- sr-users - kamailio.org

Request Time Out
by Vasim Antule 24 Nov '17

24 Nov '17

Hello, I am new to SIP/Kamailio , I am attempting to use Kamailio 5.0 in test environment. I am getting Request Time Out for Android Phones. Apple to Apple phones are working fine. But Android to Apple and Android to Android I am getting Request Time Out. Please Guide. Regards, Vasim Antule

1 0

Issue dumping an huge htable through xmlrpc
by Marco Capetta 24 Nov '17

24 Nov '17

Hello everyone, On Kamailio version 4.4.6 I have an issue dumping an huge htable through xmlrpc. We are using htables to store the subscriber registration attempts to prevent bruteforcing password attacks. Usually there aren't any issues to dump information using xmlrpc. On the contrary, if the number of records in the htable become huge (in the order of 10.000), xmlrpc output is truncated, therefore it is not possible to parse it. I have the same results either using a Perl library or the "xmlrpc_test.py" script included in package. I tried to increase the module parameters "binrpc_max_body_size" and "binrpc_struct_max_body_size" without success. Do you have any other suggestion? Thank you Regards Marco -- *Marco Capetta * Operations Engineer Sipwise GmbH <http://www.sipwise.com> , Campus 21/Europaring F15 AT-2345 Brunn am Gebirge Phone: +43(0)1 301 2044 <tel:+4313012044> Email: mcapetta(a)sipwise.com <mailto:mcapetta@sipwise.com> Website: www.sipwise.com <http://www.sipwise.com> Particulars according Austrian Companies Code paragraph 14 "Sipwise GmbH" - Europaring F15 - 2345 Brunn am Gebirge FN:305595f, Commercial Court Vienna, ATU64002206

2 1

Failed to find command http_client_query
by Alexandru Covalschi 24 Nov '17

24 Nov '17

Hello everyone, Updated to 5.0.4 from repo, up-to-date debian 8, http_client.so is certainly loaded. When I try to change http_query to http_client_query it fails to load with error "failed to find command http_client_query". Any ideas? It is same on all my test servers. ________________________________ Alexandru Covalschi VoIP Engineer and System Administrator tel: +373 673 98 493

2 12

msilo support for picture, audio and video?
by sscc 24 Nov '17

24 Nov '17

Hello Folks does msilo support picture, audio and video? regards sscc? -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 5

Loadbalancing authenticated agetways
by SamyGo 23 Nov '17

23 Nov '17

Hi, I've a scenario where multiple mediaservers will be registered to Kamailio and for any incoming call from Upstream provider I've to perform load-balancing to the actively registered media-servers. dispatcher module does load-balancing for IP endpoints, is there any other module where I can perform load-balancing on registered clients(media-servers)? Thanks, Sammy

3 3

why do I need rtpproxy
by Wilkins, Steve 23 Nov '17

23 Nov '17

The way I understand it, rtpproxy is needed to pass video/audio through Kamailio to Asterisk. However, even without rtpproxy running, I get video through to Asterisk via Kamailio, and the video plays with no issues on a sip device (e.g. MicroSIP). I am confused what I need rtpproxy for. Thank you

3 2

rfc: clang format the code
by Daniel-Constantin Mierla 23 Nov '17

23 Nov '17

Hello, it was discussed during the last irc devel meeting and everyone there agreed to use clang-format to format the source code: - https://clang.llvm.org/docs/ClangFormat.html Is any developer (maybe not present during the irc devel meeting) opposing this? It is not going to be all at once, because some of the structures and initialization of them may be preferred to be kept with a custom format for being easier to read -- to have a section in the file not formatted, it has to be enclosed in between comments: /* clang-format off */ ... /* clang-format o. */ But for the rest of the code (i.e., functions) I think it will be better to agree on a format and try to use it. The clang-formatting should be done when one needs to update code in some files. Over the years, developers come and go, having each one with a custom formatting style is hard to maintain properly in long term. There are modules that do not indent coherently code inside blocks, using a mixture, more or less no style. This makes it hard for other developers to review in case of reported issues, but also for new comers to get familiar with the code. In the source tree there is already a .clang-format file which defines the options for formatting. It was used by me and other developers already -- I did it mainly in my modules or those components that do not have an active maintainer. But I plan to do it also on the components that have currently a maintainer at the moment I need to work on that code (e.g., fix some issue or export functions to kemi framework). My process of clang-formatting a module is like: - edit the files to add clang-format off/on comments around the parts I want to preserve as they are (typically mod exports -- params, functions, kemi -- and intialization of structures that are aligned to a specific indentation that makes it easier to read). - run: clang-format -i src/modules/modname/*.{c,h} - commit with these changes, no new feature or bug fix must be done in the same commit (can be combined with removing the history comments) If one wants different options inside .clang-format, please propose and let's discuss which one is preferred by the majority of the current developers. If no one shows anything against in a matter of few days, it will be considered accepted (after all, one can re-indent on own style if the rule is not accepted or overturned). 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

1 0

Error when attempting to use rtpproxy
by Wilkins, Steve 23 Nov '17

23 Nov '17

Hello, I entered this question before but did not include the tcpdump file. I am re-posting just in case this message was over looked for lack of information. I am using Kamailio version 5.0.3 (x86_64/linux), and it is installed and working on a Centos 7 Sever. I use Kamailio as a SIP Proxy Server for Asterisk and things seem to work fine, however, I do have an issue and a question for which I am trying to understand and am hoping someone can help me with. My server name is vadev2 (Kamailio run on this server), and it's IP Address and port are 172.21.1.108:5060; Asterisk servers are (172.21.1.97:5060, and 172.21.1.124:5060). The rtpproxy is 172.21.1.108:17722 Issue: Every 10 seconds I get an Error message in the Kamailio log which says => Nov 13 18:10:55 vadev2/usr/local/sbin/kamailio[29188]: ERROR: <core> [core/tcp_main.c:2660]: tcpconn_1st_send(): connect 172.21.1.108:59794<http://172.1.1.13:59794> failed (RST) Connection refused. I do not know what triggers this error, and as mentioned, things do seem to work, other than I do not believe anything is going through rtpproxy; I can make calls and use sip tools which pass requests through Kamailio to Asterisk. Does anyone know what could be causing this Error? I do not understand what is trying to connect to what and why. Should I be using rtpengine instead of rtpproxy? I did attempt rtpengine but it would not compile due to missing json.h files. Thank you for any help -Steve

1 0

MSILO with encryption/decryption
by SamyGo 23 Nov '17

23 Nov '17

Hi, Is there any internal mechanism for msilo to store messages encrypted and retrieve them decrypted before sending? Thanks, Sammy

1 0

RV: CANCEL not being matched by tm module when via address does not have port included
by CARLOS GARCIA MORCHON 23 Nov '17

23 Nov '17

Hi, I have an IMS core deployed using kamailio 5.0.2. Calls from clients are reaching the core pcscf from a freeswitch (with address A.A.A.A) + a kamailio acting as proxy (with address B.B.B.B), so the INVITE request looks like this: INVITE sip:1111111111@mnc001.mcc001.3gppnetwork.org SIP/2.0 Record-Route: <sip:A.A.A.A;lr> Via: SIP/2.0/UDP A.A.A.A;branch=z9hG4bK8735.ef7a61e110f335b2e92a8c1d430de585.0 Via: SIP/2.0/UDP B.B.B.B:5080;received=B.B.B.B;rport=5080;branch=z9hG4bKHpB704ZaXSKtm Max-Forwards: 65 From: "222222222" <sip:222222222@B.B.B.B>;tag=eytD49ymQr71S To: <sip:1111111111@mnc001.mcc001.3gppnetwork.org> Call-ID: 77d249c0-487c-1236-1997-00163edbaa97 CSeq: 115249948 INVITE Contact: <sip:mod_sofia@B.B.B.B:5080> User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20170927T175834Z~38f568d343~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 224 P-Charge-Info: <sip:222222222@mobile.tuenti.int>;npi=ISDN;noa=4 X-FS-Support: update_display,send_info Remote-Party-ID: "222222222" <sip:222222222@B.B.B.B>;party=calling;screen=yes;privacy=off If the client cancels the request, the CANCEL request that gets to the core pcscf looks like this: CANCEL sip:1111111111@mnc001.mcc001.3gppnetwork.org SIP/2.0 Via: SIP/2.0/UDP A.A.A.A;branch=z9hG4bK8735.ef7a61e110f335b2e92a8c1d430de585.0 Max-Forwards: 65 From: "222222222" <sip:222222222@B.B.B.B>;tag=eytD49ymQr71S To: <sip:1111111111@mnc001.mcc001.3gppnetwork.org> Call-ID: 77d249c0-487c-1236-1997-00163edbaa97 CSeq: 115249948 CANCEL Content-Length: 0 Reason: Q.850;cause=16;text="Normal Call Clearing" The processing of the CANCEL at the pcscf is eventually reaching the t_check_trans() function, but the logs return “no CANCEL matching found!” and thus the processing ends. I have followed the code up to the via_matching function in the t_lookup.c file of the tm module source and added some logs. The via matching is failing at the port comparison (the port stored for the INVITE is “5060”, the port stored for the CANCEL is “0”. The flow works ok with a different client (connected directly to the core) which includes the port in the Via headers. ¿Can this be a bug in kamailio or am I doing something wrong?. AFAIK The RFC 3261 does not mandate to include the ports in the sent-by element of the Via header Thanks! Carlos ________________________________ Este mensaje y sus adjuntos se dirigen exclusivamente a su destinatario, puede contener información privilegiada o confidencial y es para uso exclusivo de la persona o entidad de destino. Si no es usted. el destinatario indicado, queda notificado de que la lectura, utilización, divulgación y/o copia sin autorización puede estar prohibida en virtud de la legislación vigente. Si ha recibido este mensaje por error, le rogamos que nos lo comunique inmediatamente por esta misma vía y proceda a su destrucción. The information contained in this transmission is privileged and confidential information intended only for the use of the individual or entity named above. If the reader of this message is not the intended recipient, you are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited. If you have received this transmission in error, do not read it. Please immediately reply to the sender that you have received this communication in error and then delete it. Esta mensagem e seus anexos se dirigem exclusivamente ao seu destinatário, pode conter informação privilegiada ou confidencial e é para uso exclusivo da pessoa ou entidade de destino. Se não é vossa senhoria o destinatário indicado, fica notificado de que a leitura, utilização, divulgação e/ou cópia sem autorização pode estar proibida em virtude da legislação vigente. Se recebeu esta mensagem por erro, rogamos-lhe que nos o comunique imediatamente por esta mesma via e proceda a sua destruição

3 4

Kamailio new install, unable to create users
by Atux Atux 23 Nov '17

23 Nov '17

Hi. New to the area and trying to find my way around with kamailio. In one of my debian servers i need to have kamailio with a gui to start playing around with it. i am following the http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour to install kamailio. When running kamdbctl create it asks what format coding that i must have to configure the DB. i simply add latin1 and it goes on. then when i issue kamctl add daniel 1234qwet it comes with the following error: root@debian:~# kamctl add daniel 1234qwet MySQL password for user 'kamailio@localhost': ERROR 1045 (28000): Access denied for user 'kamailio'@'localhost' (using password: YES) ERROR: introducing the new user 'daniel' to the database failed root@debian:~# Any ideas on how to proceed with the following error please?

2 3

kemi with lua and dispatcher - attempt to index field 'dispatcher' (a nil value)
by Karsten Horsmann 23 Nov '17

23 Nov '17

Hello Daniel, hello List, after reading the fresh released kemi documentation (Thanks for that!) on github i start to configure an kemi lua configuration. It seems to be that there is not an dispatcher export to kemi in my installation. Or i missconfigured something? How to use dispatcher with kemi? 1.) First try to solve this - use param "register" "modulename" to app_lua - doesnt work 2.) grep around in the auto kemi in/export messages in debug startup - nothing to see. 3.) grep around in the kamcmd api.list command for lua - no dispatcher here the intressting parts of the configs and lua-script: kamailio.cfg ... loadmodule "dispatcher.so" loadmodule "permissions.so" loadmodule "avpops.so" #!ifdef WITH_CFGLUA loadmodule "app_lua.so" modparam("app_lua", "reload", 1) modparam("app_lua", "register", "dispatcher") #!endif #!ifdef WITH_CFGLUA modparam("app_lua", "load", "/etc/kamailio/lua/kamailio-kemi-lua.lua") cfgengine "lua" #!else import_file "request-route.cfg" #!endif ... kamailio-kemi-lua.lua function ksr_request_route() ... -- dispatch destinations if KSR.permissions.allow_source_address(FLT_EDGE) then KSR.info("new $fU -> $rU in permissions"); direction = "inbound"; routing_target = FLT_FSNODE; ksr_route_dispatch(direction, routing_target); KSR.x.exit(); end KSR.info("NOT-OKAY - $rU dropped"); -- End of Route() KSR.sl.sl_send_reply(407, "Proxy Authentication Required. See IP GUI"); return 1; ... end --- dispatcher route function ksr_route_dispatch(direction, routing_target) KSR.xlog.xinfo("new $fU -> $rU in routing_target ksr_route_dispatch"); KSR.info(routing_target); if (routing_target == nil) then KSR.xlog.xinfo("routing_target nil"); KSR.sl.sl_send_reply(604, "Does Not Exist Anywhere"); KSR.x.exit(); end if KSR.dispatcher.ds_select_dst(routing_target, 4) then send_reply(404, "No destination"); KSR.x.exit(); end ksr_route_myheader(); ksr_route_relay(); KSR.x.exit(); end /usr/sbin/kamailio[26763]: ERROR: app_lua [app_lua_api.c:713]: app_lua_run_ex(): error from Lua: /etc/kamailio/lua/kamailio-kemi-lua.lua:131: attempt to index field 'dispatcher' (a nil value) kamailio -v version: kamailio 5.0.4 (x86_64/linux) 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: unknown compiled on 20:20:24 Nov 16 2017 with gcc 4.8.5 rpm -aq | grep kam kamailio-mysql-5.0.4-26.el7.centos.x86_64 kamailio-5.0.4-26.el7.centos.x86_64 kamailio-lua-5.0.4-26.el7.centos.x86_64 kamailio-ims-5.0.4-26.el7.centos.x86_64 rpm -ql kamailio-5.0.4 | grep disp /usr/lib64/kamailio/modules/dispatcher.so /usr/share/doc/kamailio/modules/README.dispatcher /usr/share/kamailio/dbtext/kamailio/dispatcher -- Kind Regards *Karsten Horsmann*

2 5

"Federated" PBX solution with GUI to use with Kamailio - Suggestions needed
by George Diamantopoulos 23 Nov '17

23 Nov '17

Hello all! I'm looking at "PBX" platforms to use with kamailio as a load-balancer. This is not a strictly kamailio question, but I thought that if anyone has a suggestion to make, they would be watching this list. I'm investigating a use case with the following requirements: * Kamailio will act as a load-balancer, dispatching initial requests to PBX entities (e.g. Asterisk, Freeswitch) * PBX solution must have a single GUI interface to manage PBX functions (think FreePBX, FusionPBX etc) * Configuration via the GUI must apply to all PBX instances * Some sort of "asterisk-realtime + device state synchronization"-like setup must ensure every PBX box can service any request at any time (for failover) * Solution needn't be multi-tenant oriented, but if it is, it must work fine for single-tenant setups as well without multi-tenant capability getting in the way all the time * Feature list needn't be as full as, say, FreePBX's, but most common functionality must be available (inbound/outbound routing, trunks, queues, hunting to name a few) * Must be open source As a proprietary example, MiRTA PBX seems to check all the boxes above, apart from being open source. Now this "product" also offers high availability for databases and web apps, which is beyond the scope of this investigation. I'm only looking at a solution that provides a federated, synchronized farm of PBX instances. Any input would be greatly appreciated. Thanks! Best regards, George

3 2

Re: [SR-Users] Kamailio IMS + Asterisk PBX
by Amar Tinawi 22 Nov '17

22 Nov '17

My tests results : Unfortunately, no scenario was success, maybe because Asterisk does not support sh interface (Diameter) My test was for presence service and couldn’t get it done. The results of Having asterisk as an AS with kamailio IMS : with sip trunk configured towards S-cscf And whether the user is identified in Asterisk as extension user or not asterisk responded with 404 user not found. Sent from Mail for Windows 10

2 1

debian setup without registration
by Atux Atux 22 Nov '17

22 Nov '17

Hi. i am new to the are of kamailio. i am looking to install in a PC running debian 9 kamailio. that is easy since all i had to do was apt-get install kamailio. I am looking on how to setup the system in a way that everyone sending a register, to be able to register no matter what user name or passwd sends. i do not want a MYSQL server on that server. Could someone give me some hints on how to configure it please? In /etc/kamailio/kamctrlc i have SIP_DOMAIN=192.168.124.5 which is my IP. Everybody from my LAN 192.168.124.0/24 network must be able to send a register with any username and any passwd and be able to register. Also from the outside world, through port forward from my router. Some hints please?

3 2

kamcmd gives segmentation fault (5.0.4)
by José Lopes 22 Nov '17

22 Nov '17

Hello, I invoke kamcmd inside Freeswitch Lua Scripts and right now i am getting segmentation fault any time. This started to happen after I upgraded kamailio from 4.4 to 5.0.4 . I tried to analyse the core dump and it give me the next information. When I execute on the console, the command works. Am I doing something wrong? There is any alternative or better way to communicate with kamailio? Thanks in advance Best regards, Jose Lopes Core was generated by `/usr/sbin/kamcmd -s udp:127.0.0.1:2050 uac.reg_refresh s:29010017'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x000000000040340d in send_binrpc_cmd (s=<error reading variable: Cannot access memory at address 0x7ffec894a60c>, cmd=<error reading variable: Cannot access memory at address 0x7ffec894a600>, cookie=<error reading variable: Cannot access memory at address 0x7ffec894a608>) at kamcmd.c:591 591 kamcmd.c: No such file or directory. (gdb) bt #0 0x000000000040340d in send_binrpc_cmd (s=<error reading variable: Cannot access memory at address 0x7ffec894a60c>, cmd=<error reading variable: Cannot access memory at address 0x7ffec894a600>, cookie=<error reading variable: Cannot access memory at address 0x7ffec894a608>) at kamcmd.c:591 #1 0x0000000000403eee in run_binrpc_cmd (s=3, cmd=0x7ffec898a8d0, fmt=0x0) at kamcmd.c:924 #2 0x0000000000404499 in run_cmd (s=3, cmd=0x7ffec898a8d0, format=0x0) at kamcmd.c:1068 #3 0x00000000004073aa in main (argc=5, argv=0x7ffec898d1e8) at kamcmd.c:2114

2 2

Kamailio IMS + Asterisk PBX
by Amar Tinawi 21 Nov '17

21 Nov '17

Dears; Good day ! is it possible to use Asterisk pbx server as an Application server for the kamailio IMS users ? as if i want to correlate an IMS user with an extension number. My though is kamailio ims will handle the registration process and authentication and Asterisk will deal with IMS users as they are PBX users to get all asterisk's benefits (call conference, monitoring, waiting, voice mail ... etc) i'm thinking to integrate Asterisk with S-cscf, and when an IMS user register to the IMS to domain, a trigger will initiate the S-cscf to send notify to asterisk . i didnt find any solution or tutorial for this procedure, and idont know if it is possible to be done. Regards Ammar

5 8

Adding strip and preix attribute to Dispatcher table
by Ashutosh chaubey 21 Nov '17

21 Nov '17

Hi, Is there a way to add prefix and strip in dispatcher table attribute column. I have tried prefix=11,strip=3 but this seems to not work. Any help? Regards, Ashutosh Chaubey

2 4

http_async_client problem with tls_ca_path
by gmele 20 Nov '17

20 Nov '17

Hello, I'm having strange behavior with the http_async_client module and https connections. Our kamailio config use REST interfaces to send push to mobile apps when calls are initiated. We also use a REST interface to register the client in our push database (and then be able to use the info to send a push). Our REST interface use TLS to cypher exchanged data between the SIP registrar and the Pusher service. We have configured the tls connection using modparam: # ------ http async client params ------- modparam("http_async_client", "workers", 16) modparam("http_async_client", "connection_timeout", 20000) # 20 seconds timeout on HTTP requests modparam("http_async_client", "hash_size", 2048) # update this value depending of the load #modparam("http_async_client", "curl_verbose", 1) # curl debug modparam("http_async_client", "tls_ca_path", "/etc/kamailio/ssl/ca") What we have noticed is when there is 1 SIP Register requests triggering and HTTPS request all works fine, but as soon as we have several HTTPS request in //, the HTTPS request fails with curl error 77: curl: (77) Problem with the SSL CA cert (path? access rights?). We tried to put the Root CA Path when building the request: $http_req(all) = $null; # reset the parameters $http_req(method) = "POST"; $http_req(hdr) = "authorization: "+$hdr(authorization); $http_req(hdr) = "Content-Type: application/json"; $http_req(suspend) = 0; # don't suspend the transaction, continue routing script's execution $http_req(body) = "{ 'pushRegistration':{'callId':'" + $ci + "', 'fromUri':'" + $fU + "@" + $fd + "', 'contactLine':'" + $ct + "'}}"; $http_req(tls_ca_path) = "/etc/kamailio/ssl/ca/"; http_async_query("https://pusher:443/rest/push/register", "HTTP_REGISTER_REPLY"); But we have exactly the same behavior. Does the http_async_client module supports muti-thread? Regards Giovanni -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 17

Re: [SR-Users] 183 acc records even if early_media equals to 0
by Daniel-Constantin Mierla 20 Nov '17

20 Nov '17

Hello, can you run list command in gdb to see the code? Because I am not sure what is exactly at that line in your version with the patch. Cheers, Daniel On 09.11.17 18:10, Marco Capetta wrote: > I applied the patch but kamailio crashes with the following error: > > (gdb) bt full > #0 0x00007f8af1e9b9ae in relay_reply (t=t@entry=0x7f8a29c05868, > p_msg=p_msg@entry=0xffffffffffffffff, branch=branch@entry=0, > msg_status=<optimized out>, > cancel_data=cancel_data@entry=0x7ffd92443570, > do_put_on_wait=do_put_on_wait@entry=0) at t_reply.c:1968 > relay = 0 > [...] -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com Kamailio World Conference - www.kamailioworld.com

3 7

Dmq usage
by Aidar Kamalov 20 Nov '17

20 Nov '17

Hello, I want to distribute some messages-actions for redis to my kamailio servers. I use this code: For send: dmq_bcast_message("redis", "set early:$fU", "text/plain"); For recieve: request_route { xnotice("ROOT: $rm $rU [$ci]"); if(is_method("KDMQ")) { if($rU == 'redis') { route(REDIS); } else { dmq_handle_message(); } } ...... route[REDIS] { xnotice("REDIS: $rb"); redis_cmd("redis", "$rb", "r"); return; } When dmq_bcast_message starts - other node recieve 7 KDMQ sip packets. And not always it executed. Is it right way to use dmq? Dmq user locations are ok.

2 8

kamailio-wiki dead links to github.com/....master branch
by Karsten Horsmann 20 Nov '17

20 Nov '17

Hello, https://www.kamailio.org/wiki/devel/config-engines has links to https://github.com/kamailio/kamailio/blob/master/examples/kemi/kamailio-bas… for example configs. All the "kemi" stuff is gone. 404 Error. Where i found actuall (kemi / kamailio 5.0.x) samples? -- Kind Regards *Karsten Horsmann*

3 5

no event received for call disconnection
by Aleksandar Sosic 20 Nov '17

20 Nov '17

Hi Everyone, we're trying to implement cgrates in our kamailio nodes. Everything is working fine except for the fact that kamailio is not signaling the call end to cgrates for prepaid users... The versions, logs and confs are: ``` # kamailio -v version: kamailio 5.0.3 (x86_64/linux) 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: unknown compiled with gcc 5.3.1 ``` ``` # cgr-engine --version CGRateS 0.9.1~rc8 git+15a0793 (2017-10-23T12:15:40+02:00) ``` The kamailio conf regarding our problem is: ``` # Inform CGRateS about CALL_END (stop debit loops, perform accounting if desired in this way) route[CGR_CALL_END] { if $sht(cgrconn=>cgr) == $null { xlog("Charging controller unreachable"); exit; } $var(callDur) = $TS - $dlg(start_ts); evapi_relay("{\"event\":\"CGR_CALL_END\", \"callid\":\"$dlg(callid)\", \"from_tag\":\"$dlg(from_tag)\", \"cgr_reqtype\":\"$dlg_var(cgrReqType)\", \"cgr_tenant\":\"$dlg_var(cgrTenant)\", \"cgr_account\":\"$dlg_var(cgrAccount)\", \"cgr_destination\":\"$dlg_var(cgrDestination)\", \"cgr_answertime\":\"$dlg(start_ts)\", \"cgr_duration\":\"$var(callDur)\", \"cgr_supplier\":\"$dlg_var(cgrSupplier)\", \"cgr_disconnectcause\":\"$T_reply_code\"}"); } ``` The kamailio logs: ``` 17(114) ERROR: evapi [evapi_dispatch.c:707]: _evapi_relay(): failed to pass the pointer to evapi dispatcher 17(114) ERROR: evapi [evapi_mod.c:261]: w_evapi_relay(): failed to relay event: {"event":"CGR_CALL_END", "callid":"6gJ-ynyLm0-vS7StVQA3nMjwJ.YY257z", "from_tag":"Qf6tsQP.Js66FFOdCzv6QlEXomggf3gf", "cgr_reqtype":"*prepaid", "cgr_tenant":"evox.it", "cgr_account":"0409828030", "cgr_destination":"0039040123123", "cgr_answertime":"1509379822", "cgr_duration":"27", "cgr_supplier":"*any", "cgr_disconnectcause":"0"} 17(114) ERROR: evapi [evapi_dispatch.c:707]: _evapi_relay(): failed to pass the pointer to evapi dispatcher 17(114) ERROR: evapi [evapi_mod.c:261]: w_evapi_relay(): failed to relay event: {"event":"CGR_CALL_END", "callid":"6gJ-ynyLm0-vS7StVQA3nMjwJ.YY257z", "from_tag":"Qf6tsQP.Js66FFOdCzv6QlEXomggf3gf", "cgr_reqtype":"*prepaid", "cgr_tenant":"evox.it", "cgr_account":"0409828030", "cgr_destination":"0039040123123", "cgr_answertime":"1509379822", "cgr_duration":"27", "cgr_supplier":"*any", "cgr_disconnectcause":"0"} ``` Thanks for your help, -- Aleksandar Sosic mail: alex.sosic(a)timenet.it skype: alex.sosic cell: +385 91 2505 146

2 13

kamailio 4.4 - jansson_get_field parse error actions
by Jonathan Hunter 17 Nov '17

17 Nov '17

Hi Guys, I am looking to send calls to a default route should we have issues when connecting to our API which provides routing instructions. On scenario is handling an invalid JSON response , which would result in a parse error. Is it possible to capture when there is a failed to parse error when dealing with JSON responses to an http query in kamailio? In particular when using the jansson_get_field, I am looking to default to a route should their be a parse error, so I am forcing an erro, I get this in the logs, but can I grab to act on it? Nov 17 14:37:25 ip-172-31-34-165 /usr/local/kamailio-4.4/sbin/kamailio[9735]: ERROR: jansson [jansson_funcs.c:59]: janssonmod_get(): failed to parse: 200 OK Nov 17 14:37:25 ip-172-31-34-165 /usr/local/kamailio-4.4/sbin/kamailio[9735]: ERROR: jansson [jansson_funcs.c:61]: janssonmod_get(): json error at line 1: '[' or '{' expected near '200' Nov 17 14:37:25 ip-172-31-34-165 /usr/local/kamailio-4.4/sbin/kamailio[9735]: ERROR: jansson [jansson_funcs.c:59]: janssonmod_get(): failed to parse: 200 OK Nov 17 14:37:25 ip-172-31-34-165 /usr/local/kamailio-4.4/sbin/kamailio[9735]: ERROR: jansson [jansson_funcs.c:61]: janssonmod_get(): json error at line 1: '[' or '{' expected near '200' On receipt of this I would then like to hit a default route. Many thanks Jon

1 1

SUBSCRIBE NAT Keep-Alive
by Asgaroth 17 Nov '17

17 Nov '17

Hi All, I'd like to ask the guru's out there their opinion on how to solve the nat problem for presence subscriptions. We have an issue with subscriptions that get timed out and the notify clears the subscription out of the active_watchers table due to the notify timing out, I think this is expected and the right way to do it. However, how do others out there ensure that their subscription nat states are kept alive? For registrations we have nathelper, but, what is the sanest way of doing the same for subscriptions? I was originally thinking of sending notifications through our registrar/lookup server, but, I'm not sure if that is the correct way to do it. Any thoughts, tips, tricks would be much appreciated. Cheers

3 3

Kamailio questions
by Wilkins, Steve 17 Nov '17

17 Nov '17

Hello, I am new to Kamailio and to the Kamailio Mailing list and hope I am using it appropriately. I am using Kamailio version 5.0.3 (x86_64/linux), and it is installed and working on a Centos 7 Sever. I use Kamailio as a SIP Proxy Server for Asterisk and things seem to work fine, however, I do have an issue and a question for which I am trying to understand and am hoping someone can help me with. For the sake of the argument, my server name is myServer, and it's IP Address is 172.1.1.13; Asterisk is on a different server. Issue: Every 10 seconds I get an Error message in the Kamailio log which says => Nov 13 18:10:55 myServer /usr/local/sbin/kamailio[29188]: ERROR: <core> [core/tcp_main.c:2660]: tcpconn_1st_send(): connect 172.1.1.13:59794 failed (RST) Connection refused. I do not know what triggers this error, and as mentioned, things do seem to work; I can make calls and use sip tools which pass requests through the Kamailio to Asterisk. Does anyone know what could be causing this Error? I do not understand what is trying to connect to what at such a high port. Question: The way I understand it, rtpproxy is needed to pass video/audio through Kamailio to Asterisk. However, even without rtpproxy running, I get video through to Asterisk and the video plays with no issues on a sip device (e.g. MicroSIP). I am confused what I need rtpproxy for. Note: Even with no rtpproxy, I get the same error mentioned above in the Issue section. Thank you,

2 4

dr_load_routing_info(): failed to add rule id 4 -> skipping
by Karsten Horsmann 16 Nov '17

16 Nov '17

Hello List, i use kamailio-5.0.4 and i experiment with the drouting module with mysql-db. I try to route with "alphanumeric" stuff - like 'tetris' in sip://tetris@mytest.example https://www.kamailio.org/docs/db-tables/kamailio-db-devel.html#idp31352844 The DB-Field is an 64 string like route-id. In dr_rules.route-id i can use "named-routes" if i leave the prefix empty or i fill in only numbers. With strings i got this error: ERROR: drouting [dr_load.c:555]: dr_load_routing_info(): failed to add rule id 4 -> skipping SQL Result for rule-id 4: # dr_rules # ruleid, groupid, prefix, timerec, priority, routeid, gwlist, description '4', '9000', 'tetris', '', '0', 'tetris', '4', 'testroute' Is drouting only for "number-routing"? -- Kind Regards *Karsten Horsmann*

3 6

best practice to dynamic route block based on $rd - Domain in R-URI
by Karsten Horsmann 16 Nov '17

16 Nov '17

Hello List, i use kamailio-5.0.4 and i am now at the point that one kamailio is not enough :). So i decide to set another kamailio internal as sip-router based on drouting. My config is straight forward the example config with additional drouting stuff. Whats the best practice to route based on $rd with kamailio? I dont want to end with an bunch of if then else. Should be come from database. Pseudo Code: if ($rd == old.sbc.domain) { route[oldstuff]; } else { route[newstuff]; } Is there an module that i can use? -- Kind Regards *Karsten Horsmann*

1 1

How to use $var as arguments to module functions
by Duarte Rocha 16 Nov '17

16 Nov '17

Greetings, Is there a way to use the NATHELPER functions with a variable insted of a string with the value ? For example, I can do "fix_nated_sdp("11");" but Kamailio throws an error if i use "fix_nated_sdp("$(var(nat_fix)");" or any other variations. How should i proceed to be able to do this ? Thanks

2 1

play music while processing a http request
by 赵国杰 16 Nov '17

16 Nov '17

Hello guys, I have the following senario: ua1 calls ua2. when the INVITE reaches the kamailio, before foward to ua2, kamailio has to do a http request using “http_async_query”. However, the http request can take a really long time. I want the kamailio start playing music when the http request is sent and stop the music when the http response is received. My config file is like this: request_route { ... t_newtran(); http_async_query("url here", "HTTP_REPLY"); } route[HTTP_REPLY] { ... # forward the invite to ua2 } how do i complete the config? Thanks

2 3

Dmq_usrloc
by sofos koko 14 Nov '17

14 Nov '17

Hi ...i would like to connect two kamailio in diffwrent network so i can get call from one to another is any kamailio.cfg example to help me do it!

1 0

topos and topos_redis : error log -> ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store
by Abdoul Osséni 14 Nov '17

14 Nov '17

hello, After activated redis backend for topos module, I have some error logs. Can you help me to understand these logs? # kamailio -V version: kamailio 5.1.0-rc0 (x86_64/linux) 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: unknown compiled on 14:18:08 Nov 14 2017 with gcc 4.9.2 Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4805]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4805]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4807]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4807]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4808]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4808]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4808]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:51 sd-110402 /usr/local/sbin/kamailio[4808]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4805]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4805]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4806]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4806]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4807]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4807]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4807]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:52 sd-110402 /usr/local/sbin/kamailio[4807]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:53 sd-110402 /usr/local/sbin/kamailio[4806]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:53 sd-110402 /usr/local/sbin/kamailio[4806]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Nov 14 15:15:53 sd-110402 /usr/local/sbin/kamailio[4808]: INFO: topos_redis [topos_redis_storage.c:163]: tps_redis_insert_dialog(): no uuid for this message Nov 14 15:15:53 sd-110402 /usr/local/sbin/kamailio[4808]: ERROR: topos [tps_storage.c:394]: tps_storage_record(): failed to store Abdoul OSSENI

1 0

AVPOPS: is_avp_set/avp_check "name" parameter as variable.
by Asgaroth 14 Nov '17

14 Nov '17

Hi All, I was having a read of the AVP module documentation and I see mentioned that in future releases of kamailio that the AVP module may become deprecated in favour of using AVPOPS module. To this end, I started looking to switch some of AVP functions over to AVPOPS functions when I cam across an issue where I cannot have the name parameter of the avp to check a variable. I have some attributes defined for a domain which I load up using lookup_domain, one of these attributes is a dispatcher setid based on the request method being processed. So, with AVP I can do something like the following to see if it is set: if ( attr_exists( $_s(dm_ds_setid_$rm) ) ) The above works as expected, but if I try using something equivelent using avpops functions, something like: if ( is_avp_set("$avp(s:$_s(dm_ds_setid_$rm))") ) or if ( is_avp_set("$avp(s:dm_ds_setid_$rm)") ) or if ( is_avp_set("$avp($_s(dm_ds_setid_$rm))") ) or if ( is_avp_set("$avp(dm_ds_setid_$rm)") ) Then it fails the check even though the avp is set correctly, there are no errors in the logs so it all appears to be okay. Likewise, when checking with the following AVP function, it works: if ( attr_equals("$_s(dm_ds_setid_$rm)", "100") ) But, if I try with AVPOPS function, it fails (I'm guessing its bacause the avp cannot be found) if ( avp_check("$avp(s:$_s(dm_ds_setid_$rm))", "100") ) or if ( avp_check("$avp(s:dm_ds_setid_$rm)", "100") ) or if ( avp_check("$avp($_s(dm_ds_setid_$rm))", "100") ) or if ( avp_check("$avp(dm_ds_setid_$rm", "100") ) Are the AVPOPS functions able to take nested variables like the above? Is this a bug or am I using them incorrectly? I'm currently trying this out on kamailio v5.0.4. Any thoughts on the issue would be appreciated. Thanks

2 2

max_contacts does not work as expected
by Gholamreza Sabery 14 Nov '17

14 Nov '17

I have a Kamailio server that has: modparam("registrar", "max_contacts", 2) But recently I noticed some users have more than two records in the location table. For example one of the users has three different records with nearly the same information. The records are as follows. Is this behavior normal? (I am using Kamailio 4XX). *************************** 1. row *************************** id: 171296745 ruid: uloc-2c69-59db6c73-7ca-132d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: XpCnmaAhRF cseq: 43 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 14 *************************** 2. row *************************** id: 171296746 ruid: uloc-2c69-59db6c73-7f2-eb5d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: BzMKuyJN8C cseq: 50 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 23 *************************** 3. row *************************** id: 171296747 ruid: uloc-2c69-59db6c73-80f-541d92 username: 999999999999 domain: NULL contact: sip:999999999999@5.106.35.18:10362 received: sip:5.106.35.18:10361 path: NULL expires: 2017-11-14 08:45:50 q: -1.00 callid: 3p9P4PgZSv cseq: 51 last_modified: 2017-11-14 08:44:50 flags: 0 cflags: 64 user_agent: android_1.1.0_samsung_SM-E500H_7b271714c08b942aa821fc756cdd2e890a37069e/1.1.0 (belle-sip/1.4.2) socket: udp:192.168.103.21:5060 methods: NULL instance: <urn:uuid:8369ef6e-202b-4844-8928-492a7678c269> reg_id: 0 server_id: 11369 connection_id: -1 keepalive: 1 partition: 15

1 0

Load Gateway according hostname
by Ashutosh chaubey 13 Nov '17

13 Nov '17

Hi, I am trying to find a way to load a specific gateway but instead of lcr_id i want it load gateways according to the hostname. I am passing Hostname as a header to kamailio so its easy to access that but unabale to load gateways according to it. Any help? Regards, Ashutosh Chaubey

1 0

using bcrypt passwd hashing
by Walter Martín Villalba 13 Nov '17

13 Nov '17

Hello, I did some searches online and talked to some colleagues and it seems Kamailio only supports the traditional HTTP digest authentication, which uses MD5. I would like to know if any of you has been successful in using bcrypt/scrypt/pbkdf2 passwd hashing, instead of MD5, which has been deemed as obsolete and insecure a long time ago. Perhaps you've written your own auth module, or just modified the config script to call some other credential checking routine using a custom python/perl script (I'm thinking of doing the latter, of nothing better is available). If any of you have done something like this, using bcrypt or any other current and secure hashing algorithm, I would appreciate some guidance. If you haven't, aren't you concerned about storing MD5 password hashes in your database? Note: if I can't find a good answer using this list, I will try the developer's list next. Thanks in advance, Martín.

4 5

Kamailio issue
by bazou2 bazou2 13 Nov '17

13 Nov '17

Hello, I have a kamailio version 4.4 with an Asterisk 13 ... and i need to make a call from a sip phone app.(i have a user subscribed on the kamailio to be used on the sip phone app). The weird thing that first time, the call works (I can see the sip messages working fine between the app kamailio and asterisk) but when i try again it doesn't work. SIP messages show that the kamailio and Asterisk keep send trying request between them and the loop of these kind of messages doesn't end.. I need to know if i am missing something in the configuration file of the kamailio or what? Regards Sent from Outlook<http://aka.ms/weboutlook>

2 2

Development open in master branch (to be v5.2.x)
by Daniel-Constantin Mierla 12 Nov '17

12 Nov '17

Hello, git branch 5.1 was just created (to host the release series v5.1.x), therefore new features can now be pushed again in master branch. They will be part of the next future release, likely to be numbered 5.2.x. Any fixes that affect existing code in branches 5.1 or older version have to be backported - push first to master and then cherry pick -- see the contributing guidelines at: - https://www.kamailio.org/wiki/devel/git-commit-guidelines#backporting_commi… Many thanks to all contributors so far! Testing of branch 5.1 and giving feedback for it is very appreciated! Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

Branch 5.1 created
by Daniel-Constantin Mierla 12 Nov '17

12 Nov '17

Hello, the GIT branch 5.1 has just been created, it will host the release series 5.1.x. To get this branch from GIT, you can use: git clone https://github.com/kamailio/kamailio.git kamailio cd kamailio git checkout -b 5.1 origin/5.1 Hopefully in two-three weeks time frame the full release of 5.1.0 will be out. >From now on, any corresponding fix has to be pushed first to master branch and then cherry-picked to branch 5.1. No new features can get in branch 5.1. Enhancements to documentation or helping tools, as well as kemi exports are still allowed. If you are not sure about doing or not a backport, ask on sr-dev mailing list. Cheers, Daniel -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com

1 0

GeoIP2 module deb package
by Joel Serrano 12 Nov '17

12 Nov '17

Hi, I can't seem to find the geoip2 module for debian stretch (installing from packages, kamailio 5.0 branch). Is it not available in the repository? Thanks, Joel.

2 4

Kamilio for IMS and VoLTE Setup Questions
by Jack Folgolburg 11 Nov '17

11 Nov '17

Hello, I saw the Kamailio World 2017 presentation "Kamilio for IMS and VoLTE" (https://www.youtube.com/watch?v=Wdhcdv5JVPw). I would like to install the servers for evaluation and testing. The slides say to use Ubuntu 16.04, but there are no xenial packages at the ng-voice repository. Is it okay to install on Ubuntu 14.04, since there are trusty packages? I have done this and got through all the configuration prompts. After installation, is there any way to confirm that the site is operational? How to bring up the management console (i.e. what is local URL)? In the presentation, it looks like a commercial site at ng-voice is accessed. I understand that there is a REST API. Is this API documented anywhere? Maybe, the most important question is, after installation, what is the fastest way to add two users and test access from IMS clients? Thank you, James

1 0

183 acc records even if early_media equals to 0
by Marco Capetta 09 Nov '17

09 Nov '17

Dear All, I'm facing a strange problem with the call accounting module: even if in my configuration I have the parameter: modparam("acc", "early_media", 0) I can find some ACC records with sip_code 180 or 183. I investigated those cases and this issue seems to happen when an endpoint sends an 183 message immediately followed by a 200 OK. The issue seems started from kamailio version 4.4.1, but I cannot be really sure of this. Have you ever had a similar problem? Do you have any suggestions on how to solve it? Thanks Marco

2 11

t_relay_to_udp - 500 server error occurred (6/SL)
by Donat Zenichev 09 Nov '17

09 Nov '17

Hi all. I have a problem with t_relay_to_udp function. My topology: UAC -> tcp -> kamailio <-> udp <-> asterisk Kamailio processes requests from clients over TCP and relays requests to asterisk over UDP. Kamailio gets INVITE from client, relays it to asterisk, then he sends 100 trying to user client and abruptly sends 500 internal error to client (after 100 trying). But the main interesting thing is, that asterisk answers on INVITE and dialog continues between kamailio and asterisk (without user agent client) - it reaches the destination then, without originator : ) How error on kamailio looks like: Nov 8 15:23:37 ip-10-2-1-229 kamailio[10339]: ERROR: tm [t_fwd.c:1716]: t_forward_nonack(): ERROR: t_forward_nonack: no branches for forwarding Nov 8 15:23:37 ip-10-2-1-229 kamailio[10339]: ERROR: sl [sl_funcs.c:363]: sl_reply_error(): ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (6/SL) The same problem is, when I change transport protocol between kamailio and asterisk to TCP, using the t_relay_to_tcp function. When I use t_relay function, it works fine, nothing happens that can drop dialogs between kamailio and clients. Relaying code part: if ($var(to_asterisk)=="1") { xlog("L_NOTICE","Relaying to asterisk - change socket to udp:10.1.1.1:5070 \n"); force_send_socket(udp:10.1.1.1:5070); if (!t_relay_to_udp()) { sl_reply_error(); } } I need to use t_relay_to_udp with force_send_socket, because kamailio changes the source port (source socket) every time he generate new INVITEs to asterisk and this is irrelevant. Dump between kamailio and user agent client: INVITE sip:1111111111example@kam1.domain.com:5070;transport=TCP SIP/2.0 Via: SIP/2.0/TCP uac.domain.example:5065;branch=z9hG4bK-d8754z-67b9ad24dbc5c1d0-1---d8754z- Max-Forwards: 70 Contact: <sip:debug.device-example@uac.domain.example:5065;transport=TCP> To: <sip:1111111111example@kam1.domain.com:5070;transport=TCP> From: <sip:debug.device-example@kam1.domain.com:5070 ;transport=TCP>;tag=93394e63 Call-ID: NzQ1NWU5NzUxYmY4ZWVjMjYwNDYwMTBmYzYxODZkZmM. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Proxy-Authorization: Digest username="debug.device-example",realm=" kam1.domain.com ",nonce="WgQGlVoEBWnhH3CF6DE77uKxKJx0tT4e",uri="sip:1111111111example@kam1.domain.com:5070 ;transport=TCP",response="4753aa849715d26d0dde1b3c2314c843",a orithm=MD5 Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.3.25608 r25552 Allow-Events: presence, kpml Content-Length: 241 v=0 o=Z 0 0 IN IP4 uac.domain.example s=Z c=IN IP4 uac.domain.example t=0 0 m=audio 8000 RTP/AVP 8 3 110 0 98 101 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/TCP uac.domain.example:5065;branch=z9hG4bK-d8754z-67b9ad24dbc5c1d0-1---d8754z-;rport=43486 To: <sip:1111111111example@kam1.domain.com:5070;transport=TCP> From: <sip:debug.device-example@kam1.domain.com:5070 ;transport=TCP>;tag=93394e63 Call-ID: NzQ1NWU5NzUxYmY4ZWVjMjYwNDYwMTBmYzYxODZkZmM. CSeq: 2 INVITE Server: MS Lync Content-Length: 0 SIP/2.0 500 I'm terribly sorry, server error occurred (6/SL) Via: SIP/2.0/TCP uac.domain.example:5065;branch=z9hG4bK-d8754z-67b9ad24dbc5c1d0-1---d8754z-;rport=43486 To: <sip:1111111111example@kam1.domain.com:5070 ;transport=TCP>;tag=9a78ee8ed9a3d4ba441c41d83ba2e904.4113 From: <sip:debug.device-example@kam1.domain.com:5070 ;transport=TCP>;tag=93394e63 Call-ID: NzQ1NWU5NzUxYmY4ZWVjMjYwNDYwMTBmYzYxODZkZmM. CSeq: 2 INVITE Server: example Content-Length: 0 Thanks in advance for hints. -- -- BR, Donat Zenichev Wnet VoIP team Tel: +380(44) 5-900-800 http://wnet.ua

1 0

Topoh module configuration for SIP proxy and SIP registrar
by Christian Conus 08 Nov '17

08 Nov '17

Hello, I am trying to configure the topoh module for kamailio 5.0.2. I read the documentation but some points are still not clear to me. The topology of my network is as follow: - Two SIP proxies - Two SIP registrars - Only one SIP proxy is running (the other take over automatically in case the first fails) - Both registrars are running (proxy sends requests to both registrars using round robin algorithm) Text mode graph of the topology (hope it helps understand): I tried to enable topoh on SIP proxy only, on both proxy and registrars, with same mask_key everywhere and with different mask_key between proxy and registrars but everytime I could still see parts of the SIP header which was not obfuscated. My questions are: - Is it necessary to enable topoh module on each SIP proxy and SIP registrar ? - The mask_key should be the same on each server or it should be different between proxy and registrars? - the mask address should be the same on each server or it should be different between proxy and registrars ? Thanks in advance for your help, Christian -- Sent from: http://sip-router.1086192.n5.nabble.com/Users-f3.html

2 10

evapi_async_relay() failed
by Abdul Basit 07 Nov '17

07 Nov '17

Hi list, I am configuring CGRateS with kamailio 4.4 using deb packages on debian 8. I am facing a issue while processing event_route[dialog:start] for answering call INVITE at evapi_async_relay("{\"event\":\"CGR_CALL_START\", with following error /usr/sbin/kamailio[13887]: ERROR: evapi [evapi_mod.c:288]: w_evapi_async_relay(): failed to suspend request processing However, after discussion with Dan on IRC i replaced function evapi_async_relay() with evapi_relay() resolved the issue. Whats causing this error? Can anyone suggest some way out for keep using evapi_async_relay? -- regards, abdul basit

2 1

Kamailio SIP-I
by Saleem Manasfi 07 Nov '17

07 Nov '17

Dears, Please if you can let us know how does Kamailio reads the SIP-I body (ISUP)? Is there any document or link that may help?

2 1

Re: [SR-Users] Regarding Opensource DID Routing Solution using Kamailio
by surendra tiwari 06 Nov '17

06 Nov '17

Hi All, I have used the kamailio and redis to create the DID routing solution.please find below in github.this repository is created to help the person who want to setup there own DID routing solution with following feature: 1. Inbound Termination with Carrier IP validation. 2. Carrier LCR for DID/TFN to PSTN forwarding. 3. Inbound Abuse Block. 4. CDR in MongoDB. 5. IPTables Block for Sip-Scanners. 6. Integration with RtpEngine. 7. RedisDB for quick DB Access. https://github.com/surendratiwari3/KamInboundSIP On Mon, Nov 6, 2017 at 9:56 PM, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: > Hello, > > your email was addressed to kamailio-announce mailing list (I took your > message from its bounces filter), but that list is only for announcement > issued by Kamailio project (major news about our releases, security > incidents, events, etc...). > > News from community members are welcome and can be sent to: > > - sr-users(a)lists.kamailio.org for non-commercial announcements > > - business(a)lists.kamailio.org for announcements related to commercial > offerings/products (but here you can also send non-commercial announcements) > > Your announcement is not commercial and can be sent to sr-users. Once it > will be there, I will also make a news article on kamailio.org website. > > I didn't have the time yet to check in detail what you put together, but > sounds interesting. > > Thanks for sharing, > Daniel > > (cc-ed Fred as he is moderator for kamailio announce mailing list) > > On 01.11.17 18:46, surendra tiwari wrote: > > Hi All, > > I have used the kamailio and redis to create the DID routing > solution.please find below in github.this repository is created to help the > person who want to setup there own DID routing solution with following > feature: > > > 1. Inbound Termination with Carrier IP validation. > 2. Carrier LCR for DID/TFN to PSTN forwarding. > 3. Inbound Abuse Block. > 4. CDR in MongoDB. > 5. IPTables Block for Sip-Scanners. > 6. Integration with RtpEngine. > 7. RedisDB for quick DB Access. > > > https://github.com/surendratiwari3/KamInboundSIP > > -- > With Regards, > Surendra Tiwari > VoIP Application Developer, > +91-9967609476 <+91%2099676%2009476> > > > -- > Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training, Nov 13-15, 2017, in Berlin - www.asipto.com > Kamailio World Conference - www.kamailioworld.com > > -- With Regards, Surendra Tiwari VoIP Application Developer, Plivo Comm +91-9967609476

1 0

Sipcapture HEP Values
by Soltanici Ilie 06 Nov '17

06 Nov '17

Hi all, How can i get the Destination IP from HEP packet? I tried $hep(dst _ip) like in documentation (https://www.kamailio.org/wiki/cookbooks/devel/pseudovariables#sipcapture_mo…) - but is not working. (pvar "hep" has an invalid name param [dst_ip]). Thank You.

1 0

Invalid JSON when dumping htable with JSONRPC-S
by Marrold 03 Nov '17

03 Nov '17

Hi All, I am currently trying to retrieve the contents of a htable using the JSONRPC-S module over HTTP, however I have observed if a slot contains multiple values, the JSON returned has duplicate keys which could be considered invalid. Here are the relevant lines from kamailio.cfg - event_route[xhttp:request] { if ($hu =~ "^/carrier") { jsonrpc_exec('{"jsonrpc": "2.0", "method": "htable.dump", "params" : {"htable": "carrier"}, "id": 1}'); xhttp_reply("200", "OK", "text/html", "$jsonrpl(body)"); } } And the output, showing a slot with two items- { "entry": 5, "size": 2, "slot": { "item": { "name": "carrierA", "value": "90", "type": "str" }, "item": { "name": "carrierB", "value": "190", "type": "str" } } } And the same from `kamcmd -s unix:/tmp/kamailio_ctl htable.dump carrier` - { entry: 5 size: 2 slot: { item: { name: carrierA value: 90 type: str } item: { name: carrierB value: 190 type: str } } } Would it be possible to change it to something like the following - { "entry": 5, "size": 2, "slot": [ { "name": "carrierA", "value": "90", "type": "str" }, { "name": "carrierB", "value": "190", "type": "str" } ] } Any other suggestions or work arounds would be appreciated, Thanks Matthew

2 1

Contacts in 302
by Diego Nadares 03 Nov '17

03 Nov '17

Hi Guys, I'm generating a 302 reply from kamailio. In this 302 I append new branches with new contacts. if($var(routing)=~"redirect"){ jansson_get("contacts_len", "$var(evmsg)", "$var(contacts_len)"); xlog("L_INFO", "Contacts len $var(contacts_len)"); $var(i) = 0; while ($var(i) < $var(contacts_len)){ jansson_get("contacts[$var(i)]", "$var(evmsg)", "$var(contact)"); append_branch($var(contact), "0.5"); $var(i) = $var(i) + 1; } send_reply("302", "Moved Temporarily"); exit; } The problem i'm facing is that I can't delete the original contact *<sip:1111@188.111.111.111:5060 <http://sip:1111@188.111.111.111:5060>>* The resulting conctact: Contact: *<sip:1111@188.111.111.111:5060 <http://sip:1111@188.111.111.111:5060>>*, <sip:1111@188.111.111.112>;q=0.5, <sip:1111@188.111.111.112>;q=0.5 I all ready tried with remove_hf('Contact') and adding the new ones after that but it doesn't worke either. The problem is that one gw takes the first contact over and over again and never the other two. Thanks in advance. Diego.

2 4

2025

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users