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37 participants
37510 discussions

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

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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

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