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37394 discussions

Deletion of Prometheus metrics
by Marat Gareev 07 Aug '20

07 Aug '20

Hello, I have confuse about this timeout: https://www.kamailio.org/docs/modules/devel/modules/xhttp_prom.html#xhttp_p… Is there reason for forced deletion of unused metrics? Let's imagine a situation: Zabbix reads metrics at a certain interval, unused metrics are deleted if they are not used for a period of time (xhttp_prom_timeout). I drew 2 cases when Zabbix will receive the same value, but we cannot say unambiguously whether these are new values or not. [image: xhttp_prom_timeout.jpg] Regards, Marat

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Route to PSTN no works
by sip user 07 Aug '20

07 Aug '20

Hi, I have kamailio connect to Teams, and works form Asterisk -> Teams calls. For Teams -> Asterisk calls I'd worked using extension and register Asterisk with that extension. But I'd like to use direct routing with IP. In kamailio.cfg I activate define WITH_PSTN. I configured the IP and PORT for my PSTN. I'm using the default route[PSTN]: route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined xlog("L_INFO","PSTN ACTIVADO"); if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN routing enabled but pstn.gw_ip not defined\n"); return; } # route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(\+|00)[1-9][0-9]{3,20}$")){ xlog("L_INFO", "Error en el formato numerico!!"); return; } # only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; } # normalize target number for pstn gateway # - convert leading 00 to + #if (starts_with("$rU", "00")) { # strip(2); # prefix("+"); #} if (strempty($sel(cfg_get.pstn.gw_port))) { #$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip); xlog("L_INFO","SELECCION CON PUERTO"); $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } else { xlog("L_INFO","SELECCION CON PUERTO"); $ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip) + ":" + $sel(cfg_get.pstn.gw_port); } route(RELAY); exit; #!endif return; } And in my request_route: remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { if($src_ip != "IP ASTERISK"){ xlog("L_INFO", "***********ROUTE PSTN***********"); route(PSTN); } else { xlog("L_INFO","LLamada desde $si con puerto $sp"); record_route_preset("FQND:5061;transport=tls", "IP KAMAILIO:5060"); add_rr_param(";r2=on"); route(DISPATCH); route(RELAY); } } But never see that the call go to PSTN route.. I'd made any wrong?? Thanks

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Info: 2 Kamailio presentations soon live at ClueCon - 1:30pm CST (20:30 Berlin)
by Daniel-Constantin Mierla 06 Aug '20

06 Aug '20

Hello, quick info for those that want to watch the 2 presentations about Kamailio at ClueCon Deconstructed, starting in about 1 hour 40 minutes from now (or 13:30CST, 20:30Berlin/Paris/Madrid/...). One will be done by me and the 2nd by Fred Posner. A few more details are available at: - https://www.kamailio.org/w/2020/07/cluecon-deconstructed-2020/ The sessions should be live streamed on youtube - should appear in the freeswitch channel there at the time of start: - https://www.youtube.com/c/FreeSWITCH/ Of you can join the video conference room for a more interactive participation -- broadcast room listed at: - https://cluecon.cantina.video/#/home Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

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Audio on Calls Stopping After 15 Minutes
by admin＠conradit.com 06 Aug '20

06 Aug '20

All, hi: I'm a Kamailio newbie! I've managed to set up a proxy server with the minimal configuration file found here: https://github.com/kamailio/kamailio/blob/master/misc/examples/mixed/kamail…. The issue that I'm having is that the audio on calls simply stop precisely after 15 minutes. It's always precisely at 15 minutes. The calls continue connected but without any audio. A summary about my set up: 1. This is an Amazon AWS EC2 server so it doesn't have a public IP directly attached to the interface. This makes me believe that the issue might be related to NAT. 2. I use Flowroute as the SIP trunk for these tests. 3. My phone system, for testing purposes, works on UDP port 6060 an the RTP ports are 20,000 to 30,000. 4. I am not using, for the time being, TLS nor SRTP. 5. My phone system is inside the same subnet as the Kamailio server. Questions: 1) Am I missing the NATHELPER module and if so how could we combine it to the minimal proxy configuration described above? Assuming that my Kamailio server uses local IP 10.80.10.5 and the PBX 10.80.10.6, both on UDP port 6060 for SIP. 2) May I simply combine the example file for nathelper.cfg as per this URL: https://github.com/kamailio/kamailio/blob/master/misc/examples/mixed/nathel… to the proxy example files, change IP addresses, ports, etc.? If not, how would one go about adding the NATHELPER module to the minimal proxy configuration? 3) Could it be some other issue that is causing the 15-minute interruption on audio? Thank you so much for your help!

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Use-case does Kamailio support PBX - webrtc (web sockets)
by Johnny Ritzer 06 Aug '20

06 Aug '20

We have a PBX that doesn’t support websockets . But we want a PWA webaapp ctxSIP that uses sip.js to work . Due to its not SIP generic do we setup like this PBX <----> Kamailio <---->webapp Im looking for developers for android build via react native or ionic framework but not sure what solution/archtectiure Question. Does Kamailio register the SIP user/pass on the Kamailio server meaning it keeps it registered. And thus the ctxSIP client app just handles the pass thru for that connection. What about RTP do I need a separate handler for that. Tha nks

2 1

Info: Kamailio Getting Started Workshop at ClueCon - Live in 1 hour
by Daniel-Constantin Mierla 05 Aug '20

05 Aug '20

Hello, in about 1 hour I will do a quick (1 hour long) workshop at ClueCon deconstructed. It can be watched live on youtube -- should appear in their youtube channel: * https://www.youtube.com/channel/UC0i2xbniNHJQNfPPs_v4T0Q Or if you want to join the live video conference, which allows you to ask questions and interact with the participants, use an webrtc capable browser and go to: * https://cluecon.cantina.video/ Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Funding: https://www.paypal.me/dcmierla

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kamailio and rtpengine_manage() / rtpengine_delete()
by Karsten Horsmann 05 Aug '20

05 Aug '20

Hi list, I use rtpengine 7.2.1 branch with kamailio 5.2.x and 5.3.x in different setups. Due to some issues with my 5.2.x config I saw the "Total timed-out sessions via TIMEOUT" grow but "Total regular terminated sessions never" goes up from 0. That was an indicator for an non-existing rtpengine_manage() or rtpengine_delete() at the end of dialogs. Okay. Now I want to observe that a bit more. I use parallel forking to webrtc clients, that i guess will be the "more Offers" than Answers. Any hints on what numbers are important? rtpengine-ctl -ip 127.0.0.1:4444 list totals Total statistics (does not include current running sessions): Uptime of rtpengine :27726 seconds Total managed sessions :3435 Total rejected sessions :0 Total timed-out sessions via TIMEOUT :0 Total timed-out sessions via SILENT_TIMEOUT :0 Total timed-out sessions via FINAL_TIMEOUT :0 Total timed-out sessions via OFFER_TIMEOUT :0 Total regular terminated sessions :3435 Total forced terminated sessions :0 Total relayed packets :69127802 Total relayed packet errors :5063 Total number of streams with no relayed packets :128 Total number of 1-way streams :8 Average call duration :233.321876 Graphite interval statistics (last reported values to graphite): Total calls duration :0.000000 Min managed sessions :0 Max managed sessions :0 Min/Max/Avg offer processing delay :0.000000/0.000000/0.000000 sec Min/Max/Avg answer processing delay :0.000000/0.000000/0.000000 sec Min/Max/Avg delete processing delay :0.000000/0.000000/0.000000 sec Min/Max/Avg offer requests per second :0/0/0 per sec Min/Max/Avg answer requests per second :0/0/0 per sec Min/Max/Avg delete requests per second :0/0/0 per sec Control statistics: Proxy | Offer | Answer | Delete | Ping | List | Query | StartRec | StopRec | Errors | BlkDTMF | UnblkDTMF 127.0.0.1 | 3896 | 3757 | 3499 | 0 | 0 | 0 | 0 | 0 | 0 | 0 | 0 Kind Regards -- Mit freundlichen Grüßen *Karsten Horsmann*

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Redundancy of Progress Call using Kamailio
by Amit Pal 04 Aug '20

04 Aug '20

Hi Team , Hope all is fine. My requirement : I have a customer who wants high availability for ringing calls and calls that in process of being set up. Can you suggest an alternate mechanism to achieve replication of transaction state. Can you guide how it can be achieved in Kamailio or is there a mechanism to get commercial support to achieve the desired objective. I am waiting your response. Thanks Regards Amit Pal

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Kamailio Kubernetes AWS NLB UDP
by Ewgeny Berladin 04 Aug '20

04 Aug '20

Hi ! Does anyone have success Kamailio deployments via service on AWS EKS with UDP port exposing ? apiVersion: v1 kind: Service metadata: name: sip-proxy annotations: service.beta.kubernetes.io/aws-load-balancer-type: "nlb" spec: selector: app: kamailio-sip externalTrafficPolicy: Local type: LoadBalancer ports: - name: sip-up protocol: UDP port: 5060 As result we have this ERROR: ---- ------ ---- ---- ------- Normal EnsuringLoadBalancer 3m20s (x7 over 8m35s) service-controller Ensuring load balancer Warning SyncLoadBalancerFailed 3m20s (x7 over 8m35s) service-controller Error syncing load balancer: failed to ensure load balancer: Only TCP LoadBalancer is supported for AWS ELB As I know this is a BUG in AWS EKS. Maybe there’s some workarounds ? Kubernetes v 1.17 Thanks

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ACC events missing to-tag when doing t_reply
by Julien Chavanton 04 Aug '20

04 Aug '20

Hi, considering this example : modparam("acc_json", "acc_missed_flag", 6) failure_route[test] { setflag(6); // MISSED_ACC, the failure will be recorded by ACC t_flush_flags(); // This will set the flags in the newly created transaction t_reply("500", "Service Unavailable"); } This does generate an ACC event, however the to-tag is not populated, my guess this is because the TM callback is done before the to-tag is generated. I wonder if there is a way to populate the to-tag immediately ? I will dig it further, but maybe there is a trick I am missing

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