 Apr '18

18 Apr '18

Hi All, I want to compile kamailio 5.1.2 for debian packages amd install on ubuntu 16.04. And i have couple of queries. 1. can i compile and run kamailio 5.1.2 on ubuntu 16.04(Xenial)? 2. Im currently using kamailio 4.3.6 with polaris configuration. And the same way can i use kamailio 5.1.2 with polaris configuration files and scripts to bringup kamailo 5.1.2? and if yes, please share me steps and scripts to bringup kamailio 5.1 2 with Open HSS( i need to use CX and RX interface). Please help me out. Thanks in advance. Regards, vinay

2 1

Kamailio support for IPv6 VoLTE Calls
by Sairam Subramanian 18 Apr '18

18 Apr '18

Hi All, Can anyone tell me which version of the Kamailio Open IMS supports IPv6 VoLTE Calls? Thanks, Sairam S

1 1

Branching and rejects.
by Kjeld Flarup 18 Apr '18

18 Apr '18

Hello What is the correct way to handle, if a branched call reaches a 404 on one of the branches. Currently I discard all 404 in onreply_route, but if all branches has 404, then this is never send to the caller. -- -------------------- 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 2

Failure Route Issue
by mack＠dopensource.com 18 Apr '18

18 Apr '18

Hey All, I have a weird issue. I’m sending a SIP INVITE to a SIP Service that send’s me back a 433, which let’s me know that the number is invalid. I then send an INVITE to a media server (FreeSwitch) to play a message that the number can’t be called, which looks to be working. But, the final ACK from the SIP client is sent back to the SIP Service versus the media server. The following diagram below depicts this: https://snag.gy/qMWfvk.jpg. The code snippet in the failure_route for the the SIP Service has this: if (t_check_status("403|433")) { if ($avp(s:teleblock_media_enabled) == "1") { # make sure media server can route back to kamailio record_route(); revert_uri(); $ru = "sip:" + $T_reply_code + $rU + "@" + $sel(cfg_get.teleblock.media_ip) + ":" + $sel(cfg_get.teleblock.media_port); append_branch(); t_relay(); exit; }

2 5

Kamailio Websocket Questions
by Nagorny, Dimitry 17 Apr '18

17 Apr '18

Hi List, two questions regarding Kamailio Websockets: 1. How do I send INVITE request TO the listening Websocket? I usually used for UDP/TCP/TLS these format, but somehow this is not working for Websocket because it gets in an endless loop: if ($rU=~"^(\+|00|0)?[1-9][0-9]{4,20}$") { $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ";user=phone;transport=udp"; force_send_socket(udp:MY_EXTERN_IP:MY_EXTERN_PORT); rtpengine_offer("internal trust-address RTP AVP"); route(RELAY); exit; } 2. Why do I get these errors every time my Websocket Client is registering, but it still works and DB is inserted with registered location: ERROR: <core> [parser/parse_fline.c:257]: parse_first_line(): parse_first_line: bad message (offset: 22) ERROR: <core> [parser/msg_parser.c:690]: parse_msg(): ERROR: parse_msg: message=<HTTP/1.1 101 Switching Protocols#015#012Sia: SIP/2.0/TCP 10.250.5.17:51283#015#012Sec-WebSocket-Protocol: sip#015#012Upgrade: websocket#015#012Connection: upgrade#015#012Sec-WebSocket-Accept: 6uLXlD/aBrzu/j3PYP3DWO7rvLE=#015#012Server: kamailio (4.3.5 (x86_64/linux))#015#012Content-Length: 0#015#012#015#012> I am using Kamailio 4.3.5 + CentOS 7.2. I appreciate every help I can get. Best Regards Dimitry Nagorny Trainee

3 4

Dialog module's keep-alive feature uses wrong CSeq?
by Oded Arbel 17 Apr '18

17 Apr '18

Hi all. I'm trying to use the dialog module's keep alive feature (ka_interval() and dlg_set_property()), and the documentation (under dlg_set_property) says this: If keep alive is enabled for a dialog, the module will send SIP OPTIONS requests with CSeq lower or equal than last request within dialog This is seems really weird to me, but the problem is worse than that - when setting dlg_set_property("ka-src"), the OPTIONS messages sent to the caller always have CSeq set to "0", regardless of what CSeq the UA sent on its invite. The problem is that while some UAs respond to these purposefully broken messages with some kind of a 5xx reply (for example, Linphone funnily sends "500 Internal Server Error"), several UAs don't respond at all. Worse than that - some of the UAs that send back 5xx replies to OPTIONS with a non-zero but invalid CSeq will completely ignore the OPTIONS keep-alive message that is sent to the caller with a CSeq 0 - for example, PJSIP responds with "503 Invalid CSeq" to a "ka-dst" OPTIONS message but will ignore a "ka-src" OPTIONS message with CSeq 0. My questions: 1. Why are keep-alive messages sent with a wrong CSeq? 2. If we're keeping the "wrong CSeq" approach, can we at least not send CSeq "0"? 3. Is there a way to change any of the above behaviors with a configuration (i.e. without patching the sources)? Thanks in advance -- Oded Arbel odeda(a)cloudonix.io

3 5

Kamailio with 2 Public and Private IPs- SIP Call flow issues
by arik halperin 16 Apr '18

16 Apr '18

Hello, I have the following setup: Kamailio: Public Interface 70.36.25.65 Private Interface 10.168.10.227 Freeswitch: Private Interface 10.168.10.200 I’m routing calls from kamailio to freeswitch, but when I try to hang up the call the freeswitch sends the BYE to the public address(70.36.25.65) and not to the private address 10.168.10.227. 200OK and ACK work correctly. It’s only the BYE that has an issue. Showing INVITES: Client Invite: T 82.80.164.63:53871 -> 70.36.25.65:5099 [AP] INVITE sip:+972549428460@70.36.25.65:5099 SIP/2.0. Via: SIP/2.0/TCP 10.0.0.42:51619;branch=z9hG4bK-524287-1---52dcc73ec891de14;rport. Max-Forwards: 70. Contact: <sip:972545234585@82.80.164.63:51620;rinstance=1f9d21b54116df89;transport=tcp>. To: <sip:+972549428460@70.36.25.65:5099>. From: "972545234585"<sip:972545234585@70.36.25.65:5099>;tag=902c5325. Kamailio to FS: T 10.168.10.227:46506 -> 10.168.10.200:5080 [AP] INVITE sip:+972549428460@70.36.25.65:5099 SIP/2.0. Record-Route: <sip:70.36.25.65:5099;transport=tcp;lr=on;ftag=902c5325;nat=yes>. Via: SIP/2.0/TCP 70.36.25.65:5099;branch=z9hG4bK1117.1b91cfe0632bd6f036b308f85898e862.1;i=1. Via: SIP/2.0/TCP 10.0.0.42:51619;received=82.80.164.63;branch=z9hG4bK-524287-1---9b6758521ff0f528;rport=53871. Max-Forwards: 69.

3 3

Problem with resuming suspended transacation
by Arik Halperin 16 Apr '18

16 Apr '18

Hello, I’m trying to resume a suspended transaction, but keep getting this error: WARNING: tm [t_suspend.c:193]: t_continue(): transaction is not suspended [20608:1256194941] WARNING: tmx [tmx_mod.c:686]: w_t_continue(): resuming the processing of transaction [20608:1256194941] failed WARNING: tm [t_lookup.c:1483]: t_unref(): script writer didn't release transaction I’m implementing push handling, in INVITE I do the following: route[INVITE] { if (!lookup("location")) { send_reply("100", "Trying"); record_route(); route(SUSPEND); } else { # NAT detection route(NATMANAGE); if(!t_is_set("onreply_route")) t_on_reply("MANAGE_REPLY"); record_route(); t_relay(); ts_store(); $sht(vtp=>stored::$rU) = 1; } route(SENDPUSH); } # suspend the transaction route[SUSPEND] { if(!t_suspend()) //Transaction is suspended! { exit; } $sht(vtp=>join::$rU) = "" + $T(id_index) + ":" +$T(id_label); } In Register I call the PUSHJOIN route: # append branches or resume the transaction route[PUSHJOIN] { $var(hjoin) = 0; lock("$tU"); $var(hjoin) = $sht(vtp=>join::$tU); $var(hstored) = $sht(vtp=>stored::$tU); $sht(vtp=>join::$tU) = $null; unlock("$tU"); if ($var(hjoin)==0) { if ($var(hstored)) ts_append("location", "$tu"); return; } $var(id_index) = $(var(hjoin){s.select,0,:}{s.int}); $var(id_label) = $(var(hjoin){s.select,1,:}{s.int}); ($var(hjoin))\n"); t_continue("$var(id_index)", "$var(id_label)", "INVRESUME"); } For some reason I keep getting: Apr 12 12:37:11 kamprod /usr/local/sbin/kamailio[6039]: ERROR: <script>: 1111 suspended transaction [20608:1256194941] FreeSWITCH => 972587102881 pr 12 12:37:19 kamprod /usr/local/sbin/kamailio[6092]: ERROR: <script>: 1111 tU=972587102881 20608:1256194941 0 Apr 12 12:37:19 kamprod /usr/local/sbin/kamailio[6092]: ERROR: <script>: 1111 resuming trasaction [20608:1256194941] 972587102881 (20608:1256194941) Apr 12 12:37:19 kamprod /usr/local/sbin/kamailio[6092]: WARNING: tm [t_suspend.c:193]: t_continue(): transaction is not suspended [20608:1256194941] Apr 12 12:37:19 kamprod /usr/local/sbin/kamailio[6092]: WARNING: tmx [tmx_mod.c:686]: w_t_continue(): resuming the processing of transaction [20608:1256194941] failed Apr 12 12:37:19 kamprod /usr/local/sbin/kamailio[6092]: WARNING: tm [t_lookup.c:1483]: t_unref(): script writer didn't release transaction

3 2

Audio stops after resuming call from hold
by gerry kernan 16 Apr '18

16 Apr '18

Hi Hoping someone can point me in the right direction. I have a Kamailio Ver: 4.2.3-1.1 running in front of a few asterisk servers Ver: 13.17.2 sip is routed to an asterisk server depending the domain name in the sip request, all working as expected . but if a call is put on hold after resuming the call the party that placed the call on hold can't hear any audio. The other party can hear . do I need to do anything special to handle re-invites for calls put on hold? Gerry Kernan Infinity IT | 17 The Mall | Beacon Court | Sandyford | Dublin D18 E3C8 | Ireland Tel: +353 - (0)1 - 293 0090 | E-Mail: <mailto:gerry.kernan@infinityit.ie> gerry.kernan(a)infinityit.ie Managed IT Services Infinity IT - <http://www.infinityit.ie/> www.infinityit.ie IP Telephony Asterisk Consulting - <http://www.asteriskconsulting.com> www.asteriskconsulting.com Contact Centre Total Interact - <http://www.totalinteract.com> www.totalinteract.com

3 8

Help me!
by Do Quang Trung 15 Apr '18

15 Apr '18

I use kamailio 5.1.1 with openssl-1.0.2n. I built openssl with gost engine supported and the result when i list cipher list as follow: .................... GOST2001-GOST89-GOST89:GOST94-GOST89-GOST89:ECDH-RSA-AES256-GCM-SHA384:ECDH-ECDSA-AES256-GCM-SHA384:ECDH-RSA-AES256-SHA384:ECDH-ECDSA-AES256-SHA384:ECDH-RSA-AES256-SHA:ECDH-ECDSA-AES256-SHA:AES256-GCM-SHA384:AES256-SHA256:AES256-SHA::GOST94-GOST89-GOST89:ECDH-RSA-AES256-GCM-SHA384:ECDH-ECDSA-AES256-GCM-SHA384:ECDH-RSA-AES256-SHA384:ECDH-ECDSA-AES256-SHA384:ECDH-RSA-AES256-SHA:ECDH-ECDSA-AES256-SHA:AES256-GCM-SHA384:AES256-SHA256:AES256-SHA: ................ i must use exactly GOST2001-GOST89-GOST89 in kamailio to protect sip protocol. Please help me to resolve my problem. Best regards, Trung.

3 2

Kamailio 4.2.8 has crashed
by igor.potjevlesch＠gmail.com 13 Apr '18

13 Apr '18

Hello, I had a crash on one Kamailio instance with the following backtrace: Core was generated by `/usr/local/sbin/kamailio -m 704 -M 128 -P /run/kamailio/kamailio.pid'. Program terminated with signal 6, Aborted. #0 0x00007f92366761f7 in raise () from /lib64/libc.so.6 Missing separate debuginfos, use: debuginfo-install bzip2-libs-1.0.6-13.el7.x86_64 elfutils-libelf-0.168-8.el7.x86_64 glibc-2.17-196.el7_4.2.x86_64 keyutils-libs-1.5.8-3.el7.x86_64 krb5-libs-1.15.1-8.el7.x86_64 libacl-2.2.51-12.el7.x86_64 libattr-2.4.46-12.el7.x86_64 libcap-2.22-9.el7.x86_64 libcom_err-1.42.9-10.el7.x86_64 libdb-5.3.21-21.el7_4.x86_64 libgcc-4.8.5-16.el7_4.1.x86_64 libselinux-2.5-11.el7.x86_64 libstdc++-4.8.5-16.el7_4.1.x86_64 lm_sensors-libs-3.4.0-4.20160601gitf9185e5.el7.x86_64 lua-5.1.4-15.el7.x86_64 mariadb-libs-5.5.56-2.el7.x86_64 net-snmp-agent-libs-5.7.2-28.el7_4.1.x86_64 net-snmp-libs-5.7.2-28.el7_4.1.x86_64 nspr-4.13.1-1.0.el7_3.x86_64 nss-3.28.4-15.el7_4.x86_64 nss-softokn-freebl-3.28.3-8.el7_4.x86_64 nss-util-3.28.4-3.el7.x86_64 openssl-libs-1.0.2k-8.el7.x86_64 pcre-8.32-17.el7.x86_64 perl-libs-5.16.3-292.el7.x86_64 popt-1.13-16.el7.x86_64 rpm-libs-4.11.3-25.el7.x86_64 tcp_wrappers-libs-7.6-77.el7.x86_64 xz-libs-5.2.2-1.el7.x86_64 zlib-1.2.7-17.el7.x86_64 (gdb) backtrace #0 0x00007f92366761f7 in raise () from /lib64/libc.so.6 #1 0x00007f92366778e8 in abort () from /lib64/libc.so.6 #2 0x000000000061fb2c in qm_free (qm=0x7f91f4eaf000, p=0x7f92110d0300, file=0x7f9226dd7072 "dialog: dlg_hash.c", func=0x7f9226dd9aeb <__FUNCTION__.12761> "destroy_dlg", line=381) at mem/q_malloc.c:474 #3 0x00007f9226d6fb62 in destroy_dlg (dlg=0x7f920f775598) at dlg_hash.c:381 #4 0x00007f9226d75681 in dlg_unref (dlg=0x7f920f775598, cnt=2) at dlg_hash.c:874 #5 0x00007f9226da14d5 in dlg_ontimeout (tl=0x7f920f7755f8) at dlg_handlers.c:1488 #6 0x00007f9226db1e67 in dlg_timer_routine (ticks=8621904, attr=0x0) at dlg_timer.c:283 #7 0x00000000004af537 in compat_old_handler (ti=137950479, tl=0x7f91f7ad8cc8, data=0x7f91f7ad8cc8) at timer.c:1011 #8 0x00000000004aff14 in slow_timer_main () at timer.c:1145 #9 0x000000000052a2af in main_loop () at main.c:1684 #10 0x000000000052fe71 in main (argc=7, argv=0x7fff51f21418) at main.c:2581 (gdb) Is there enough information to understand the reason of the crash? Regards, Igor.

3 5

Help needed to compile and install kamailio 5.1.2
by vinay kumar 13 Apr '18

13 Apr '18

Hi All, can anyone share me the installtion steps for kamailio 5.1.2? can i compile and run kamailio 5.1.2 on ubuntu 14.04? i have already used 4.3.6 with polaris configuration to bringup kamailio and i wanna upgrade to kamailio 5.1.2 and the same way can i use 5.1.2 with polaris configuration? I have used polaris scripts to bringup kamailio 5.1.2 and its failing with status kamailio service with Result resources. if i use latest debian packages , are polaris configurations are compatible with latest packages?? Please help me out. Thanks in advance. Regards, vinay

1 0

Is RTPProxy the way to go?
by Robert Johnson 12 Apr '18

12 Apr '18

I'm encountering a situation where an endpoint will renegotiate it's RTP ports but RTPProxy does not honor the new ports and continues to send audio to the old ports. I realize that I'm running a (very) old version of RTPProxy and that I should upgrade before submitting any bugs or asking for help with the program. My question is, is RTPProxy still the way to go? And if so, is v2.0.0 the best version to go with? https://github.com/sippy/rtpproxy/releases At a basic level, the network looks like this +---------------+ +--------------+ | | | | | Priv. Network | | Pub. Network | | | | | +------+ +---------+--+ +---+-------+--+ +--+--------+ | | | | | | | | | PSTN <---> Softswitch <--------> Kamailio + <--------> EndPoints | | | | | | RTPProxy | | | +------+ +---------+--+ | | +--+--------+ | +---+-------+--+ | | | | | +---------------+ +--------------+

2 1

Kamailio + tls
by Do Quang Trung 12 Apr '18

12 Apr '18

Plz help me. I want config kamailio 5.1 support tls using gost89 cipher to protect data.

2 3

Re: [SR-Users] MSILO: SIP stored offline MESSAGE repetitive delivery
by Abdul Basit 12 Apr '18

12 Apr '18

Hi Team, My issue for IM handling has been resolved. @MS helped to look into the matter. From to the example https://www.kamailio.org/docs/modules/5.1.x/modules/msilo.html#idp45433036 # -- tm params -- modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) I replaced it as below # -- tm params -- modparam("tm", "fr_timer", 10000 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) Lower fr_timer was initiating 408 without waiting for 200 OK from destination because 10ms is too low. This was confusing msilo module and the sender device that MESSAGE sent was failed. Increasing the fr_timer value resolved the issue. This also resolved a ripple effect. i.e, kamailio delivery of offline messages from DB store to the destination party. Since kamailio was get message delivery error, it was executing failure_route[1] that was storing the message again in DB store as offline message and so on. lower fr_timer values are also exits in 3.x, 4.x, 5x and dev branches documentation. Correction is required. -- regards, abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 > > On 9 April 2018 at 19:34, Abdul Basit <basit.engg(a)gmail.com> wrote: > >> Dear Team, >> >> I have added traces via pasetbin >> >> MESSAGE Stored When B is OffLine https://pastebin.com/NLwMePmf >> >> [trace screenshot removed] >> >> Next this messages should be delivered when B party REGISTER with the >> server >> >> *Issue 1*: Message was delivered to B-party but it didn't remove form DB >> and as a result MESSAGE deliver to B-party as many times it come line >> https://pastebin.com/7pb0wCKx >> [trace screenshot removed] >> >> Now next issue is >> *Issue 2*: Messages deliver to B-party when its already online without >> storing in DB. B-party receive MESSAGE successfully but A-party gets >> MESSAGE delivery failed. >> https://pastebin.com/zgDbK4F6 >> >> [trace screenshot removed] >> >> I intentionally modified the 202 - description for better troubleshooting. >> >> See if attachments were received properly through mailing list. >> >> For convenience I have attached the pcap as well. >> >> MESSAGE: 1: Accepted and Stored as expected >> MESSAGE: 2: Server delivered Message when B comes online >> MESSAGE: 3: Both A & B are online. A-party receives MESSAGE delivery >> failure where as B-party receives MESSAGE successfully >> >> >> Modified default kamailio.cfg is https://pastebin.com/thZig5RC >> It looks that mingled up the configuration. >> >> Please see whats gone wrong here. >> >> >> -- >> regards, >> >> abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 >> >> On 9 April 2018 at 18:20, Abdul Basit <basit.engg(a)gmail.com> wrote: >> >>> Dear team, >>> >>> Thank you for your reply. >>> >>> @MS & @ Davld, Let me test again and share traces. >>> >>> @Zaka, I will check the save() result as you suggested. Location is >>> being saved and properly looked up because messages are being delivered >>> (and not stored in DB ) when both parties are online. >>> >>> -- >>> regards, >>> >>> abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 >>> >>> On 9 April 2018 at 16:38, Muhammad Zaka <muhammad.zaka(a)cloudcall.com> >>> wrote: >>> >>>> Please check the return value of save and replace the logic: - >>>> >>>> >>>> >>>> $var(save_result) = save("location") ; >>>> >>>> if ($var(save_result) <= -1) { >>>> >>>> xlogl("L_ERROR", "[$rm] -> Unable to save user location \n"); >>>> sl_reply_error(); >>>> >>>> exit(); >>>> >>>> } >>>> >>>> >>>> >>>> >>>> >>>> if ($var(save_result) == 1) { >>>> >>>> xlogl("L_INFO", "[$rm] -> dumping messages with MSILO \n"); >>>> # MSILO - dumping user's offline messages to user >>>> if (m_dump("$fu")) { >>>> xlogl("L_NOTICE", "MSILO: offline messages for $fu dumped - if >>>> they were \n"); >>>> } >>>> else { >>>> xlogl("L_NOTICE", "MSILO: no offline messages dumped \n"); >>>> }; >>>> } >>>> >>>> >>>> >>>> This means dump the message only when it is not re-registration. >>>> >>>> >>>> >>>> Thanks >>>> >>>> >>>> >>>> Regards >>>> >>>> Muhammad Zaka >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> *From:* sr-users [mailto:sr-users-bounces@lists.kamailio.org] *On >>>> Behalf Of *Abdul Basit >>>> *Sent:* 08 April 2018 18:43 >>>> *To:* Kamailio (SER) - Users Mailing List <sr-users(a)lists.kamailio.org>; >>>> Kamailio (SER) - Users Mailing List <sr-users(a)lists.sip-router.org> >>>> *Subject:* Re: [SR-Users] MSILO: SIP stored offline MESSAGE repetitive >>>> delivery >>>> >>>> >>>> >>>> Dear Team, >>>> >>>> Still stuck with the issue. Kindly suggest where i did wrong. >>>> >>>> >>>> >>>> >>>> -- >>>> regards, >>>> >>>> >>>> abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445 >>>> >>>> >>>> >>>> On 6 April 2018 at 20:50, Abdul Basit <basit.engg(a)gmail.com> wrote: >>>> >>>> Hi team, >>>> >>>> I am facing an issue while setting up IM server based on kamailio. >>>> >>>> SIP-SIP online messaging is fine. For offline messaging i am using >>>> msilo module which is storing offline messages in 'silo' table. >>>> >>>> >>>> >>>> As B-party come online ... it receive offline messages stored in >>>> database. But >>>> >>>> *Problem is*.. these messages will deliver to B-party as many time it >>>> REGISTER online. >>>> >>>> >>>> >>>> How can I configure msilo module to mark messages delivery status? >>>> or how can i delete the delivered messages so that there will be no >>>> chance of delivering them again on next REGISTER? >>>> >>>> Message parts of kamailio.cfg are >>>> .... >>>> /* add local domain aliases */ >>>> alias="192.168.10.18" >>>> port=9060 >>>> >>>> >>>> # ----- msilo params ----- >>>> modparam("msilo", "db_url", DBURL) >>>> modparam("msilo", "db_table", "silo") >>>> modparam("msilo", "sc_mid", "id") >>>> modparam("msilo","from_address","sip:registrar@192.168.10.18:9060") >>>> modparam("msilo","contact_hdr","Contact: registrar@192.168.10.18:9060;m >>>> silo=yes\r\n") >>>> modparam("msilo","content_type_hdr","Content-Type: text/plain\r\n") >>>> >>>> ........ >>>> >>>> route[MSILO_MESSAGE] { >>>> if (!is_method("MESSAGE")) return; >>>> xlogl("L_NOTICE", "[$rm] r-uri: $ru d-uri: $du \n"); >>>> if (lookup("location")) { >>>> xlogl("L_NOTICE", "[$rm] -> loaction found. Relying message >>>> [[$rb]] to the UAC \n"); >>>> t_relay(); >>>> } else { >>>> if (!t_newtran()) { >>>> xlogl("L_ERROR", "[$rm] -> ERROR creating new transaction >>>> \n"); >>>> sl_reply_error(); >>>> exit; >>>> }; >>>> xlogl("L_NOTICE", "[$rm] -> Storing MSILO_MESSAGE \n >>>> [$ru]-[$tu]-[$ou] \n "); >>>> if (m_store("$ru")) { >>>> xlogl("L_NOTICE", "MSILO: offline message stored! \n"); >>>> if (!t_reply("202", "Accpeted and Strored")) { >>>> xlogl("L_ERROR", "MSILO: Err sending 202 reply... \n"); >>>> sl_reply_error(); >>>> } >>>> } else { >>>> xlogl("L_ERROR", "MSILO: offline message NOT stored... \n"); >>>> if (!t_reply("503", "Service Unavailable")) { >>>> sl_reply_error(); >>>> } >>>> } >>>> } >>>> exit; >>>> } >>>> >>>> ..... >>>> >>>> # Handle SIP registrations >>>> route[REGISTRAR] { >>>> if (!is_method("REGISTER")) return; // discard all non-REGISTER >>>> methods >>>> >>>> if(isflagset(FLT_NATS)) { >>>> setbflag(FLB_NATB); >>>> #!ifdef WITH_NATSIPPING >>>> setbflag(FLB_NATSIPPING); //# do SIP NAT pinging >>>> #!endif >>>> } >>>> $avp(received) = "sip:" + $si + ":" + $sp + ";transport=" + $proto; >>>> xlogl("L_INFO", "[$rm] -> UAC actual location $avp(received) \n"); >>>> if (!save("location",0x04)) { >>>> xlogl("L_ERROR", "[$rm] -> Unable to save user location \n"); >>>> sl_reply_error(); >>>> } >>>> else { >>>> xlogl("L_INFO", "[$rm] -> dumping messages with MSILO \n"); >>>> # MSILO - dumping user's offline messages to user >>>> if (m_dump("$fu")) { >>>> xlogl("L_NOTICE", "MSILO: offline messages for $fu dumped - >>>> if they were \n"); >>>> } >>>> else { >>>> xlogl("L_NOTICE", "MSILO: no offline messages dumped \n"); >>>> }; >>>> } >>>> exit; >>>> } >>>> >>>> Any idea? >>>> >>>> >>>> -- >>>> regards, >>>> >>>> >>>> abdul basit >>>> >>>> >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Users Mailing List >>>> sr-users(a)lists.kamailio.org >>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> >>> >> >

2 1

Limiting Kamailios accepted methods
by Gholamreza Sabery 12 Apr '18

12 Apr '18

Hi, I want to limit Kamailios accepted methods to REGISTER, INVITE and their related methods like BYE and CANCEL. How to do this in Kamailio? Is there a standard way or I should use is_method function? Regards

3 4

Re: [SR-Users] Is 't_newtran()' needed when using reliable TCP/TLS transport?
by Cody Herzog 11 Apr '18

11 Apr '18

Another thought... Perhaps 'async_task_route()' automatically creates a transaction, which is why I was having problems with the calls to 't_newtran()' and 't_release()' after the route had been changed to execute through 'async_task_route()'. If so, then maybe it's safe for my async presence route to not call 't_newtran()' and 't_release()'. Thanks.

1 0

Re: [SR-Users] Is 't_newtran()' needed when using reliable TCP/TLS transport?
by Cody Herzog 11 Apr '18

11 Apr '18

Thanks for the quick reply, Alex. I will try to provide more context information. The route in question is related to presence. Our usage of t_newtran() is very similar to the following presence examples: https://github.com/kamailio/kamailio/blob/master/test/unit/presence.cfg https://lists.kamailio.org/pipermail/sr-users/2016-April/092544.html https://www.kamailio.org/dokuwiki/doku.php/presence:configuration-file (see route[2]) I believe our config was probably based off one of those. In that context, with reliable transport, is 't_newtran()' useful? Thanks.

1 0

Is 't_newtran()' needed when using reliable TCP/TLS transport?
by Cody Herzog 11 Apr '18

11 Apr '18

Hello. I have read that the primary use of 't_newtran()' is to handle SIP retransmissions. Given that, is there any reason to use it when operating over TCP/TLS? I recently changed a route to operate asynchronously through 'async_task_route()'. When I did, calls to 't_newtran()' started failing. I'm hoping that I can safely remove all the calls to 't_newtran()' and the matching calls to 't_release()' on this async route, because I am using TLS as the transport. Does that sound correct? Thanks.

2 1

Kamailio logs in RHEL 7 environment
by igor.potjevlesch＠gmail.com 11 Apr '18

11 Apr '18

Hello ! Since I moved my OS to RHEL 7, the logs management has changed. I cannot have all logs in /var/log/messages as I used to have. Indeed, there is journald/rsyslog which blocked logs to be written after hundreds/thousands of messages. I tried to set these values to 0: vim /etc/systemd/journald.conf [Journal] [.] RateLimitInterval=0 RateLimitBurst=0 But the problem is still present. Anyone has already solved this problem? Regards, Igor.

2 2

Default AUTH route potential issue?
by Володимир Іванець 11 Apr '18

11 Apr '18

Hello all! I'm using Kamailio 5.1.0 on my testing machine. Configuration includes slightly modified AUTH route from http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb route[AUTH] { xlog("L_DBG", "== TRACE. AUTH\n"); # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && uri!=myself) { xlog("L_DBG", "== TRACE. AUTH. Not relaying. Exiting.\n"); sl_send_reply("403","Not relaying"); exit; } if(isflagset(TRUSTEDIP)) { xlog("== TRACE. AUTH. TRUSTEDIP. Returning.\n"); return; } if (is_method("REGISTER") || from_uri==myself) { xlog("L_DBG", "== TRACE. AUTH. Method REGISTER\n"); # authenticate requests if (!auth_check("$fd", "sipusers", "1")) { auth_challenge("$fd", "0"); xlog("L_DBG", "== TRACE. AUTH. Exiting.\n"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) { xlog("L_DBG", "== TRACE. AUTH. Method is not REGISTER|PUBLISH\n"); consume_credentials(); } } xlog("L_DBG", "== TRACE. AUTH. Returning.\n"); return; } I opened port UDP/5060 to everyone today and started receiving some SIP requests. Most INVITEs were stopped by *auth_challenge* but then I received this one: 2018/04/11 16:32:44.385689 38.91.106.211:5069 -> 172.16.30.205:5060 INVITE sip:100@MY_PUB_IP_ADDRESS SIP/2.0 v: SIP/2.0/UDP 38.91.106.211:5060;branch=z9hG4bK-929181129;rport Content-Length: 0 f: "pbx"<sip:100@1.1.1.1>;tag=3535306165633930313363340131373533363938373235 i: 757925348661465531074812 m: sip:100@38.91.106.211:5069 Accept: application/sdp CSeq: 1 INVITE t: "pbx"<sip:100@1.1.1.1> Max-Forwards: 70 ... and it came through AUTH route. Below are two fragments of Kamailio log: Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <script>: == TRACE. INVITE From: sip:100@1.1.1.1 (IP:38.91.106.211:5069) Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <script>: == TRACE. To: sip:100@1.1.1.1 Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: pv [pv_core.c:1286]: pv_get_dsturi(): no destination URI Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <script>: == TRACE. Destination URI : <null> Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <script>: == TRACE. SIP Request header : sip:100@MY_PUB_IP_ADDRESS Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/parser/msg_parser.c:89]: get_hdr_field(): found end of header Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: pv [pv_core.c:966]: pv_get_useragent(): no User-Agent header Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <script>: == TRACE. User Agent header : <null> **************************************************************************************************** Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <script>: == TRACE. request_route ==> AUTH Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <script>: == TRACE. AUTH Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:564]: grep_sock_info(): checking if host==us: 7==9 && [1.1.1.1] == [127.0.0.1] Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:567]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:564]: grep_sock_info(): checking if host==us: 7==13 && [1.1.1.1] == [172.16.30.205] Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:567]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:564]: grep_sock_info(): checking if host==us: 7==9 && [1.1.1.1] == [127.0.0.1] Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:567]: grep_sock_info(): checking if port 8088 (advertise 0) matches port 5060 Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/forward.c:412]: check_self(): host != me Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:564]: grep_sock_info(): checking if host==us: 7==9 && [1.1.1.1] == [127.0.0.1] Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:567]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:564]: grep_sock_info(): checking if host==us: 7==13 && [1.1.1.1] == [172.16.30.205] Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:567]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:564]: grep_sock_info(): checking if host==us: 7==9 && [1.1.1.1] == [127.0.0.1] Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:567]: grep_sock_info(): checking if port 8088 (advertise 0) matches port 5060 Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/forward.c:412]: check_self(): host != me Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:564]: grep_sock_info(): checking if host==us: 13==9 && [ MY_PUB_IP_ADDRESS ] == [127.0.0.1] Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:567]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:564]: grep_sock_info(): checking if host==us: 13==13 && [ MY_PUB_IP_ADDRESS ] == [172.16.30.205] Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:567]: grep_sock_info(): checking if port 5060 (advertise 0) matches port 5060 Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:564]: grep_sock_info(): checking if host==us: 13==9 && [ MY_PUB_IP_ADDRESS ] == [127.0.0.1] Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <core> [core/socket_info.c:567]: grep_sock_info(): checking if port 8088 (advertise 0) matches port 5060 Apr 11 16:32:44 kamailio-dev /usr/sbin/kamailio[31373]: DEBUG: <script>: == TRACE. AUTH. Returning. As you can see all tests failed to catch this INVITE request and Kamailio continued processing it. And I'm now wondering what would be the best way to identify such packet. Thanks.

2 1

[SOLVED]: rtpengine no voice when working between DTLS-SRTP endpoints
by Aqs Younas 11 Apr '18

11 Apr '18

DTLS=passive solved it. Thanks guys. On Tue, 10 Apr 2018, 11:37 pm Aqs Younas, <aqsyounas(a)gmail.com> wrote: > Sometimes, I see below logs in RTP engine. > > > Apr 10 17:59:21 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: Received command 'answer' from > 127.0.0.1:44933 > Apr 10 17:59:21 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: answer time = 0.000163 sec > Apr 10 17:59:21 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: Replying to 'answer' from > 127.0.0.1:44933 > [1523383161.279950] ERR: [24b95195-3da3-4e12-8400-5fcf908183e5 port > 8268]: SRTP output wanted, but no crypto suite was negotiated > Apr 10 17:59:21 centos-1024mb-nyc-02 rtpengine[65101]: ERR: > [24b95195-3da3-4e12-8400-5fcf908183e5 port 8268]: SRTP output wanted, but > no crypto suite was negotiated > Apr 10 17:59:25 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5 port 8268]: Confirmed peer address > as 72.214.35.171:64834 > [1523383171.481023] ERR: [24b95195-3da3-4e12-8400-5fcf908183e5 port > 8269]: SRTCP output wanted, but no crypto suite was negotiated > Apr 10 17:59:31 centos-1024mb-nyc-02 rtpengine[65101]: ERR: > [24b95195-3da3-4e12-8400-5fcf908183e5 port 8269]: SRTCP output wanted, but > no crypto suite was negotiated > Apr 10 17:59:31 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5 port 8269]: Confirmed peer address > as 72.214.35.171:50108 > [1523383176.025296] ERR: [24b95195-3da3-4e12-8400-5fcf908183e5 port > 8268]: SRTP output wanted, but no crypto suite was negotiated > Apr 10 17:59:36 centos-1024mb-nyc-02 rtpengine[65101]: ERR: > [24b95195-3da3-4e12-8400-5fcf908183e5 port 8268]: SRTP output wanted, but > no crypto suite was negotiated > [1523383186.000280] ERR: [02d8ccfb-831e-4a81-bd2b-09b007d39209]: DTLS > error: 1 (read timeout expired) > [1523383186.000335] ERR: [02d8ccfb-831e-4a81-bd2b-09b007d39209]: DTLS > error on local port 8248 > [1523383186.000419] ERR: [02d8ccfb-831e-4a81-bd2b-09b007d39209]: DTLS > error: 1 (read timeout expired) > [1523383186.000429] ERR: [02d8ccfb-831e-4a81-bd2b-09b007d39209]: DTLS > error on local port 8249 > *Apr 10 17:59:46 centos-1024mb-nyc-02 rtpengine[65101]: ERR: > [02d8ccfb-831e-4a81-bd2b-09b007d39209]: DTLS error: 1 (read timeout > expired)* > *Apr 10 17:59:46 centos-1024mb-nyc-02 rtpengine[65101]: ERR: > [02d8ccfb-831e-4a81-bd2b-09b007d39209]: DTLS error on local port 8248* > *Apr 10 17:59:46 centos-1024mb-nyc-02 rtpengine[65101]: ERR: > [02d8ccfb-831e-4a81-bd2b-09b007d39209]: DTLS error: 1 (read timeout > expired)* > *Apr 10 17:59:46 centos-1024mb-nyc-02 rtpengine[65101]: ERR: > [02d8ccfb-831e-4a81-bd2b-09b007d39209]: DTLS error on local port 8249* > *Apr 10 18:00:41 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: Closing call due to timeout* > Apr 10 18:00:41 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: Final packet stats: > Apr 10 18:00:41 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: --- Tag > 'f6d2237d-f960-4542-b138-f39a7fb52770', created 1:32 ago for branch '', in > dialogue with '6bdf30d1-2da6-4b6d-b917-aaa720c9c1fa' > Apr 10 18:00:41 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: ------ Media #1 (audio over > UDP/TLS/RTP/SAVP) using unknown codec > *Apr 10 18:00:41 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: --------- Port 209.182.216.71:8288 > <http://209.182.216.71:8288> <> 100.84.103.245:4002 > <http://100.84.103.245:4002> , SSRC 0, 0 p, 0 b, 0 e, 92 ts* > *Apr 10 18:00:41 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: --------- Port 209.182.216.71:8289 > <http://209.182.216.71:8289> <> 100.84.103.245:4003 > <http://100.84.103.245:4003> (RTCP), SSRC 0, 0 p, 0 b, 0 e, 92 ts* > Apr 10 18:00:41 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: --- Tag > '6bdf30d1-2da6-4b6d-b917-aaa720c9c1fa', created 1:32 ago for branch '', in > dialogue with 'f6d2237d-f960-4542-b138-f39a7fb52770' > Apr 10 18:00:41 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: ------ Media #1 (audio over > UDP/TLS/RTP/SAVP) using G722/8000 > Apr 10 18:00:41 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: --------- Port > 209.182.216.71:8268 <> 72.214.35.171:64834, SSRC 653128b4, 935 p, > 160820 b, 0 e, 60 ts > Apr 10 18:00:41 centos-1024mb-nyc-02 rtpengine[65101]: INFO: > [24b95195-3da3-4e12-8400-5fcf908183e5]: --------- Port > 209.182.216.71:8269 <> 72.214.35.171:50108 (RTCP), SSRC 653128b4, 3 p, > 278 b, 0 e, 60 ts > > > Any suggestion what might be happening? > > Br, Aqs. > > On 10 April 2018 at 22:59, Aqs Younas <aqsyounas(a)gmail.com> wrote: > >> I could see SRTP packets coming from one device but they never leave >> rtpeninge. >> >> I put a link to Pastebin containing a call trace with the hope that >> someone might help me out. >> >> I could provide more info if required. >> >> https://pastebin.com/tYVpFQAh >> >> Br, Aqs. >> >> >> On 10 April 2018 at 01:39, Aqs Younas <aqsyounas(a)gmail.com> wrote: >> >>> Greetings list, >>> >>> I am trying to make two endpoints talking on DTLS-SRTP. But I hear on >>> audio. >>> >>> Things work perfectly fine if I use RTP or SRTP with TLS. >>> >>> Endpoints are pjsip based application not webrtc based clients. >>> >>> >>> Below are logs from rtpengine. I hope someone could point out amiss. >>> >>> Apr 9 20:02:43 centos-1024mb-nyc-02 rtpengine[58438]: INFO: >>> [66d2da58-21fe-48bd-9999-a1f3a22afa6d]: --------- Port >>> 209.182.216.71:8176 <> 72.214.35.171:63577, SSRC 1234c6eb, 641 p, >>> 110252 b, 0 e, 60 ts >>> Apr 9 20:02:43 centos-1024mb-nyc-02 rtpengine[58438]: INFO: >>> [66d2da58-21fe-48bd-9999-a1f3a22afa6d]: --------- Port >>> 209.182.216.71:8177 <> 72.214.35.171:63056 (RTCP), SSRC 1234c6eb, 4 >>> p, 372 b, 0 e, 60 ts >>> Apr 9 20:12:24 centos-1024mb-nyc-02 rtpengine[58438]: INFO: Version >>> git-master-3ef300b shutting down >>> Apr 9 20:12:37 centos-1024mb-nyc-02 rtpengine[58958]: INFO: Generating >>> new DTLS certificate >>> Apr 9 20:12:37 centos-1024mb-nyc-02 rtpengine[58959]: INFO: Startup >>> complete, version git-master-3ef300b >>> Apr 9 20:13:43 centos-1024mb-nyc-02 rtpengine[58959]: INFO: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08]: Received command 'offer' from >>> 127.0.0.1:57645 >>> Apr 9 20:13:43 centos-1024mb-nyc-02 rtpengine[58959]: NOTICE: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08]: Creating new call >>> Apr 9 20:13:43 centos-1024mb-nyc-02 rtpengine[58959]: INFO: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08]: offer time = 0.002612 sec >>> Apr 9 20:13:43 centos-1024mb-nyc-02 rtpengine[58959]: INFO: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08]: Replying to 'offer' from >>> 127.0.0.1:57645 >>> Apr 9 20:13:56 centos-1024mb-nyc-02 rtpengine[58959]: INFO: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08]: Received command 'answer' from >>> 127.0.0.1:42309 >>> Apr 9 20:13:56 centos-1024mb-nyc-02 rtpengine[58959]: INFO: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08]: answer time = 0.000220 sec >>> Apr 9 20:13:56 centos-1024mb-nyc-02 rtpengine[58959]: INFO: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08]: Replying to 'answer' from >>> 127.0.0.1:42309 >>> Apr 9 20:13:56 centos-1024mb-nyc-02 rtpengine[58959]: INFO: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08 port 8000]: DTLS: Peer certificate >>> accepted >>> Apr 9 20:13:56 centos-1024mb-nyc-02 rtpengine[58959]: INFO: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08 port 8000]: DTLS-SRTP successfully >>> negotiated >>> Apr 9 20:13:57 centos-1024mb-nyc-02 rtpengine[58959]: ERR: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08 port 8000]: SRTP output wanted, but >>> no crypto suite was negotiated >>> Apr 9 20:14:00 centos-1024mb-nyc-02 rtpengine[58959]: INFO: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08 port 8000]: Confirmed peer address >>> as 72.214.35.171:58634 >>> Apr 9 20:14:07 centos-1024mb-nyc-02 rtpengine[58959]: ERR: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08 port 8001]: SRTCP output wanted, but >>> no crypto suite was negotiated >>> Apr 9 20:14:07 centos-1024mb-nyc-02 rtpengine[58959]: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08] version=2, padding=0, count=1, >>> payloadtype=200, length=12, ssrc=2045607967, ntp_sec=1379282714, >>> ntp_fractions=439054259, rtp_ts=1838072137, sender_packets=3366262813, >>> sender_bytes=2383498210, ssrc=815258372, fraction_lost=96, >>> packet_loss=13713522, last_seq=3314313929, jitter=2878956247, >>> last_sr=2456273253, delay_since_last_sr=3351655681 >>> Apr 9 20:14:07 centos-1024mb-nyc-02 rtpengine[58959]: INFO: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08 port 8001]: Confirmed peer address >>> as 72.214.35.171:57732 >>> Apr 9 20:14:12 centos-1024mb-nyc-02 rtpengine[58959]: ERR: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08 port 8000]: SRTP output wanted, but >>> no crypto suite was negotiated >>> Apr 9 20:14:17 centos-1024mb-nyc-02 rtpengine[58959]: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08] version=2, padding=0, count=1, >>> payloadtype=200, length=12, ssrc=2045607967, ntp_sec=2811881700, >>> ntp_fractions=4080266212, rtp_ts=371429680, sender_packets=958830616, >>> sender_bytes=2579186043, ssrc=1909756377, fraction_lost=174, >>> packet_loss=11416637, last_seq=3106722675, jitter=758758394, >>> last_sr=2663618457, delay_since_last_sr=1399181077 >>> Apr 9 20:14:27 centos-1024mb-nyc-02 rtpengine[58959]: ERR: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08 port 8000]: SRTP output wanted, but >>> no crypto suite was negotiated >>> Apr 9 20:14:27 centos-1024mb-nyc-02 rtpengine[58959]: ERR: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08 port 8001]: SRTCP output wanted, but >>> no crypto suite was negotiated >>> Apr 9 20:14:27 centos-1024mb-nyc-02 rtpengine[58959]: >>> [a5ca8335-84d2-4daf-a0e2-6465e72b6d08] version=2, padding=0, cou >>> >>> >>> It is how I have programmed it in my Kamailio configuration. >>> >>> ON INVITE >>> >>> rtpengine_offer("replace-origin replace-session-connection ICE=remove >>> UDP/TLS/RTP/SAVP"); >>> >>> ON 200-ok >>> >>> rtpengine_answer("replace-origin replace-session-connection ICE=remove >>> UDP/TLS/RTP/SAVP"); >>> >>> >>> Best Regards, >>> >>> Aqs Younas >>> >> >> >

1 0

Is it possible to alter SDP content before forwarding the packet?
by Wilkins, Steve 11 Apr '18

11 Apr '18

Hello all, I was curious whether or not it possible to alter SDP content before forwarding the packet? Thank you all as always!

2 1

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