April 2019 - sr-users - kamailio.org

Via header force change protocol to TLS?
by Anthony Alba 30 May '24

30 May '24

I have kamailio behind a TLS termination proxy so the sockets are correctly deduced to be TCP. However the clients only talk TLS to the proxy and are confused when the top Via header added by Kamailio is TCP. Is there a way for Kamailio to forcibly pretend its protocol is TLS? Like advertised_address but "advertised_protocol" instead. (With pjsip testing: it has a flag use_tls which ignores TCP from Kamailio and continues to use the persistent TLS transport to proxy. Linphone fails because it tries to honor TCP in Via and is unable to establish TCP transport). BTW I am using t_relay_to_tcp so Kamailio will return traffic to the proxy as TCP even though the contact addresses specify transport=TLS.

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Re: [SR-Users] TOPOS (5.1.[23]) breaks PRACK
by Daniel Tryba 02 May '23

02 May '23

On Mon, May 07, 2018 at 04:44:14PM +0200, Daniel Tryba wrote: > Sure. Attached. Problem appears to be that the topos query can't find > callid-totag (from the response). > > I'll try the same scenario with the mysql backend to see if it behaves > different. Config works fine with mysql as topos backend. So the bug is restricted to topos-redis.

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Re: [SR-Users] t_drop_replies not working with t_suspend in failure route
by Huber Andreas 03 Sep '21

03 Sep '21

Hi, We’re still using kamailio 4.4 but we’ll be migrating to 5.0 soon. Cool so it will be fixed when we migrate ! Thanks, Andreas From: sr-users [mailto:sr-users-bounces@lists.kamailio.org] On Behalf Of Federico Cabiddu Sent: vendredi 12 mai 2017 11:56 To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] t_drop_replies not working with t_suspend in failure route Hi, which version are you using? A similar case had been reported some months ago and it should be fixed in 5.0. Regards, Federico On Fri, May 12, 2017 at 11:44 AM, Huber Andreas <andreas.huber(a)nagra.com<mailto:andreas.huber@nagra.com>> wrote: Hello, We have a use case where we suspend a transaction in a failure_route to give UEs that might be woken by a push notification more time to REGISTER and join the INVITE. We’d like to drop the previous branches in this case. I tried using t_drop_replies() but it has no effect. The doc states that t_drop_replies() is only working if a new branch is added. And from my understanding t_suspend() adds a new branch. But is it possible that t_drop_replies() cannot be used with t_suspend()? Or am I missing something? Kind Regards, Andreas _______________________________________________ Kamailio (SER) - Users Mailing List sr-users(a)lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users

2 1

nat_uac_test(8)
by Sergiu Pojoga 24 Apr '20

24 Apr '20

When receiving an INVITE over a specific LTE carrier, I'm seeing 'c=IN IP4 192.0.0.4' in SDP, which isn't technically a RFC1918 or RFC6598 IP address and thus nat_uac_test(8) fails. What elegant workaround can be done to catch such specific cases? Thanks.

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need help with RTPProxy in bridged mode
by Grant Bagdasarian 04 Apr '20

04 Apr '20

Hello, I'm using RTPproxy for the first time in bridged mode and I can't get kamailio/rtpproxy to rewrite the c parameter to the correct public ip address of kamailio. The setup is as following: Carrier ------[fiber]------ Kamailio ---------[lan]--------- Freeswitch Kamailio is listening on two interfaces: 1) Private: 172.0.0.1 2) Public: 192.168.0.1 (since we have a dedicated fiber with our carrier, this is its public address) Freeswitch is listening on: 1) 172.0.0.2 Carrier is on: 1) 10.0.0.1 I've started an rtpproxy instance on the Kamailio box using: rtpproxy -s udp:127.0.0.1:7721 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -l 192.168.0.1 172.0.0.1 I've played around with rtpproxy_manage() and the various flags (ie, ei), but I can't get kamailio to set the correct public IP when the 200 OK has to be sent back to the carrier. It always sets it to its private address, instead of its public address. I'm using Kamailio 4.2 with sippy/rtpproxy 2.0. Could someone please point me into the right direction? Thanks! Grant

4 8

Kamailio OPTIONS Round-Trip
by Soltanici Ilie 27 Mar '20

27 Mar '20

Hi, With Asterisk, we are able to get some peer round-trip connection statistic by setting qualify=yes for the specified peer. It sends periodic OPTIONS to the peer and calculates the time round trip time. It's something like - "Status: OK (30 ms)". Is there any way to achieve this in Kamailio by using nathelper module, or any other? Thank you.

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Kamailio stop to process incoming SIP traffic via TCP.
by Kristijan Vrban 16 Jan '20

16 Jan '20

Hi kamailios, i have a creepy situation with v5.2.1 stable Kamilio. After a day or so, Kamailio stop to process incoming SIP traffic via TCP. The incoming TCP network packages get TCP-ACK from the OS (Debian 9, 4.18.0-15-generic-Linux) but Kamailio does not show any processing for the SIP-Traffic incoming via TCP. No logs, nothing. While traffic via UDP is working just totally fine. When i look via command "netstat -ntp" is see, that the Recv-Q get bigger and bigger. e.g.: Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name tcp 4566 0 172.17.217.12:5060 xxx.xxx.xxx.xxx:57252 ESTABLISHED 31347/kamailio After Kamailio restart, all is working fine again for a day. We have maybe 10-20 devices online via TCP and low call volume (1-2 call per minute). The only settings for tcp we have is "tcp_delayed_ack=no" How to could we debug this situation? Again, no error, no warings in the log. Just nothing. Kristijan

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WebRTC ACK Protocol
by Ilie Soltanici 08 Jan '20

08 Jan '20

Hello, I am trying to set up a WebRTC2SIP Gateway by using Kamailio and rtpengine. So far, everything is working fine, I'm able to register an extension and make a call, but for some reason, when i'm trying to call a WebRTC extension from any SIP Extension Kamailio is sending INVITE, WebRTC extension is sending back 200 OK, and then Kamailio is trying to send an ACK through UDP protocol, and not through wss, as it's supposed to do. This is how invite is looking: INVITE sip:nl7oe4ss@vjbh7r4im6j7.invalid;transport=wss SIP/2.0 Record-Route: <sip:my-company.net ;transport=udp;ftag=as1789445c;lr=on;nat=yes> Via: SIP/2.0/WSS 123.123.123.123:10443 ;branch=z9hG4bKe655.29d7c135a302f3eb803902d4f5a8da7e.0 Via: SIP/2.0/UDP 192.168.50.237:5060 ;received=192.168.50.237;branch=z9hG4bK7d2e534e;rport=5060 Max-Forwards: 70 From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c To: <sip:15@192.168.50.210:5060> Contact: <sip:11@192.168.50.237:5060> Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net CSeq: 102 INVITE User-Agent: Proxy Date: Wed, 03 Apr 2019 17:11:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Alert-Info: Content-Type: application/sdp Content-Length: 596 Server: SIP Proxy and then WebRTC app is replying with 200 OK: SIP/2.0 200 OK Record-Route: <sip:my-company.net ;transport=udp;ftag=as1789445c;lr=on;nat=yes> Via: SIP/2.0/WSS 123.123.123.123:10443 ;branch=z9hG4bKe655.29d7c135a302f3eb803902d4f5a8da7e.0 Via: SIP/2.0/UDP 192.168.50.237:5060 ;received=192.168.50.237;branch=z9hG4bK7d2e534e;rport=5060 To: <sip:15@192.168.50.210:5060>;tag=dk4fa8ftt6 From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net CSeq: 102 INVITE Contact: <sip:nl7oe4ss@vjbh7r4im6j7.invalid;transport=wss> Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: Proxy-WEBRTC Content-Type: application/sdp Content-Length: 901 and finally, Kamailio is trying to send this ack through UDP protocol: ACK sip:nl7oe4ss@22.22.22.22:57421;transport=wss SIP/2.0 Via: SIP/2.0/UDP 192.168.50.237:5060;branch=z9hG4bK56363ddf;rport Route: <sip:my-company.net;transport=udp;ftag=as1789445c;lr=on;nat=yes> Max-Forwards: 70 From: "WebRTC" <sip:11@my-company.net>;tag=as1789445c To: <sip:15@192.168.50.210:5060>;tag=dk4fa8ftt6 Contact: <sip:11@192.168.50.237:5060> Call-ID: 7fc800de060197fa2315c93763873092(a)my-company.net CSeq: 102 ACK User-Agent: Proxy Content-Length: 0 If i'm trying to force it through TLS, i'm receiving error: get_send_socket2(): protocol/port mismatch (forced tls:123.123.123.123:10443, to udp:22.22.22.22:23317) Can someone point me in the right direction, please? Thank you.

3 4

rtpproxy does not response, disable it
by Zodiac 03 Jan '20

03 Jan '20

Dear friends, I am working on a program on Kamailio and rtpengine proxy. I am wondering whether can I set Kamailio and rtpengine daemon on different physical machines. For example, I set Kamailio on a machine with IP address:10.109.247.80, and launch rtpengine daemon on another machine with interface parameter as 10.109.247.90 and ng port 7723. I set parameter in Kamailio.cfg with modparam(“rtpengine”, “rtpengine_sock”, “udp:10.109.247.90:7723”). Unfortunately I got debug message like this: ERROR: rtpengine [rtpengine.c:1710]: send_rtpp_command(): can't send command to a RTP proxy ERROR: rtpengine [rtpengine.c:1746]: send_rtpp_command(): proxy <udp:10.109.247.90:7723> does not respond, disable it ERROR: rtpengine [rtpengine.c:1616]: rtpp_test(): proxy did not respond to ping And, I also tried to set Kamailio and rtpengine daemon in a same machine,and use modparam(“rtpengine”, “rtpengine_sock”, “udp:localhost:7723”). And Kamailio can work functionally under this situation. rtpengine daemon can receive ping message from Kamailio and rtpengine daemon can work as suspected. So for the later case, is it supposed that Kamailio be in the same machine with same localhost address? Otherwise, what’s the reason for my ERROR? ------------------------------------ 北京邮电大学网络技术研究院网络与交换技术国家重点实验室田军 +86 18810315790 mozillafire(a)bupt.edu.cn ------------------------------------

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Re-invite after RTPengine pacemaker switching?
by Жан Базаров 07 Nov '19

07 Nov '19

I need to send re-invite after pacemaker fails over on new rtpengine server. Because new rtpengine dont participate in DTLS handshake and i hear nothing, but silence. I think, may me its would be work. Do you have any idea on this issue?

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Best practises for websocket authentication
by Grant Bagdasarian 12 Jun '19

12 Jun '19

Hello, I was wondering if there are any best practices for authenticating websocket connection establishment inside the xhttp event route? Thanks and regards, Grant

3 3

forcing socket doesn't work for ACK
by David Villasmil 03 May '19

03 May '19

Hello guys, I have public and private IPs, and i need to force the sending socket replying to internal out externals endpoint, so i'm using this: function to check the destination (i tried using $sndto(ip) but it always returns NULL): route[CHECK_DEST_NET] { if ( $du=~"sip:172\..*" ) { setflag(FLAG_TO_PRIVATE); xlog("L_ERR", "[CHECK_SOURCE] Packet going to PRIVATE -> [$du]\n" ); } else { setflag(FLAG_TO_PUBLIC); xlog("L_ERR", "[CHECK_SOURCE] Packet going to PUBLIC -> [$du]\n" ); } return; } on my relay: route(CHECK_DEST_NET); if (isflagset(FLAG_TO_PRIVATE)) { xlog("L_ERR", "[RELAY] forcing socket to PRIVATE NET\n" ); force_send_socket(LISTEN_INSIDE_IF:LISTEN_INSIDE_PORT); } else { xlog("L_ERR", "[RELAY] forcing socket to PUBLIC NET\n" ); force_send_socket(LISTEN_OUTSIDE_IF:LISTEN_OUTSIDE_PORT); } This works nicely, except for ACKs and potentially other packets sent statelessly? Anyone knows how to fix this? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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CSeq not incrementing - app_ruby
by Andrew White 01 May '19

01 May '19

Hi all, I’m still building with app_ruby and loving it - for the most part! I’m having an issue with one of my providers responding with a 482 on auth invite. Researching, it appears that my CSeq isn’t incrementing and this is the reason for the issue. I already have the relevant flag set in my kamailio.cfg however: modparam("dialog", "track_cseq_updates", 1) During my uac_auth, I’ve tried manually setting the diff, as well as running msg_iflag_reset (as suggested by Daniel - https://github.com/kamailio/kamailio/issues/1359 <https://github.com/kamailio/kamailio/issues/1359> - unfortunately the function isn’t exported in KEMI): if KSR::UAC.uac_auth() then KSR.info("UAC authed, relaying") #KSR::COREX.msg_iflag_reset("UAC_AUTH") KSR::PV.sets("$dlg_var(cseq_diff)", "1") KSR.info("CSeq diff: #{KSR::PV.gete("$dlg_var(cseq_diff)")}") KSR::TM.t_relay() Unfortunately in both cases the actual CSeq doesn’t increment in the second INVITE: INVITE 1 (unauthenticated): > CSeq: 102 INVITE INVITE 2 (with Authorization header): > CSeq: 102 INVITE I’m unsure if I’ve missed something silly around the track_cseq_updates flag, or if app_ruby somehow isn’t interacting with the dialog module when running uac_auth to trigger the increment? Thanks! Kind regards, Andrew

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listening at link local IPv6 address
by Juha Heinanen 01 May '19

01 May '19

My K host is behind a mobile WiFi hotspot and I tried to make K listen at the IPv6 address fe80::6e29:95ff:fe7d:37e6/64 that the host has received from the hotspot. Using that address on listen line gives error: Apr 21 08:55:18 salmon /usr/bin/sip-proxy[6887]: ERROR: <core> [core/udp_server.c:395]: udp_init(): bind(8, 0x7ff97dfb3fe4, 28) on FE80:0:0:0:6E29:95FF:FE7D:37E6: Invalid argument Apr 21 08:55:18 salmon /usr/bin/sip-proxy[6887]: ERROR: <core> [core/udp_server.c:401]: udp_init(): might be caused by using a link local address, try site local or global I can ping that address like this: $ ping6 fe80::6e29:95ff:fe7d:37e6%wlp1s0 PING fe80::6e29:95ff:fe7d:37e6%wlp1s0(fe80::6e29:95ff:fe7d:37e6%wlp1s0) 56 data bytes 64 bytes from fe80::6e29:95ff:fe7d:37e6%wlp1s0: icmp_seq=1 ttl=64 time=0.030 ms I then tried to add %wlp1s0 suffix to the listen address, but got error: Apr 21 09:10:35 salmon sip-proxy[7896]: 0(7948) CRITICAL: <core> [core/cfg.y:3510]: yyerror_at(): parse error in config file /etc/sip-proxy/sip-proxy.cfg, line 408, column 34: ip address, interface name or hostname expected Did I do something wrong? If not, why is it that K cannot listen at a link local address? It is something that Linux networking stack does not support? -- Juha

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Pipelimit with clean_unused
by Cinthia Leung 30 Apr '19

30 Apr '19

Hello, What is your recommendation on using clean_unused with db defined pipes? >>pl_check("systemwide"); If I leave the system idle, they are eventually "cleaned" as expected. In our use case, that means failed calls unfortunately. I am trying to see if I can mix both db/static pipes with dynamic/run-time pipes. Thanks Cindy

1 1

Next hop of reply, in onreply_route
by Alex Balashov 30 Apr '19

30 Apr '19

I am fairly sure it’s been asked a number of times, but is there a way to know the destination, and more especially the transport, of the next hop of a reply message, viewable from onreply_route? For requests, we have $nh and many other PVs which in some way evidence where it’s going. And yes, I’m aware that this can be previewed from the onsend_routes*, but I need to amalgamate that knowledge with some transaction-specific data. Thanks! — Sent from mobile, with due apologies for brevity and errors.

2 2

"ERROR: reply too big" and ctl binrpc size
by Karsten Horsmann 30 Apr '19

30 Apr '19

Hi all, i use kamailio 5.0.7 in production and i noticed that i cant listen the dialogs after a few days of traffic on this server. Then i got "kamcmd dlg.list_ctx " - "ERROR: reply too big". So now my question - i want to debug my "sleeping dialogs" with dlg.list - how big should it change the values of without knowing the size of the current filled up binrpc stuff? Or is there a better option to debug dialog-module? modparam("ctl", "binrpc_max_body_size", ??) modparam("ctl", "binrpc_struct_max_body_size", ?) -- Cheers and see you in Berlin *Karsten Horsmann*

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Re: [SR-Users] SIP basic flow question: R-URI in ACK
by Yufei Tao 30 Apr '19

30 Apr '19

Hi Daniel, Thank you very much for clarifying! Cheers, Yufei On 30 Apr 2019 at 08:10, <Daniel-Constantin Mierla <miconda(a)gmail.com>> wrote: Hello, there is an errata for that RFC, but it refers only to the BYE, where it adds back the transport=tcp, but the ACK R-URI should follow the same rules, see: * https://www.rfc-editor.org/errata/eid5294 Cheers, Daniel On 28.04.19 22:54, Yufei Tao wrote: Hi, I have a question regarding basic SIP flow for establishing a call session, and wonder if anyone can help me clarify. For normal call set up, INVITE-OK-ACK, from RFC3261, I believe the ACK’s R-URI should be a copy of the Contact header of the OK message. However in RFC3665 sections 3.1 and 3.2 for example, the ACK’s R-URI missed the parameters from the Contact header of the OK, e.g. In OK the Contact header is: Contact: <sip:bob@client.biloxi.example.com;transport=tcp> <sip:bob@client.biloxi.example.com;transport=tcp> And ACK the R-URI is: ACK sip:bob@client.biloxi.example.com SIP/2.0 Which has got the parameter ‘transport=tcp’ removed – why is this? In this case if the proxy connected to Bob handles ACK in the usual way as it would for all in-dialog requests, i.e. based on routing headers only and not extra processing, it’ll try to relay the ACK message to Bob using the default transport UDP which is not expected and will fail. Can anyone help explain why the parameters are removed in RFC3665 examples please? Have I missed anything? Thank you very much! Yufei _______________________________________________ Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

1 0

docker image for non x86 arch
by Sergey Safarov 30 Apr '19

30 Apr '19

Is the community involved in kamailio docker image for ARM architecture and others not x86 platforms? ARM ARM 64 IBM POWER IBM Z PowerPC 64 LE

3 3

SIP basic flow question: R-URI in ACK
by Yufei Tao 30 Apr '19

30 Apr '19

Hi, I have a question regarding basic SIP flow for establishing a call session, and wonder if anyone can help me clarify. For normal call set up, INVITE-OK-ACK, from RFC3261, I believe the ACK’s R-URI should be a copy of the Contact header of the OK message. However in RFC3665 sections 3.1 and 3.2 for example, the ACK’s R-URI missed the parameters from the Contact header of the OK, e.g. In OK the Contact header is: Contact: <sip:bob@client.biloxi.example.com;transport=tcp> And ACK the R-URI is: ACK sip:bob@client.biloxi.example.com SIP/2.0 Which has got the parameter ‘transport=tcp’ removed – why is this? In this case if the proxy connected to Bob handles ACK in the usual way as it would for all in-dialog requests, i.e. based on routing headers only and not extra processing, it’ll try to relay the ACK message to Bob using the default transport UDP which is not expected and will fail. Can anyone help explain why the parameters are removed in RFC3665 examples please? Have I missed anything? Thank you very much! Yufei

3 2

remove_record_route - bug?
by Sergiu Pojoga 29 Apr '19

29 Apr '19

Hi ppl, After a branch route fails, I need to re-evaluate the next destination and adjust RR parameters, if need be, like for example adjust the nat=yes|priv. failure_route[MANAGE_PSTN_FAILURE] { ... remove_record_route(); record_route(); route(NATMANAGE); ... route(RELAY); } The result is that the new branch is stripped of "Record-Route: <sip:MYIP>", but the params remained. Moreover, record_route hasn't been re-added as instructed in the failure route. Request headers of the initial branch that will eventually timeout and fail: 2019/04/27 18:29:18.034992 10.22.0.1:5060 -> 10.22.0.100:5060 INVITE sip:1514XXXXXXX@10.22.0.100 SIP/2.0 Record-Route: <sip:10.22.0.1;r2=on;lr=on;did=9e2.3dd2;nat=priv> Record-Route: <sip:65.XX.XX.1;r2=on;lr=on;did=9e2.3dd2;nat=priv> New request branch after failure: 2019/04/27 18:29:19.538292 65.XX.XXX.1:5060 -> 65.XX.XX.2:5060 INVITE sip:1514XXXXXXX@65.XX.XX.2 SIP/2.0 ;lr=on;did=9e2.3dd2;nat=priv> kamailio -v version: kamailio 5.1.6 (x86_64/linux) 7d1964 Much obliged. --Sergiu

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Vide SIP conferencing kamailio with freeswitch
by Gaurav Bmotra 28 Apr '19

28 Apr '19

HI i m gaurav i have installed kamailio 5.1.x on Ubuntu server 18.4 i want a solution for video conferencing with kamailio as i am using freeswitch with it is there reference guide for that plz help -- *Regards:* Gaurav Kumar

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Re: [SR-Users] [sr-dev] Vide SIP conferencing kamailio with freeswitch
by Henning Westerholt 28 Apr '19

28 Apr '19

Hello, please ask this usage related questions on our "sr-users" list. Kamailio is not doing much regarding the video calls. E.g. if your calls work for voice with kamailio, they should also work for video. All the setup needs to be done in freeswitch. I found a (outdated) howto with google: https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video Maybe this helps. Henning Am 27.04.19 um 13:22 schrieb Gaurav Bmotra: HI i m gaurav i have installed kamailio 5.1.x on Ubuntu server 18.4 i want a solution for video conferencing with kamailio as i am using freeswitch with it is there reference guide for that plz help _______________________________________________ Kamailio (SER) - Development Mailing List sr-dev(a)lists.kamailio.org<mailto:sr-dev@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-dev -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

1 0

How to configure ipv6 in kamailio
by Santosh Kumar Dash 28 Apr '19

28 Apr '19

Hi All, I have a kamailio installed in my setup which is working properly in IPV4 configuration. I want to perform IPV6 configuration, I don't have any idea how to configure it, I have tried few things by changing *kamailio.cfg* file but did not work. I am sharing my *kamailio.cfg *file which is working fine in ipv4 Kindly help me to configure for IPV6. lets say my Ipv6 address is : 2402:3a80:d1a:f45a:fe44:2e54:ed2b:9bb2 Thanks,

3 3

Can't set module parameter
by Tmbc 26 Apr '19

26 Apr '19

Hi all Having this issue with kamailio: Can't set module parameter Apr 25 09:57:10 kamailio: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <dst_avp> of type <1> not found in module <dispatcher> Apr 25 09:57:10 kamailio: CRITICAL: <core> [core/cfg.y:3511]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 119, column 50: Can't set module parameter Apr 25 09:57:10 kamailio: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <grp_avp> of type <1> not found in module <dispatcher> Apr 25 09:57:10 kamailio: CRITICAL: <core> [core/cfg.y:3511]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 120, column 50: Can't set module parameter Apr 25 09:57:10 kamailio: ERROR: <core> [core/modparam.c:141]: set_mod_param_regex(): parameter <cnt_avp> of type <1> not found in module <dispatcher> Apr 25 09:57:10 kamailio: CRITICAL: <core> [core/cfg.y:3511]: yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 121, column 50: Can't set module parameter here is full conf file: https://dpaste.de/bP3E OS: NAME="CentOS Linux" VERSION="7 (Core)" Software version: kamailio 5.2.2 (x86_64/linux) 67f967 Thanks in advance, tmbc

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Re: [SR-Users] Config to control TCP connection reuse
by R, Rajkumar (Raj) 24 Apr '19

24 Apr '19

I think elaborating on the setup may help someone to provide inputs on this. Below is the setup. Client 1 --> Kamailio proxy --> Backend SIP server Client 2 --> Kamailio proxy --> Backend SIP server Client 3 --> Kamailio proxy --> Backend SIP server After I register all three clients, see only one TCP connection is being used between "Kamailio proxy - Backend SIP server". Is there a way to achieve separate TCP connection to Backend SIP server from kamailio. thanks, raj From: R, Rajkumar (Raj) Sent: Monday, April 22, 2019 3:45 PM To: sr-users(a)lists.kamailio.org Subject: Config to control TCP connection reuse Hi, In my setup(kamailio in stateless mode to a backend server), for any number of client registrations kamailio (in stateless mode) is reusing the TCP connection. Let's say if I register 3 different clients through kamailio to my backend server - see only one TCP connection is used, which I do not want. I have gone through the core modules to see if any TCP config param exists to control on this, but couldn't find one. Could anyone help how to achieve different TCP connections to same destination in kamailio? thanks, raj

2 1

DMQ_USRLOC
by Gholamreza Sabery 24 Apr '19

24 Apr '19

Hi, Recently, I started to use dmq_usrloc module. I successfully set it up. Unfortunately, the documentation does not specify the behavior of dmq_usrloc module with other Kamailio modules. For example, I see a bunch of KDMQ messages are exchanged between nodes, but how the lookup() function behaves with dmq_usrloc, or what is it's effect on a db_mode of 3! Regards

5 6

JSON error - control character 0xd
by David Dean 23 Apr '19

23 Apr '19

I'm trying to send an SDP from a SIP client to Janus using JSON over HTTP. The problem is that Janus is erroring on the SDP content with the following error: JANUS HTTP: Get SDP for echo plugin: {#012 "janus": "error",#012 "error": {#012 "code": 454,#012 "reason": "JSON error: on line 9: control character 0xd near '\"v=0'"#012 }#012} Result code 200 I tried using {s.escape.common} to escape any quotations, commas etc but it doesn't appear to replace the control characters with newlines. Any suggestions on how best to remove these control characters so that Janus will accept the SDP over JSON? Here is the relevant section of kamailio.cfg: # HTTP: Request SDP for echo test plugin sdp_get("$avp(sdp)"); $var(res) = http_connect("janus", "/janus/$var(JANUS-ID)/$var(ECHO-ID)", "application/json", "{ \"janus\" : \"message\", \"transaction\" : \"testtesttest99\", \"body\" : { \"audio\" : true }, \"jsep\" : { \"type\" : \"offer\", \"sdp\" : \"$(avp(sdp){s.escape.common})\" }}", "$avp(janus-pluginsession)"); xlog("L_INFO", "JANUS HTTP: Get SDP for echo plugin: $avp(janus-pluginsession) Result code $var(res)\n");

3 6

Direction of replies
by Duarte Rocha 23 Apr '19

23 Apr '19

Greetings, Is there a way to know in reply_route if a reply has direction downstream or upstream regarding the first INVITE? I know there are vars and methods for finding this out on requests, but i can't find anything for replies. Thanks in advance, Best Regards, Duarte Rocha

2 1

CDR start time of a canceled call
by Jan-Hendrik Dörner 23 Apr '19

23 Apr '19

Hello, I just noticed that the logged „start_time" as well as „end_time" of a canceled call is set to the „invite time“. If done this way, it is harder to calculate the „ringing time“. Is there an option to change this behaviour (i.e. to log the time of the received „bye“ instead)? (If done so, the ringing time would always be the time between start and invite …) Or is there a special reason for this behaviour? My CDR settings are as followed: modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_expired_dlg_enable", 1) # log unanswered / open dialogs modparam("acc", "cdr_on_failed", 1) # log also not answered calls modparam("acc", "cdr_extra", "inviteTime=$dlg_var(inviteTime)" modparam("acc", "cdr_log_enable", 1) # activate write to syslog modparam("acc", "cdr_facility", "LOG_LOCAL6") # syslog modparam("acc", "cdr_start_on_confirmed", 1) modparam("acc", "time_mode", 0) #syslog timestamp modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) modparam("acc", "log_flag", FLT_ACC) # needed for syslog modparam("acc", "log_missed_flag", FLT_ACCMISSED) # need for logging missed calls in syslog Regards Jan-Hendrik

1 0

Kamailio-5.2.1 - Issues in NDB_MONGODB module initialisation
by José Seabra 22 Apr '19

22 Apr '19

Hello there, I have NDB_MONGODB loaded in kamailio with following configurations: *loadmodule "ndb_mongodb"* *modparam("ndb_mongodb", "server", "name=mgs1;uri='mongodb://mongodbserver:27017/proxyregistrar'")* Kamailio breaks up on start up and prints the following messages: *Apr 18 13:16:10.464 proxy-registrar[-]: DEBUG: <core> [core/modparam.c:83]: set_mod_param_regex(): 'ndb_mongodb' matches module 'ndb_mongodb'* *Apr 18 13:16:10.464 proxy-registrar[-]: DEBUG: <core> [core/sr_module.c:711]: find_param_export(): found <server> in module ndb_mongodb [/usr/lib64/kamailio/modules/ndb_mongodb.so]* *Apr 18 13:16:10.464 proxy-registrar[-]: DEBUG: <core> [core/modparam.c:99]: set_mod_param_regex(): found <server> in module ndb_mongodb [/usr/lib64/kamailio/modules/ndb_mongodb.so]* *Apr 18 13:16:10.464 proxy-registrar[-]: DEBUG: ndb_mongodb [mongodb_client.c:155]: mongodbc_add_server(): added server[mgs1]=mongodb://10.225.121.12:27017/proxyregistrar <http://10.225.121.12:27017/proxyregistrar>* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: DEBUG: <core> [core/sr_module.c:942]: init_mod_child(): idx 21 rank 8: ndb_mongodb [tcp receiver (generic) child=3]* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: CRITICAL: <core> [core/pass_fd.c:277]: receive_fd(): EOF on 23* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: ALERT: <core> [main.c:739]: handle_sigs(): child process 3181 exited by a signal 11* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: ALERT: <core> [main.c:742]: handle_sigs(): core was generated* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: DEBUG: <core> [core/tcp_main.c:3513]: handle_ser_child(): dead child 16, pid 3181 (shutting down?)* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: INFO: <core> [main.c:764]: handle_sigs(): terminating due to SIGCHLD* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: DEBUG: <core> [core/io_wait.h:602]: io_watch_del(): DBG: io_watch_del (0xa6d0e0, 23, -1, 0x0) fd_no=23 called* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: DEBUG: <core> [main.c:766]: handle_sigs(): terminating due to SIGCHLD* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: INFO: <core> [main.c:819]: sig_usr(): signal 15 received* *123: INFO: <core> [main.c:819]: sig_usr(): signal 15 received* *123: INFO: <core> [main.c:819]: sig_usr(): signal 15 received* *123: INFO: <core> [main.c:819]: sig_usr(): signal 15 received* *123: INFO: <core> [main.c:819]: sig_usr(): signal 15 received* *1a7311e30a6793da-3165(a)10.225.121.93 <1a7311e30a6793da-3165(a)10.225.121.93>: DEBUG: ndb_mongodb [ndb_mongodb_mod.c:128]: mod_destroy(): cleaning up * Core dump back trace output: *(gdb) bt full* *#0 0x00007f3daa134f93 in _mongoc_client_new_from_uri () from /lib64/libmongoc-1.0.so.0* *No symbol table info available.* *#1 0x00007f3daa156553 in mongoc_uri_get_ssl () from /lib64/libmongoc-1.0.so.0* *No symbol table info available.* *#2 0x00007f3daa134f88 in _mongoc_client_new_from_uri () from /lib64/libmongoc-1.0.so.0* *No symbol table info available.* *#3 0x00007f3daa135029 in mongoc_client_new () from /lib64/libmongoc-1.0.so.0* *No symbol table info available.* *#4 0x00007f3daae6d75f in mongodbc_init () at mongodb_client.c:63* * rsrv = 0x7f3db0f86728* * __FUNCTION__ = "mongodbc_init"* *#5 0x00007f3daae73c46 in child_init (rank=3) at ndb_mongodb_mod.c:113* * __FUNCTION__ = "child_init"* *#6 0x00000000005470d7 in init_mod_child (m=0x7f3db0f85e88, rank=3) at core/sr_module.c:846* * __FUNCTION__ = "init_mod_child"* *#7 0x00000000005474b6 in init_child (rank=3) at core/sr_module.c:874* * ret = 32573* *#8 0x00000000004b3840 in fork_process (child_id=3, desc=0x7ffecc20c7f0 "udp receiver child=2 sock=10.225.121.93:5060 <http://10.225.121.93:5060>", make_sock=1) at core/pt.c:341* * pid = 0* * child_process_no = 3* * ret = -1* * new_seed1 = 1273263324* * new_seed2 = 1531871627* * sockfd = {14, 15}* * __FUNCTION__ = "fork_process"* *#9 0x0000000000424b9b in main_loop () at main.c:1631* * i = 2* * pid = 3807* * si = 0x7f3db123a988* * si_desc = "udp receiver child=2 sock=10.225.121.93:5060 <http://10.225.121.93:5060>\000\177\000\000P\310 \314\376\177\000\000\000e|\261=\177\000\000 \313 \314\376\177\000\000\063\225{\000\000\000\000\000\027\000\000\000\000\000\000\000\000\000\000 \000\000\000\000\000\000\200\000\000\000\000\000\217e|\261=\177\000\000`~x\000\000\000\000\000\230\304\022\261=\177\000"* * nrprocs = 4* * woneinit = 1* * __FUNCTION__ = "main_loop"* *#10 0x000000000042c4e9 in main (argc=11, argv=0x7ffecc20cc08) at main.c:2675* * cfg_stream = 0x2893010* * c = -1* * r = 0* * tmp = 0x7ffecc20e6fd ""* * tmp_len = 0* * port = 0* * proto = 2496* * options = 0x7697a0 ":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 = 173321635* * rfd = 4* * debug_save = 0* * debug_flag = 0* * dont_fork_cnt = 0* * n_lst = 0x7ffecc20cac0* * p = 0x7f3db177497d <intel_check_word+173> "H\205\300t\025\017\266P\003\017\266\312D9\371\017\204", <incomplete sequence \316>* * st = {st_dev = 19, st_ino = 21919, st_nlink = 2, st_mode = 16832, st_uid = 993, st_gid = 2, __pad0 = 0, st_rdev = 0, st_size = 40, st_blksize = 4096, st_blocks = 0, st_atim = {tv_sec = 1551179759, tv_nsec = 561093463}, st_mtim = {tv_sec = 1551179759,* *---Type <return> to continue, or q <return> to quit---* * tv_nsec = 561093463}, st_ctim = {tv_sec = 1551179759, tv_nsec = 561093463}, __unused = {0, 0, 0}}* * __FUNCTION__ = "main"* *(gdb)* NDB_MONGODB configuration is pretty simples and currently i'm not making any mongodb operations, i'm just loading the module and trying to connect against mongodb. Any thoughts on this issue? Thank you Best regards -- Cumprimentos José Seabra

2 1

rtpengine_offer() with ICE not updating o= and m= IP addresses
by David Dean 22 Apr '19

22 Apr '19

I'm using the following rtpengine_offer() to force the use of ICE relay and also replace o= and m= rtpengine_offer("replace-origin replace-session-connection ICE=force-relay RTP"); The SDP is being updated to include an ICE relay candidate, but the IP addresses in the o= and m= lines are not changing to the servers IP address (X.X.X.X). I also tried using the media-address parameter but that didn't change the behaviour. When I use rtpengine_answer() then it's not providing an ICE candidate nor changing o= or m=, which I imagine is expected. What am I missing? INVITE from the client (before rtpengine)------------------------------------------------------ INVITE sip:server.domain.com SIP/2.0Via: SIP/2.0/TCP 154.20.1.8:37520;branch=z9hG4bK-1710993570;rportFrom: <sip:username@server.domain.com>;tag=281751229To: <sip:server.domain.com>Contact: <sip:username@154.20.1.8:37520;transport=tcp>Call-ID: c715a8f9-4dd1-38d8-acbd-54153310f59fCSeq: 1756522847 INVITEContent-Type: multipart/mixed;boundary=7efec5a8-3311-0157-fc8b-9af106265507Content-Length: 1506Max-Forwards: 70Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFERPrivacy: noneSupported: 100rel --7efec5a8-3311-0157-fc8b-9af106265507Content-Type: application/sdp v=0o=organization 1983 678901 IN IP4 154.20.1.8s=-c=IN IP4 154.20.1.8t=0 0a=tcap:1 RTP/AVPFm=audio 15418 RTP/AVP 114 115 0 101i=speecha=ptime:20a=minptime:1a=maxptime:255a=silenceSupp:off - - - -a=rtpmap:114 AMR-WB/16000/1a=imageattr:114 octet-align=0a=fmtp:114 octet-align=0a=rtpmap:115 AMR-WB/16000/1a=imageattr:115 octet-align=1a=fmtp:115 octet-align=1a=rtpmap:0 PCMU/8000/1a=rtpmap:101 telephone-event/8000/1a=fmtp:101 0-16a=pcfg:1 t=1a=sendrecva=rtcp-muxa=ssrc:2047236273 cname:e6dcab7948b70d52cd51b75e505c49eba=ssrc:2047236273 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2a=ssrc:2047236273 label:organization@audio INVITE from the server (after rtpengine)------------------------------------------------------ SIP/2.0 200 OKVia: SIP/2.0/TCP 154.20.1.8:37520;branch=z9hG4bK-1710993570;rport=37520;received=2.221.73.230From: <sip:username@server.domain.com>;tag=281751229To: <sip:server.domain.com>;tag=0ea7681ffd64b0ffcde1bff3393fd505.ae3aCall-ID: c715a8f9-4dd1-38d8-acbd-54153310f59fCSeq: 1756522847 INVITEContact: <sip:X.X.X.X:443;transport=tcp>Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFERPrivacy: noneContent-Type: multipart/mixed;boundary=7efec5a8-3311-0157-fc8b-9af106265507Server: kamailio (5.2.2 (x86_64/linux))Content-Length: 1965 --7efec5a8-3311-0157-fc8b-9af106265507Content-Type: application/sdp v=0o=organization 1983 678901 IN IP4 154.20.1.8s=-c=IN IP4 154.20.1.8t=0 0a=tcap:1 RTP/AVPFm=audio 15418 RTP/AVP 114 115 0 101i=speecha=minptime:1a=maxptime:255a=silenceSupp:off - - - -a=imageattr:114 octet-align=0a=imageattr:115 octet-align=1a=pcfg:1 t=1a=rtcp-muxa=ssrc:2047236273 cname:e6dcab7948b70d52cd51b75e505c49eba=ssrc:2047236273 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2a=ssrc:2047236273 label:organization@audioa=rtpmap:114 AMR-WB/16000/1a=rtpmap:115 AMR-WB/16000/1a=rtpmap:0 PCMU/8000/1a=rtpmap:101 telephone-event/8000/1a=fmtp:114 octet-align=0a=fmtp:115 octet-align=1a=fmtp:101 0-16a=sendrecva=ptime:20a=ice-ufrag:kzgbFAnua=ice-pwd:hnEu9EwTrdIZnTWhYbxtmdugvra=candidate:dzkwmPcB28NPHfRA 1 UDP 16777215 X.X.X.X 30366 typ relay raddr X.X.X.X rport 30366a=candidate:dzkwmPcB28NPHfRA 2 UDP 16777214 X.X.X.X 30367 typ relay raddr X.X.X.X rport 30367a=sendrecva=ice-ufrag:uDE9wbvYa=ice-pwd:dx4CcIjqUNFfnRPJEF8GjatXdU

3 3

Config to control TCP connection reuse
by R, Rajkumar (Raj) 22 Apr '19

22 Apr '19

Hi, In my setup(kamailio in stateless mode to a backend server), for any number of client registrations kamailio (in stateless mode) is reusing the TCP connection. Let's say if I register 3 different clients through kamailio to my backend server - see only one TCP connection is used, which I do not want. I have gone through the core modules to see if any TCP config param exists to control on this, but couldn't find one. Could anyone help how to achieve different TCP connections to same destination in kamailio? thanks, raj

1 0

Best tutorial for beginners kamailio
by Gaurav Bmotra 22 Apr '19

22 Apr '19

Hi Plz sujjest best kamailio tutorial or any source where a beginner can start kamailio.. Thanks

2 1

push to talk over cellular application server and related media server implementation
by Thierry Luo 21 Apr '19

21 Apr '19

Hi All, I don't know if this is the appropriate place to ask the following questions: 1. Anyone please recommend open-source implementation of PoC (Push-to-talk Over Cellular) application server? Kamailio can do this? 2. Anyone please recommend open-source implementation of PoC related media server/replicator? Thierry Luo

3 2

Re: [SR-Users] authentication using auth_db
by arish haque 21 Apr '19

21 Apr '19

Hi David, This can solve the issue but depth(records) of that view would be too much. The architecture is like this: One kamailio fronting multiple asterisk. Each of asterisk boxes have their Specific users databases. Thus, I'm trying to create db connections inside request route or completely write my own authentication service if I get the user's Password. So, that would be great if I would be able to pass the db_url to auth_db module in request route. Let me know what you think. Thanks, Arish

2 1

Re: [SR-Users] authentication using auth_db
by arish haque 20 Apr '19

20 Apr '19

Hi David, Thanks for the reply. I checked out sqlops module, this will solve the problem only when I get the authentication password (secret) of the user inside request route. For this I tried checking kamailio pseudovars but haven't found one. Therefore, I am sticking with auth_db module.If this is possible by any other way do let me know. thanks, Arish

2 1

Kamailio, TLS, and LetsEncrypt
by Gaurav Bmotra 19 Apr '19

19 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i am trying to configure LetsEncrypt (tls) for kamailio sing following link https://www.fredposner.com/1836/kamailio-tls-and-letsencrypt/ but unable to start kamailio ,i m getting these error ------->>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> Apr 19 11:23:57 ubuntufresh systemd[1]: Starting Kamailio (OpenSER) - the Open Source SIP Server... Apr 19 11:23:58 ubuntufresh kamailio[5365]: INFO: <core> [core/sctp_core.c:74]: sctp_core_check_support(): SCTP API not enabled - if you want to use it, load sctp module Apr 19 11:23:58 ubuntufresh kamailio[5365]: INFO: <core> [core/tcp_main.c:4721]: init_tcp(): using epoll_lt as the io watch method (auto detected) Apr 19 11:23:58 ubuntufresh kamailio[5365]: Listening on Apr 19 11:23:58 ubuntufresh kamailio[5365]: Aliases: Apr 19 11:23:58 ubuntufresh kernel: kamailio[5378]: segfault at 0 ip 00007f8d6fdac646 sp 00007ffe8e2337f8 error 4 in libc-2.27.so [7f8d6fcfb000+1e7000] Apr 19 11:23:58 ubuntufresh kamailio[5365]: ERROR: <core> [core/daemonize.c:303]: daemonize(): Main process exited before writing to pipe Apr 19 11:23:58 ubuntufresh systemd[1]: kamailio.service: Control process exited, code=exited status=255 Apr 19 11:23:58 ubuntufresh systemd[1]: kamailio.service: Failed with result 'exit-code'. Apr 19 11:23:58 ubuntufresh systemd[1]: Failed to start Kamailio (OpenSER) - the Open Source SIP Server. <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<------------------------------------ plz help -- *Regards:* Gaurav Kumar

3 5

auth by IP
by Yu Boot 19 Apr '19

19 Apr '19

Following code snippet from default kamailio.cfg never gives 403 if you smart enough to set "fromdomain" parameter on Asterisk to Kamailio's IP. How to fix it? I want password-based registration (which is OK now) and permit calls via Kamailio only from permitted IPs. # 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; }

2 4

TLS implementation for kamailio
by Gaurav Bmotra 19 Apr '19

19 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i want to implement TLs for kamailio plz help -- *Regards:* Gaurav Kumar

3 2

Microsoft Direct Routing Assistance And Kamailio Integration
by Phillip Cooper 18 Apr '19

18 Apr '19

Hi All, I am looking for some assistance on setting up Microsoft Direct Routing with Kamailio as an SBC. I have a partial config working, I have rewrote the headers going to Direct routing and can place calls in to and out of Direct routing. I have problems with Call hang ups and choppy audio only on the audio stream from Direct Routing. 1) place a call in to direct routing, Hang the call up from the caller and Direct routing hangs up the call. But If you place a call in to Direct routing and hang up the call from direct routing the direct routing hangs up but the call is still active on the caller. IF you call from direct routing through kamailio to a callee and hang up from direct routing the call is hung up correctly. 2) on some calls made to or from direct routing, the audio stream coming from direct routing is sometimes choppy. Its always the audio from direct routing never the audio going to direct routing. My set up: Asterisk <-> Kamailio <-> Direct Routing SIP: Asterisk <-> Kamailio <-> direct routing RTP: Asterisk <-> Direct Routing Kamailio is rewriting the R-URI, TO, FROM, Contact headers going to Direct routing, as Microsoft need the sbc hostname to be in the contact header in a particular format. I am looking for help on the above or if someone would be kind enough to provide a working config that would be awesome! :) Many Thanks in advance for assistance. :) Phil --- This email has been checked for viruses by AVG. https://www.avg.com

1 0

authentication using auth_db module
by arish haque 18 Apr '19

18 Apr '19

Hi all, I'm trying to establish mysql database connection inside request route (at runtime) based on the type of user, as credentials are stored in multiple databases (may be different servers) for the users. Therefore, I am trying to find out that is there anyway to connect to mysql dynamically in request route or pass 'db_url' parameter dynamically to auth_db module. Hope this explains the requirement. Looking for suggestions. With Regards, Arish

2 1

How to append a body to send_reply() ?
by David Dean 18 Apr '19

18 Apr '19

Hello Kamailio gurus, I'm successfully using send_reply() with append_to_reply() to add some custom headers. Now I need to add some text into the reply body. How can I add a body to send_reply() ? Thx, Dave

2 7

After-parsing config file?
by David Villasmil Govea 18 Apr '19

18 Apr '19

Hello all, Kind of a weird question: is it possible to get a config file AFTER it’s been parsed? I.e: after it’s skipped all non-defined blocks, after substdef have been applied, etc. In the form of a normal config file Is this possible? -- DVG -- Imagination is more important than knowledge Albert Einstein

1 0

Use domain instead of IP in kamailio for register user
by Gaurav Bmotra 18 Apr '19

18 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i wan to to use domain instead of IP for registration in kamailio Plz help -- *Regards:* Gaurav Kumar

3 3

Does Kamailio provide any API for other program to create sip account
by Gaurav Bmotra 18 Apr '19

18 Apr '19

Hi i am Gaurav I am developing a sip VoIP app in Android, which want to provide the feature of allowing user to create sip account from client side. I use Kamailio as sip server. My thought is when user register to the the user account system, the user account system can communicate with Kamailio, create a sip account associated with the user and pass the sip account to the client, the client then can configure itself using the sip account. Does Kamailio provide any API or module for me to do this? Thanks! -- *Regards:* Gaurav Kumar

2 1

re-connect call in case of network issue
by Gaurav Bmotra 18 Apr '19

18 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 how can we implement re-connecting call instead of disconnect in case of network problem plz help thanks -- *Regards:* Gaurav Kumar

1 0

VRF routing with kamailio 5.2
by Jonathan Hunter 18 Apr '19

18 Apr '19

Hi All, Hope everyone in community is well. I was wondering if anyone could let me know if VRF is supported with kamailio 5.2 ? Essentially I am looking to put kamailio between two SBCs and bridge the networks together. So currently Server A 1.1.1.1 connects to Server B 2.2.2.2 and they pass calls fine, and I want to add kamailio in between without changing any IPs, so using VRF I have one network interface intA with 1.1.1.1 and the other intB with 2.2.2.2. What I want to do is listen on both interfaces, which is fine but bridge or force traffic between the two. Is this possible with kamailio? As doing force socket and listening on both 1.1.1.1 and 2.2.2.2 kamailio just loops and routes to itself, and force socket doesnt help. Do you have any suggestions? Would running two instances of kamailio on same server help or even introducing to separate kamailio instances and force routing between the two for requests. I am currently just testing with SIP OPTIONS and it is already proving tricky. Any help or comments appreciated. I appreciate this might not be the best solution however we have a lead time with providers to make changes which are proving quite slow and impractical so just trying to work a solution. Thanks Jon

2 1

integration kamailio with Freeswitch for video conferencing
by Gaurav Bmotra 18 Apr '19

18 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 and freeswitch 1.8 on debian 9 i want to configure kamailio with freeswitch for video conformance plz provide any reference guide plz help Thanks -- *Regards:* Gaurav Kumar

1 0

video chat with kamailio
by Gaurav Bmotra 18 Apr '19

18 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i m working on video chat app so how can i use kamailio for video chat OR i need to use other integration like freeSwitch with kamailio ? Plz help -- *Regards:* Gaurav Kumar

2 1

Support for TLS v1.3 (RFC 8446)
by g.Helio 18 Apr '19

18 Apr '19

2 1

Via shows wrong protocol of Kamailio
by Andrew Chen 18 Apr '19

18 Apr '19

Hi all, Got another anomaly in 4.1.5. I have this block of code which does t_relay_tcp: if ($var(isTpgw) == 1) { xlog("L_INFO","[$ci][$mi][CSeq $cs] isTpgw = $var(isTpgw)"); if (!t_relay_to_tcp()) { xlog("L_INFO","[$ci][$mi][CSeq $cs] $ua Relaying [$rm] from ($si) to ($rd) via tcp....Error...\n"); t_on_failure("DISPATCHER_FAIL"); } else { xlog("L_INFO","[$ci][$mi][CSeq $cs] $ua Relaying [$rm] from ($si) to ($rd) via tcp\n"); } } But when I looked at my sip capture tool, it showed the Via marked by the Kam to use UDP: INVITE sip:######@<some domain>:5060 SIP/2.0 Record-Route: <sip:206.81.191.30;r2=on;lr=on> Record-Route: <sip:206.81.191.30;transport=tcp;r2=on;lr=on> Call-ID: DL9dc0c8dd93-1079779342(a)gs-3924.GSLAB.COM To: "sip:6204857@sjomaintpsg50.fuzemeeting.com" <sip:6204857@sjomaintpsg50.fuzemeeting.com> From: "LifesizeSIP" <sip:LifesizeSIP@gs-3924.GSLAB.COM:5060>;tag=DL325a9e10c7;epid=062EE468 CSeq: 1 INVITE Max-Forwards: 69 Via: SIP/2.0/UDP 206.81.191.30;branch=z9hG4bKb16b.bd6a0791c798e3c859f8608993d584ae.1;i=5611 Via: SIP/2.0/TCP 10.101.199.105:5060;rport=53272;branch=z9hG4bK-fc8090ecf8-DL 206.81.191.30 is the Kam. I also log the activity as seen in the code with those xlog: Apr 11 19:38:37 sjomaintpsg50 /usr/local/sbin/kamailio[16348]: INFO: <script>: [DL9dc0c8dd93-1079779342(a)gs-3924.GSLAB.COM][42][CSeq 1] isTpgw = 1 Apr 11 19:38:37 sjomaintpsg50 /usr/local/sbin/kamailio[16348]: INFO: <script>: [DL9dc0c8dd93-1079779342(a)gs-3924.GSLAB.COM][42][CSeq 1] LifeSize Softphone 8.1.12 (Windows) Relaying [INVITE] from (10.101.19 9.105) to (206.81.191.72) via tcp So why did the Kamailio mark the Via header on the outgoing INVITE UDP? -- 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.*

2 1

"481 Dialog/Transaction Does Not Exist" after sending SUBSCRIBE with Expires: 0
by David Dean 18 Apr '19

18 Apr '19

When a client sends a SUBSCRIBE with Expires: 0, Kamailio responds correctly with a "202 OK". But then Kamailio sends a NOTIFY to the same client with "Subscription-State: terminated;reason=timeout". The client rejects this NOTIFY with a "481 Dialog/Transaction Does Not Exist". Why does Kamailio send this NOTIFY, and is there any way to fix or disable it because it's always rejected by the client? The client is IMSDroid, and Kamailio is 5.2.2 with a default configuration. Packet 1 = Client sending SUBSCRIBE with Expires: 0 SUBSCRIBE sip:X.X.X.X:443;transport=tcp SIP/2.0Via: SIP/2.0/TCP 192.168.0.21:45336;branch=z9hG4bK-670576127From: <sip:username@domain.com>;tag=1372208921To: <sip:kamailio.domain.com>;tag=c4eb72dabf44e7588b972bd3b454ea11-0cd6Contact: <sip:username@192.168.0.21:45336;transport=tcp>Call-ID: e8cffa8f-948d-eb99-69ca-c9f18bdf11a0CSeq: 1321654295 SUBSCRIBEExpires: 0Content-Length: 299Max-Forwards: 70Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFERAccept: application/pidf+xmlEvent: presence Packet 2 = Kamailio responding with 202 OK SIP/2.0 202 OKVia: SIP/2.0/TCP 192.168.0.21:45336;branch=z9hG4bK-670576127From: <sip:username@domain.com>;tag=1372208921To: <sip:kamailio@domain.com>;tag=c4eb72dabf44e7588b972bd3b454ea11-0cd6Call-ID: e8cffa8f-948d-eb99-69ca-c9f18bdf11a0CSeq: 1321654295 SUBSCRIBEExpires: 0Contact: <sip:X.X.X.X:443;transport=tcp>Server: kamailio (5.2.2 (x86_64/linux))Content-Length: 0 Packet 3 = Kamailio sending a NOTIFY NOTIFY sip:username@192.168.0.21:45336;transport=tcp SIP/2.0Via: SIP/2.0/TCP X.X.X.X:443;branch=z9hG4bKefe7.5f8ec435000000000000000000000000.0To: <sip:username@domain.com>;tag=1372208921From: <sip:kamailio@domain.com>;tag=c4eb72dabf44e7588b972bd3b454ea11-0cd6CSeq: 3 NOTIFYCall-ID: e8cffa8f-948d-eb99-69ca-c9f18bdf11a0Content-Length: 0Max-Forwards: 70Event: presenceContact: <sip:X.X.X.X:443;transport=tcp>Subscription-State: terminated;reason=timeout Packet 4 = Client responding to the NOTIFY with "481 Dialog/Transaction Does Not Exist" SIP/2.0 481 Dialog/Transaction Does Not ExistVia: SIP/2.0/TCP X.X.X.X:443;branch=z9hG4bKefe7.5f8ec435000000000000000000000000.0From: <sip:kamailio@domain.com>;tag=c4eb72dabf44e7588b972bd3b454ea11-0cd6To: <sip:username@domain.com>;tag=1372208921Call-ID: e8cffa8f-948d-eb99-69ca-c9f18bdf11a0CSeq: 3 NOTIFYContent-Length: 0 There are two entries in the Kamailio log: INFO: {1 1321654295 SUBSCRIBE e8cffa8f-948d-eb99-69ca-c9f18bdf11a0} presence [notify.c:1614]: send_notify_request(): NOTIFY sip:username@domain.com via sip:Y.Y.Y.Y:45336;transport=tcp on behalf of sip:kamailio.domain.com for event presence : e8cffa8f-948d-eb99-69ca-c9f18bdf11a0 ERROR: {2 3 NOTIFY e8cffa8f-948d-eb99-69ca-c9f18bdf11a0} presence [subscribe.c:497]: delete_subs(): Failed to delete subscription from memory [slot: 174 ev: sip:kamailio.domain.com pu: presence ci: e8cffa8f-948d-eb99-69ca-c9f18bdf11a0 ft: 1372208921 tt: c4eb72dabf44e7588b972bd3b454ea11-0cd6] These are the presence settings in the Kamailio config file: #!ifdef WITH_PRESENCE# ----- presence params -----modparam("presence", "db_url", DBURL)modparam("presence", "send_fast_notify", 0) # ----- presence_xml params -----modparam("presence_xml", "db_url", DBURL)modparam("presence_xml", "force_active", 1)modparam("presence_xml", "force_dummy_presence", 0)#!endif Appreciate your help.

2 1

Rewrite private IP in SDP
by Bonjour Madame 18 Apr '19

18 Apr '19

Hello all, I am dealing with a no audio issue, phones are not even ringing. There is a private IP address in the "VIA" section of the message header of the INVITE: Via: SIP/2.0/UDP 192.168.1.25:65291 ;rport;branch=z9hG4bKPj9e3849d1981a4094a8323dad5826ec34 and in the contact too: Contact: <sip:5002@192.168.1.25:65291;ob> In the "status 100 Attempting to connect your call" from the server, I can see the public IP of the phone under received: Via: SIP/2.0/UDP 192.168.1.25:65291 ;rport=53990;branch=z9hG4bKPj9e3849d1981a4094a8323dad5826ec34;received=PHONE_PUBLIC_IP And in the "status 200 ok" from the server, I have: Via: SIP/2.0/UDP 192.168.1.25:65291 ;received=PHONE_PUBLIC_IP;rport=53990;branch=z9hG4bKPj9e3849d1981a4094a8323dad5826ec34 Record-Route: <sip:SERVER_PUBLIC_IP;lr=on;ftag=9284528256a54f8b81e692133a15684c> I don't know what steps to take to resolve this issue.

3 3

Record route dynamic IP
by eyas barhouk 17 Apr '19

17 Apr '19

hello dears i'm trying to build kamailio & asterisk with the following : * Kamailio will listen to All the interfaces : listen=udp:0.0.0.0:5050 listen=udp:0.0.0.0:5060 * it will add the record header as in the following part : if (!isflagset(FLT_RR)) record_route(); setflag(FLT_RR); when the sip message get from kamailio to asterisk the record route header will be like "<sip:0.0.0.0:5060;....etc.>" So is there any way to make the Via and record route headers has a specific IP instead of (0.0.0.0) i mean Something like "record_route("$Ri")". thanks & regards

2 1

How can i change kamailio SIP port form 5060 to any other
by Gaurav Bmotra 17 Apr '19

17 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i want to change kamailio SIP port 5060 to other port for security purpose plz help me -- *Regards:* Gaurav Kumar

2 3

TCP connection lifetime
by Aymeric Moizard 17 Apr '19

17 Apr '19

Hi Again, Here is an issue with TCP connection being kept for more: Yesterday, I have discovered that a User-Agent (<Avaya IP Phone 1120E (SIP1120e.04.04.30.00)> tried to register a lot. It was sending REGISTER over new established TCP socket *every 2 seconds*. All the REGISTER was rejected with 401. (may be the device was misconfigured? or not receiving any of my answer? I can't tell) NOTE: You can see the expires header was very large: 86400, ie: 24 hours... I was checking the TCP/TLS connections on my server and discovered more than 1000 TCP established connection to that user/ip, and thus, I have tried to understand what happened. Checking the logs, I received 4855 REGISTER from this device from "Mar 25 03:47:09" to "Mar 25 07:56:13" which is a rate of approx one new TCP connection every 2.5 seconds... Today, I decided to check it again around 11am. jack@sip:~$ sudo kamctl stats tcp { "jsonrpc": "2.0", "result": [ "tcp:con_reset = 1857", "tcp:con_timeout = 35927", "tcp:connect_failed = 25", "tcp:connect_success = 2", "tcp:current_opened_connections = 2291", "tcp:current_write_queue_size = 0", "tcp:established = 80778", "tcp:local_reject = 0", "tcp:passive_open = 80776", "tcp:send_timeout = 2", "tcp:sendq_full = 0" ], "id": 7305 } There was still A LOT of established connections. And the connections have been established more than 24 hours ago. At 11H16: $> lsof -n -l | grep kamailio | grep TCP | grep 41.234.242.69 | grep ESTA | wc -l 1161 At 11H22: $> lsof -n -l | grep kamailio | grep TCP | grep 41.234.242.69 | grep ESTA | wc -l 1018 At 11H35: $> lsof -n -l | grep kamailio | grep TCP | grep 41.234.242.69 | grep ESTA | wc -l 655 At 13H $> lsof -n -l | grep kamailio | grep TCP | grep 41.234.242.69 | grep ESTA | wc -l 0 So the established connections are all gone now. Between 11h16 and 11H35, I was seeing the server regularly sending [FIN, ACK] over each TCP established connection, with retransmissions for all of them. (no incoming trafic) I do not have numbers/capture/stats, but I think that kamailio was already closing some connection yesterday. I don't know when kamailio started to try closing those connections. I'm now back with this status: At 13pm: jack@sip:~$ sudo kamctl stats tcp { "jsonrpc": "2.0", "result": [ "tcp:con_reset = 1896", "tcp:con_timeout = 38042", "tcp:connect_failed = 26", "tcp:connect_success = 2", "tcp:current_opened_connections = 939", "tcp:current_write_queue_size = 0", "tcp:established = 81950", "tcp:local_reject = 0", "tcp:passive_open = 81948", "tcp:send_timeout = 2", "tcp:sendq_full = 0" ], "id": 12734 } With around 155 registration entries using TCP and TLS in my location database. As you can see, tcp:current_opened_connections = 939 is still pretty high compared to my currently registred users. I have "modparam("registrar", "max_expires", 86400)", because I'm keeping contact entries (even with TCP connection down) for push notifications. I have "tcp_connection_lifetime=3600" configured. Question 1 With "tcp_connection_lifetime=3600", I would expect kamailio to close the established connection after 3600 seconds without traffic. It is pretty obvious that no data has been exchanged over the 4855 established connection during a day. Despite the issue with the Avaya phones is solved automatically after a day, I guess similar stuff or happening, at a different rate, for other users as well. (because current_opened_connections is way higher than registred TCP/TLS users) Question 2 I can list TLS connection with "kamctl rpc tls.list" Can I get a similar list for TCP? (lsof returns a lot of duplicates...) Tks Aymeric -- Antisip - http://www.antisip.com

2 4

Route header
by Duarte Rocha 17 Apr '19

17 Apr '19

Greetings, Is there a difference for Kamailio between these 2 situations. 1 - Method with 2 Route Headers, each one with one address and one uri 2 - Method with 1 Route Header that includes 2 Route URI Best Regards, Duarte Rocha

1 0

allow calls only from registered users
by Gaurav Bmotra 17 Apr '19

17 Apr '19

HI this is Gaurav im using kamailio 5.1.2 on ubuntu 18.4 i wan to implement user based calling on kamailio , that only can call how registered with user plz help -- *Regards:* Gaurav Kumar

2 1

Send 200 OK response with body to INVITE requests
by David Dean 16 Apr '19

16 Apr '19

Hi all, I'm building an automated load testing system for an rtp media server with TURN that uses Kamailio for SIP. When a client sends an INVITE to Kamailio, I need Kamailio to respond with "200 OK" and the SDP generated by rtpengine. I've enabled auto-answer in the client, so as soon as it receives the response from Kamailio it will start sending RTP packets. Right now I'm using send_reply("200", "OK"); which is sending the right 200 response back to the client, but it doesn't include any body. How can I get Kamailio to send a "200 OK" and add the rtpengine_answer into the body? Appreciate any help, I've been struggling with this for three days. Cheers. rtpengine_offer("SIP-source-address ICE=force-relay RTP");send_reply("200", "OK");

4 7

Getting a new RTP Port on reINVITE during answered call
by Sergio Charrua 16 Apr '19

16 Apr '19

Hi all, I've the following infrastructure (fake IPs) 62.128.128.68 (Telco SBC Inbound) | | 10.19.139.66 (Kamailio + RTPEngine) | | 10.19.139.69 (Asterisk 13) | | 62.128.128.68 (Telco SBC Outbound) Test scenario is: - a mobile calls a destination; the call comes in from Telco SBC Inbound, goes through Kamailio+RTPengine and then to Asterisk which redirects the call to Telco SBC Outbound. - Asterisk box is set with canreinvite=no and nat=force_rport,comedia so it stays in the path when a call comes in SBC Inbound, it is correctly redireted to SBC Outbound. The call is answered, and after 5 or so seconds, SBC Inbound sends a new INVITE to force codecs, which Asterisk replies correctly. the problem is that between Kamailio and SBC Inbund, when Asterisk replies to the second invite, the media port changes, resulting in audio loss: during the call, audio from destination to originator works OK but originator to destination is mute. According to the telco's guys,this is due to the fact that Kamailio is changing during call the media port from original media port to a new one, on reply to the second Invite, and this is wrong and reason why sound from the originator to destination is lost (but not from destination to originator) I'm lost too here..... Any clue? Can anyone help? Resuming I need to make sure that, if during a call some INVITES are sent from SBC to Asterisk, that the reply keeps the same IP_ADDR:port from beginning to the end of the call. As it is today, the second invite changes te RTP Media port and no audi is heard.... If needed I can send PCap files. please find below my kamailio.cfg file #!define WITH_NAT #!define WITH_PSTN /* enables Accounting Log functions */ #!define FLT_ACC 1 /* enable Accounting of missed or failed calls */ #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 /* sets the dispatcher list file path */ #!define DS_LIST "/etc/kamailio/dispatcher.list" /* sets the RTP Engine address and port; RTP Engine is a NAT enabled RTP Proxy */ #!define RTP_ENGINE_ADDR "udp:127.0.0.1:60000" // was udp:localhost:60000 /* defines DB connection string */ #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@10.19.139.110/kamailio" #!endif # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif #!define FLT_DISPATCH_SETID 1 #!define FLT_FS 10 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* comment the next line to enable TCP */ disable_tcp=no /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ # alias="mysipserver.com" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ //listen=udp:10.19.139.66:5060 listen=udp:eth1:5060 sip_warning=no; ####### Modules Section ######## #set module path mpath="/usr/lib64/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" /**************/ loadmodule "usrloc.so" /**************/ loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "path.so" loadmodule "dispatcher.so" loadmodule "rtpengine.so" loadmodule "nathelper.so" loadmodule "rtimer.so" loadmodule "sqlops.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "pretty_format", 1) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default we do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") //;calltype=$avp(calltype)") # ----- usrloc params ----- /* enable DB persistency for location entries */ modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) # ----- tm params ----- # ----- the TM module enables stateful processing of SIP requests modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- modparam("dispatcher", "list_file", DS_LIST) modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") modparam("dispatcher", "ds_hash_size", 9) modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("path", "use_received", 1) # ----- rtpproxy params ----- modparam("rtpengine", "rtpengine_sock", RTP_ENGINE_ADDR) modparam("nathelper", "received_avp", "$avp(s:rcv)") /************************/ # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:ping@kamailio.org") # params needed for NAT traversal in other modules modparam("usrloc", "nat_bflag", FLB_NATB) /***********************/ modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") modparam("sqlops", "sqlcon", "ca=>mysql://kamailio:**********@*************/kamailio") ####### Routing Logic ######## # main request routing logic route { xlog("L_INFO","$rm from $si"); xlog("L_DBG","[$fU@$si:$sp]{$rm} {$ru} "); # OPTIONS requests without a username in the Request-URI but one # of our domains or IPs are addressed to the proxy itself and # can be answered statelessly. if (is_method("OPTIONS")) // && $fU=="ping" )// && $si=="62.28.173.10" && $fU=="ping")// && strempty((a)ruri.user) && (uri == myself)) { sl_send_reply("200","OK"); exit; } # per request initial checks route(REQINIT); rtpengine_manage(); # NAT detection route(NATDETECT); # handle requests within SIP dialogs ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()){ route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers xlog("L_INFO","Removing Headers"); remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")){ xlog("L_INFO","Recording Route"); record_route(); if (is_method("INVITE")) { if (has_body("application/sdp")) { if (rtpengine_offer()) t_on_reply("1"); } else { t_on_reply("2"); } } } if (is_method("ACK") && has_body("application/sdp")) rtpengine_answer(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } //route(SIPOUT); xlog("L_INFO","Setting PRESENCE"); # handle presence related requests route(PRESENCE); # handle registrations xlog("L_INFO","Handling REGISTRAR"); route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations route(DISPATCH); xlog("L_INFO","**********END*************"); } onreply_route[1] { if (has_body("application/sdp")) rtpengine_answer(); } onreply_route[2] { if (has_body("application/sdp")) rtpengine_offer(); } # Per SIP request initial checks route[REQINIT] { xlog("L_INFO","REQINIT Starting"); if (!mf_process_maxfwd_header("10")) { xlog("L_INFO","483 - Too Many Hops"); sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Sanity Check -> Malformed SIP message from $si:$sp\n"); exit; } xlog("L_INFO","REQINIT Finishing"); } /***********************/ # Caller NAT detection route[NATDETECT] { xlog("L_INFO","Entering NATDTECT"); #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { xlog("L_INFO","Fix Nated Register"); fix_nated_register(); } else { if(is_first_hop()) { xlog("L_INFO","Set Contact Alias"); set_contact_alias(); } } xlog("L_INFO","Set FLT_NATS" + FLT_NATS); setflag(FLT_NATS); } #!endif xlog("L_INFO","NAT Detect set FLT_NTS = " + FLT_NATS); xlog("L_INFO","Finishing NATDETECT"); return; } /***********************/ # Handle requests within SIP dialogs route[WITHINDLG] { xlog("L_INFO","Entering WITHDLG"); if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy xlog("L_INFO","Going to NATMANAGE"); route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction. Try to route anyway - being optimistic # since it has at least a To Tag route(RELAY); exit; } } sl_send_reply("404","Not here"); xlog("L_INFO","Finishing WITHINDLG"); exit; } # Routing to foreign domains route[SIPOUT] { xlog("L_INFO","Entering SIPOUT"); if (uri==myself) { xlog("L_INFO","URI is MySelf!"); return; } append_hf("P-hint: outbound\r\n"); xlog("L_INFO","Finishing SIPOUT"); route(RELAY); exit; } # Wrapper for relaying requests route[RELAY] { xlog("L_INFO","******** RELAY *******"); # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { xlog("L_INFO","branch_route NOT SET!"); t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) { xlog("L_INFO","onreply_route NOT SET!"); t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if(!t_is_set("failure_route")) { xlog("L_INFO","failure_route NOT SET!"); t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { xlog("L_INFO","t_relay returns FALSE"); sl_reply_error(); } exit; } # URI update for dialog requests route[DLGURI] { xlog("L_INFO","Entering DLGURI"); #!ifdef WITH_NAT if(!isdsturiset()) { xlog("L_INFO","Handle ruri ALIAS"); handle_ruri_alias(); } #!endif return; } # RTPProxy control and singaling updates for NAT traversal route[NATMANAGE] { xlog("L_INFO","Entering NATMANAGE"); #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { xlog("L_INFO","nat=yes --- Setting FLB_NATB"); setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) )) { xlog("L_INFO","NO FLT_NATS/B Set!!! Getting out of NATMANAGE"); return; } xlog("L_INFO","Starting RTPEngine"); rtpengine_manage(); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { xlog("L_INFO","adding nat=yes"); add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) { xlog("L_INFO","Set Contact Alias"); set_contact_alias(); } } } #!endif return; } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; add_path_received(); route(DISPATCH); } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # Dispatch requests route[DISPATCH] { xlog("L_INFO","Entering DISPATCH"); # hash over callerid dispatching on gateways group '1' if(!ds_select_dst("1", "10","4")) { xlog("L_INFO","no destination selected from dispatcher list!"); send_reply("404", "No destination"); exit; } xlog("L_INFO","going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Sample failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } xlog("L_INFO", "Media server $du failed to answer, selecting other one!"); # next DST - only for 500 or local timeout if ( t_check_status("500") || (t_branch_timeout() && !t_branch_replied()) ) { #we mark the destination Inactive and Probing ds_mark_dst("ip"); if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } route[CDRS]{ sql_query("ca","call kamailio_cdrs();","rb"); sql_query("ca","call kamailio_rating('default');","rb"); } Cheers, *Sérgio *

1 0

new RTP Port after INVITE during call
by Sergio Charrua 16 Apr '19

16 Apr '19

Hi all, I've the following infrastructure 62.128.128.68 (Telco SBC Inbound) | | 10.19.139.66 (Kamailio + RTPEngine) | | 10.19.139.69 (Asterisk 13) | | 62.128.128.68 (Telco SBC Outbound) Test scenario is: - a mobile calls a destination; the call comes in from Telco SBC Inbound, goes through Kamailio+RTPengine and then to Asterisk which redirects the call to Telco SBC Outbound. - Asterisk box is set with canreinvite=no and nat=force_rport,comedia so it stays in the path when a call comes in SBC Inbound, it is correctly redireted to SBC Outbound. The call is answered, and after 5 or so seconds, SBC Inbound sends a new INVITE to force codecs, which Asterisk replies correctly. the problem is that between Kamailio and SBC Inbund, when Asterisk replies to the second invite, the media port changes, resulting in audio loss: during the call, audio from destination to originator works OK but originator to destination is mute. According to the telco's guys,this is due to the fact that Kamailio is changing during call the media port from original media port to a new one, on reply to the second Invite, and this is wrong and reason why sound from the originator to destination is lost (but not from destination to originator) I'm lost too here..... Any clue? Can anyone help? Resuming I need to make sure that, if during a call some INVITES are sent from SBC to Asterisk, that the reply keeps the same IP_ADDR:port from beginning to the end of the call. As it is today, the second invite changes te RTP Media port and no audi is heard.... If needed I can send PCap files. please find below my kamailio.cfg file #!define WITH_NAT #!define WITH_PSTN /* enables Accounting Log functions */ #!define FLT_ACC 1 /* enable Accounting of missed or failed calls */ #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 /* sets the dispatcher list file path */ #!define DS_LIST "/etc/kamailio/dispatcher.list" /* sets the RTP Engine address and port; RTP Engine is a NAT enabled RTP Proxy */ #!define RTP_ENGINE_ADDR "udp:127.0.0.1:60000" // was udp:localhost:60000 /* defines DB connection string */ #!ifndef DBURL #!define DBURL "mysql://kamailio:kamailiorw@10.19.139.110/kamailio" #!endif # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 ####### Global Parameters ######### #!ifdef WITH_DEBUG debug=4 log_stderror=yes #!else debug=2 log_stderror=no #!endif #!define FLT_DISPATCH_SETID 1 #!define FLT_FS 10 #!define FLT_NATS 5 #!define FLB_NATB 6 #!define FLB_NATSIPPING 7 memdbg=5 memlog=5 log_facility=LOG_LOCAL0 fork=yes children=4 /* comment the next line to enable TCP */ disable_tcp=no /* uncomment the next line to disable the auto discovery of local aliases based on revers DNS on IPs (default on) */ auto_aliases=no /* add local domain aliases */ # alias="mysipserver.com" port=5060 /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ //listen=udp:10.19.139.66:5060 listen=udp:eth1:5060 sip_warning=no; ####### Modules Section ######## #set module path mpath="/usr/lib64/kamailio/modules/" loadmodule "db_mysql.so" loadmodule "jsonrpcs.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" /**************/ loadmodule "usrloc.so" /**************/ loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "acc.so" loadmodule "path.so" loadmodule "dispatcher.so" loadmodule "rtpengine.so" loadmodule "nathelper.so" loadmodule "rtimer.so" loadmodule "sqlops.so" # ----------------- setting module-specific parameters --------------- # ----- jsonrpcs params ----- modparam("jsonrpcs", "fifo_name", "/var/run/kamailio/kamailio_rpc.fifo") modparam("jsonrpcs", "pretty_format", 1) # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- acc params ----- modparam("acc", "failed_transaction_flag", 3) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;src_ip=$si") # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default we do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra","src_user=$fU;src_domain=$fd;src_ip=$si;dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") //;calltype=$avp(calltype)") # ----- usrloc params ----- /* enable DB persistency for location entries */ modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) # ----- tm params ----- # ----- the TM module enables stateful processing of SIP requests modparam("tm", "fr_timer", 2000) modparam("tm", "fr_inv_timer", 40000) # ----- dispatcher params ----- modparam("dispatcher", "list_file", DS_LIST) modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "sock_avp", "$avp(AVP_SOCK)") modparam("dispatcher", "ds_hash_size", 9) modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("path", "use_received", 1) # ----- rtpproxy params ----- modparam("rtpengine", "rtpengine_sock", RTP_ENGINE_ADDR) modparam("nathelper", "received_avp", "$avp(s:rcv)") /************************/ # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:ping@kamailio.org") # params needed for NAT traversal in other modules modparam("usrloc", "nat_bflag", FLB_NATB) /***********************/ modparam("rtimer", "timer", "name=cdr;interval=300;mode=1;") modparam("rtimer", "exec", "timer=cdr;route=CDRS") modparam("sqlops", "sqlcon", "ca=>mysql://kamailio:**********@*************/kamailio") ####### Routing Logic ######## # main request routing logic route { xlog("L_INFO","$rm from $si"); xlog("L_DBG","[$fU@$si:$sp]{$rm} {$ru} "); # OPTIONS requests without a username in the Request-URI but one # of our domains or IPs are addressed to the proxy itself and # can be answered statelessly. if (is_method("OPTIONS")) // && $fU=="ping" )// && $si=="62.28.173.10" && $fU=="ping")// && strempty((a)ruri.user) && (uri == myself)) { sl_send_reply("200","OK"); exit; } # per request initial checks route(REQINIT); rtpengine_manage(); # NAT detection route(NATDETECT); # handle requests within SIP dialogs ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()){ route(RELAY); } exit; } # handle retransmissions if (!is_method("ACK")) { if(t_precheck_trans()) { t_check_trans(); exit; } t_check_trans(); } route(WITHINDLG); # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers xlog("L_INFO","Removing Headers"); remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")){ xlog("L_INFO","Recording Route"); record_route(); if (is_method("INVITE")) { if (has_body("application/sdp")) { if (rtpengine_offer()) t_on_reply("1"); } else { t_on_reply("2"); } } } if (is_method("ACK") && has_body("application/sdp")) rtpengine_answer(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } //route(SIPOUT); xlog("L_INFO","Setting PRESENCE"); # handle presence related requests route(PRESENCE); # handle registrations xlog("L_INFO","Handling REGISTRAR"); route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN //route(PSTN); # dispatch destinations route(DISPATCH); xlog("L_INFO","**********END*************"); } onreply_route[1] { if (has_body("application/sdp")) rtpengine_answer(); } onreply_route[2] { if (has_body("application/sdp")) rtpengine_offer(); } # Per SIP request initial checks route[REQINIT] { xlog("L_INFO","REQINIT Starting"); if (!mf_process_maxfwd_header("10")) { xlog("L_INFO","483 - Too Many Hops"); sl_send_reply("483","Too Many Hops"); exit; } if(!sanity_check("1511", "7")) { xlog("Sanity Check -> Malformed SIP message from $si:$sp\n"); exit; } xlog("L_INFO","REQINIT Finishing"); } /***********************/ # Caller NAT detection route[NATDETECT] { xlog("L_INFO","Entering NATDTECT"); #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { xlog("L_INFO","Fix Nated Register"); fix_nated_register(); } else { if(is_first_hop()) { xlog("L_INFO","Set Contact Alias"); set_contact_alias(); } } xlog("L_INFO","Set FLT_NATS" + FLT_NATS); setflag(FLT_NATS); } #!endif xlog("L_INFO","NAT Detect set FLT_NTS = " + FLT_NATS); xlog("L_INFO","Finishing NATDETECT"); return; } /***********************/ # Handle requests within SIP dialogs route[WITHINDLG] { xlog("L_INFO","Entering WITHDLG"); if (!has_totag()) return; # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { route(DLGURI); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } else if ( is_method("ACK") ) { # ACK is forwarded statelessy xlog("L_INFO","Going to NATMANAGE"); route(NATMANAGE); } else if ( is_method("NOTIFY") ) { # Add Record-Route for in-dialog NOTIFY as per RFC 6665. record_route(); } route(RELAY); exit; } if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("ACK") ) { if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server route(RELAY); exit; } else { # ACK without matching transaction. Try to route anyway - being optimistic # since it has at least a To Tag route(RELAY); exit; } } sl_send_reply("404","Not here"); xlog("L_INFO","Finishing WITHINDLG"); exit; } # Routing to foreign domains route[SIPOUT] { xlog("L_INFO","Entering SIPOUT"); if (uri==myself) { xlog("L_INFO","URI is MySelf!"); return; } append_hf("P-hint: outbound\r\n"); xlog("L_INFO","Finishing SIPOUT"); route(RELAY); exit; } # Wrapper for relaying requests route[RELAY] { xlog("L_INFO","******** RELAY *******"); # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) { if(!t_is_set("branch_route")) { xlog("L_INFO","branch_route NOT SET!"); t_on_branch("MANAGE_BRANCH"); } } if (is_method("INVITE|SUBSCRIBE|UPDATE")) { if(!t_is_set("onreply_route")) { xlog("L_INFO","onreply_route NOT SET!"); t_on_reply("MANAGE_REPLY"); } } if (is_method("INVITE")) { if(!t_is_set("failure_route")) { xlog("L_INFO","failure_route NOT SET!"); t_on_failure("MANAGE_FAILURE"); } } if (!t_relay()) { xlog("L_INFO","t_relay returns FALSE"); sl_reply_error(); } exit; } # URI update for dialog requests route[DLGURI] { xlog("L_INFO","Entering DLGURI"); #!ifdef WITH_NAT if(!isdsturiset()) { xlog("L_INFO","Handle ruri ALIAS"); handle_ruri_alias(); } #!endif return; } # RTPProxy control and singaling updates for NAT traversal route[NATMANAGE] { xlog("L_INFO","Entering NATMANAGE"); #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { xlog("L_INFO","nat=yes --- Setting FLB_NATB"); setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB) )) { xlog("L_INFO","NO FLT_NATS/B Set!!! Getting out of NATMANAGE"); return; } xlog("L_INFO","Starting RTPEngine"); rtpengine_manage(); if (is_request()) { if (!has_totag()) { if(t_is_branch_route()) { xlog("L_INFO","adding nat=yes"); add_rr_param(";nat=yes"); } } } if (is_reply()) { if(isbflagset(FLB_NATB)) { if(is_first_hop()) { xlog("L_INFO","Set Contact Alias"); set_contact_alias(); } } } #!endif return; } # Handle SIP registrations route[REGISTRAR] { if(!is_method("REGISTER")) return; add_path_received(); route(DISPATCH); } # Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; sl_send_reply("404", "Not here"); exit; } # PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN xlog("L_INFO","Entering PSTN"); # route to PSTN dialed numbers starting with '+' or '0000' # (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","$rU is not outbound call to PSTN"); return; } if (!starts_with("$si","10.19.139")) { xlog("L_INFO","$rU is not comming from any Media GW"); return; } # only local users allowed to call if(from_uri!=myself && !ds_is_from_list("1") ) { xlog("L_INFO", "from_uri $fu != myself, so this is Not Allowed!!"); sl_send_reply("403", "Not Allowed"); exit; } xlog("L_INFO","$si request"); xlog("L_INFO","$rU is an outbound call to PSTN"); $ru = "sip:" + $rU + "@62.28.173.10"; // + $sel(cfg_get.pstn.gw_ip); route(RELAY); /* if(!ds_select_dst("20", "4")) { xlog("L_INFO","no destination selected from dispatcher list!"); send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); */ exit; #!endif xlog("L_INFO"," NO PSTN SET!!"); return; } # Dispatch requests route[DISPATCH] { xlog("L_INFO","Entering DISPATCH"); # hash over callerid dispatching on gateways group '1' if(!ds_select_dst("1", "10","4")) { xlog("L_INFO","no destination selected from dispatcher list!"); send_reply("404", "No destination"); exit; } xlog("L_INFO","going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Dispatch requests route[DISPATCH_TO_PSTN] { # round robin dispatching on gateways group '20' - PSTN if(!ds_select_dst("20", "10")) { send_reply("404", "No destination"); exit; } xlog("L_DBG", "--- SCRIPT: going to <$ru> via <$du>\n"); t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } # Sample failure route failure_route[RTF_DISPATCH] { if (t_is_canceled()) { exit; } xlog("L_INFO", "Media server $du failed to answer, selecting other one!"); # next DST - only for 500 or local timeout if ( t_check_status("500") || (t_branch_timeout() && !t_branch_replied()) ) { #we mark the destination Inactive and Probing ds_mark_dst("ip"); if(ds_next_dst()) { t_on_failure("RTF_DISPATCH"); route(RELAY); exit; } } } route[CDRS]{ sql_query("ca","call kamailio_cdrs();","rb"); sql_query("ca","call kamailio_rating('default');","rb"); } Cheers, *Sérgio *

1 0

single Contact per AOR dilema
by Sergiu Pojoga 16 Apr '19

16 Apr '19

Hi ppl, Say we want to allow 1 contact per AOR. modparam("registrar", "max_contacts", 1) Say some UA looses connection before de-registering, then comes back shortly after (before Contact expires) and tries to register with a different Contact, but gets rejected with error "test_max_contacts(): too many contacts for AOR". What's the approach to handle this situation? I'd like the registrar to accept and save that AoR based on the latest registration. Similar to how Asterisk's single-contact stack `chan_sip` works - whoever was the latest to register successfully gets to be the winner. Thanks, --Sergiu

2 2

dlg_onroute(): failed to update dialog lifetime
by Olli Attila 15 Apr '19

15 Apr '19

Hello, I just activated snmpstats module on Kamailio 5.0.7 and added the following line to route(INITIAL) at the bottom of the route like this (For noc group to monitor Kamailio dialogs): # Per SIP request initial checks route[INITIAL] { ... dlg_manage(); } Now I started to gets this error few times in an hour: /usr/sbin/kamailio[22194]: CRITICAL: dialog [dlg_timer.c:200]: update_dlg_timer(): Trying to update a bogus dlg tl=0x7f798ca51b98 tl->next=(nil) tl->prev=(nil) Apr 15 09:54:05 /usr/sbin/kamailio[22194]: ERROR: dialog [dlg_handlers.c:1439]: dlg_onroute(): failed to update dialog lifetime I see no call failures because of this but still interested what is happening here? --Olli Attila

1 0

one way Audio issue
by Gaurav Bmotra 12 Apr '19

12 Apr '19

i have installed kamailio 5.1.x on Ubuntu server 18.4 issue is i am able to connect the call but there is one way audio not both way as i have install rtpproxy also with following config /etc/default/rtpproxy EXTRA_OPTS="-F -s udp:127.0.0.1:7722 -l PUBLIC_IP -d DBUG:LOG_LOCAL0" /etc/kamailio/kamailio.cnf modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") after restart kamailio i got this logs dev /usr/sbin/kamailio[6926]: INFO: rtpproxy [rtpproxy.c:1559]: rtpp_test(): rtp proxy <udp:127.0.0.1:7722> found, support for it enabled during calling i got this logs Apr 12 16:05:42 ip-172-31-38-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: rtpproxy [rtpproxy.c:2588]: force_rtp_proxy(): incorre Apr 12 16:05:42 ip-172-31-38-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} NOTICE: acc [acc.c:279]: acc_log_request(): ACC: transaction Apr 12 16:05:42 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: db_mysql [km_dbase.c:127]: db_mysql_submit_query(): dr Apr 12 16:05:42 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error whi Apr 12 16:05:42 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: acc [acc.c:477]: acc_db_request(): failed to insert in Apr 12 16:05:43 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13475]: {2 2 INVITE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: rtpproxy [rtpproxy.c:2588]: force_rtp_proxy(): incorre Apr 12 16:05:55 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 BYE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} NOTICE: acc [acc.c:279]: acc_log_request(): ACC: transaction ans Apr 12 16:05:55 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 BYE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: db_mysql [km_dbase.c:127]: db_mysql_submit_query(): drive Apr 12 16:05:55 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 BYE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: <core> [db_query.c:235]: db_do_insert_cmd(): error while Apr 12 16:05:55 ip-xxx-31-xxx-198 /usr/sbin/kamailio[13477]: {2 2 BYE ZTRiYzM4ODMwZTEwMDY3YTE2ZWRlMWRiZDM1NmViNDM.} ERROR: acc [acc.c:477]: acc_db Thanks Gaurav *Regards:* Gaurav Kumar

1 0

Re: [SR-Users] [sr-dev] Audio missing during call
by Mikko Lehto 12 Apr '19

12 Apr '19

Gaurav Bmotra <saigauravmehra91(a)gmail.com> wrote: > HI i have install Kamailio 5.1.2 on Ubuntu 18.4 > there is a audio issue , when we call call from one extension to another ,, > in bigning call goes good , after 9-10 sec it miss audio for one side (one > side can hear other side can't hear ) Hi, mailing list sr-dev is for Kamailio development related topics, please subscribe to sr-users for more general queries. Before posting, make sure you have described in detail your environment and also troubleshooting steps you have done so far: - what modifications you made to default configuration - what happens during SIP transactions in fail scenario - are you using RTP relay or not (rtpproxy/rtpengine) - what have you observed from packet captures Good luck -- Mikko

2 1

command out of sync with db_cluster and db_mysql
by Simone Lazzaris 12 Apr '19

12 Apr '19

Hi everybody; I'm trying to use db_cluster to connect my kamailio installation to a mariadb galera cluster, and it's driving me crazy. In my lab it works without issues, but when I try the setup in the production node, I get some strange errors: Apr 11 14:21:31 ERROR: db_mysql [km_dbase.c:127]: db_mysql_submit_query(): driver error on query: Commands out of sync; you can't run this command now (2014) Apr 11 14:21:31 ERROR: <core> [db_query.c:132]: db_do_query_internal(): error while submitting query This is my configuration: modparam("db_cluster", "connection","con1=>mysql://openser:PASSWD@X.X.X.30/ openser") modparam("db_cluster", "connection","con2=>mysql://openser:PASSWD@X.X.X.31/ openser") modparam("db_cluster", "connection","con3=>mysql://openser:PASSWD@X.X.X.32/ openser") modparam("db_cluster", "cluster", "cls1=>con1=9r8r;con2=9r8r;con3=9r8r") #!define DBURL "cluster://cls1" #!define SQLOPSURL "database=>cluster://cls1" modparam("dispatcher", "db_url", DBURL) modparam("acc", "db_url", DBURL) modparam("usrloc", "db_url", DBURL) modparam("auth_db", "db_url", DBURL) modparam("permissions", "db_url", DBURL) modparam("alias_db", "db_url", DBURL) modparam( "sqlops", "sqlcon", SQLOPSURL ) I'm using version 5.1.4 of kamailio, freshly compiled from source on a debian 9. Any ideas? *Simone Lazzaris* *Qcom S.p.A.* simone.lazzaris(a)qcom.it[1] | www.qcom.it[2] * LinkedIn[3]* | *Facebook[4]* [5] -------- [1] mailto:simone.lazzaris@qcom.it [2] https://www.qcom.it [3] https://www.linkedin.com/company/qcom-spa [4] http://www.facebook.com/qcomspa [5] https://www.qcom.it/includes/email-banner.gif

2 2

kamailio, asterisk and BLF seems its only using json?
by PICCORO McKAY Lenz 12 Apr '19

12 Apr '19

hi, i try to follow the how to in the wiki https://kb.asipto.com/kamailio:presence:k43-blf and configure my two only phones.. but for any states only one led its blinking, the busy state.. no other state work, (if i connect directly to asterisk all work perfeclty, i using asterisk 13 with kamailio 4.4.4) also read the archived https://lists.kamailio.org/pipermail/sr-users/2013-May/078234.html but no luck wht happened with that how to? it's different for asterisk if are integrated? do i need a pjsip asterisk event the oldes chan? see that in presense module parameters, the server address said that value of Contact header filed for 200 OK replies to SUBSCRIBE and PUBLISH and in NOTIFY messages umm if i have integrated and working asterisk (relatime mode) with kamailio, that address muswt be asterisk? taking in cosideration that i have a route rule said that send auth to asterisk and trust any incomming from it? -- Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

1 0

having issue updating $td
by Andrew Chen 11 Apr '19

11 Apr '19

Hey guys, Question. I have this block of code at the beginning of route block: if(is_ip($td)) { xlog("L_INFO", "[$ci][$mi][CSeq $cs] IP detected in To domain. Resolving. \n"); exec_avp("host $td | awk '{print substr($$5, 1, length($$5)-1)}'", "$avp(s:newtd)"); xlog("L_INFO", "[$ci][$mi][CSeq $cs] new newtd = $avp(s:newtd)"); $td = $avp(s:newtd); xlog("L_INFO", "[$ci][$mi][CSeq $cs] new td = $td"); } but $td never updates. Apr 11 18:26:00 sjomaintpsg50 /usr/local/sbin/kamailio[6475]: INFO: <script>: [8ROWdUOINx][1]*[CSeq* 20] IP detected in To domain. Resolving. Apr 11 18:26:00 sjomaintpsg50 /usr/local/sbin/kamailio[6475]: INFO: <script>: [8ROWdUOINx][1]*[CSeq* 20] new newtd = sjomaintpsg50.fuzemeeting.com Apr 11 18:26:00 sjomaintpsg50 /usr/local/sbin/kamailio[6475]: INFO: <script>: [8ROWdUOINx][1]*[CSeq* 20] new td = 206.81.191.30 Is there a reason why? Thanks -- 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 3

Kamailio v5.1.8 Released
by Daniel-Constantin Mierla 11 Apr '19

11 Apr '19

Hello, Kamailio SIP Server v5.1.8 stable release is out. This is a maintenance release of the previous stable branch, 5.1, that includes fixes since the release of v5.1.7. 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 v5.1.8 or to v5.2.x series. For more details about version 5.1.8 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/04/kamailio-v5-1-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the branch 5.1 is the previous stable branch. The latest stable branch is 5.2, at this time with v5.2.2 being released out of it. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

1 0

Kamailio v5.1.8 Released
by Daniel-Constantin Mierla 11 Apr '19

11 Apr '19

Hello, Kamailio SIP Server v5.1.8 stable release is out. This is a maintenance release of the previous stable branch, 5.1, that includes fixes since the release of v5.1.7. 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 v5.1.8 or to v5.2.x series. For more details about version 5.1.8 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/04/kamailio-v5-1-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the branch 5.1 is the previous stable branch. The latest stable branch is 5.2, at this time with v5.2.2 being released out of it. Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

1 0

Releasing Kamailio v5.1.8
by Daniel-Constantin Mierla 11 Apr '19

11 Apr '19

Hello, I am considering to release v5.1.8, an minor version to be built from branch 5.1, on Thursday, this week (April 11, 2019). As usual, check that at least the patches for issues you reported were backported in order to be sure they were not missed during the typical backporting done by developers so far. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

1 1

Using AWS boto3 with KEMI Python routing scripts
by Michael Loughrey 10 Apr '19

10 Apr '19

Hi All, looking for some advice regarding the proper way to initialise an AWS API boto3 client object for send SNS messages within a KEMI Python 2.7 routing script. I'm sure process forking has a major impact on how this works - from kamailio.cfg fork=yes children=4 - but would greatly appreciate some guidance. I have tried various methods to allocate an client object self.sns_client = boto3.client('sns', region_name=MY_AWS_REGION) 1. within __init__(), only once from module initialisation 2. on demand within a function call by any client code e.g. called from ksr_route_request(), each creating it's own client object and so far all have resulted in intermittent crashes within botocore/client.py ultimately crashing thus : Core was generated by `/usr/local/sbin/kamailio -P /usr/local/kamailio/run/kamailio.pid -f /usr/local/'. Program terminated with signal SIGSEGV, Segmentation fault. #0 0x00007f8f9734f754 in ?? () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 (gdb) bt #0 0x00007f8f9734f754 in ?? () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #1 0x00007f8f9734f82e in X509_VERIFY_PARAM_free () from /usr/lib/x86_64-linux-gnu/libcrypto.so.1.1 #2 0x00007f8f97642f5c in SSL_free () from /usr/lib/x86_64-linux-gnu/libssl.so.1.1 #3 0x00007f8f86b09c64 in PySSL_dealloc.lto_priv.5 () at ./Modules/_ssl.c:1598 #4 0x00007f8f99cbb007 in insertdict_by_entry (mp=0x7f8f841f1e88, key='_sslobj', hash=<optimized out>, ep=<optimized out>, value=<optimized out>) at ../Objects/dictobject.c:519 #5 0x00007f8f99cbe2cf in insertdict (value=None, hash=1051385741686792393, key='_sslobj', mp=0x7f8f841f1e88) at ../Objects/dictobject.c:556 #6 dict_set_item_by_hash_or_entry (value=None, ep=0x0, hash=1051385741686792393, key='_sslobj', op={'server_hostname': u'sns.us-east-1.amazonaws.com', '_connected': True, '_context': <SSLContext at remote 0x7f8f84232398>, 'server_side': False, '_makefile_refs': 0, '_closed': False, '_sslobj': None, 'do_handshake_on_connect': True, 'suppress_ragged_eofs': True}) at ../Objects/dictobject.c:795 #7 PyDict_SetItem (op=<optimized out>, key=<optimized out>, value=<optimized out>) at ../Objects/dictobject.c:848 #8 0x00007f8f99becec1 in _PyObject_GenericSetAttrWithDict (obj=<optimized out>, name='_sslobj', value=None, dict={'server_hostname': u'sns.us-east-1.amazonaws.com', '_connected': True, '_context': <SSLContext at remote 0x7f8f84232398>, 'server_side': False, '_makefile_refs': 0, '_closed': False, '_sslobj': None, 'do_handshake_on_connect': True, 'suppress_ragged_eofs': True}) at ../Objects/object.c:1529 #9 0x00007f8f99bed437 in PyObject_SetAttr ( v=<SSLSocket(server_hostname=u'sns.us-east-1.amazonaws.com', _connected=True, _context=<SSLContext at remote 0x7f8f84232398>, server_side=False, _makefile_refs=0, _closed=False, _sslobj=None, do_handshake_on_connect=True, suppress_ragged_eofs=True) at remote 0x7f8f841f62a8>, name=<optimized out>, value=None) at ../Objects/object.c:1247 Here's my setup host : Debian9/stretch python --version Python 2.7.13 kamailio -v version: kamailio 5.2.2 (x86_64/linux) 67f967-dirty 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, MEM_JOIN_FREE ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144 MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 67f967 -dirty compiled on 23:44:31 Apr 8 2019 with gcc 6.3.0 Cheers Mike

2 3

Rtpengine - no RTP packets going out
by Istvan Mogyorosi 10 Apr '19

10 Apr '19

Dear all, This is my first post after reading a lot in this mailing-list. I'm trying to use Kamailio 5.1 with the dispatcher module and rtpengine acting as SIP + RTP proxy. I have 6 asterisk servers in a private subnet that should talk with the peer via a single IP like this: Asterisk 1..n|---> | GW.PRIVATE.IP -o- GW.PUBLIC.IP |----> PEER.SIP.TRUNK I'm on Centos 7, with firewalld configured, iptables module is loaded and the rule is well defined. Packet forwarding is also enabled. Chain rtpengine (1 references) target prot opt source destination RTPENGINE udp -- anywhere anywhere RTPENGINE id:40 My call flow seems to be fine, Kamailio/rtpengine private IP is the outboundproxy parameter of Asterisk instances. My problem is that RTP packets are not present on the public interface, the rtpengine final log showing the 2 sessions, but I'm not sure this is what I want or simply the firewall does not let it out ? (To be more precise PEER.SIP.TRUNK is the trunk for SIP traffic, I have multiple IP addresses for media to connect to, reinvites are allowed) Closing call due to timeout Final packet stats: --- Tag 'as6d12caea', created 1:00 ago for branch '', in dialogue with 'as541b1e61' ------ Media #1 (audio over RTP/AVP) using unknown codec --------- Port GW.PRIVATE.IP:10000 <> 192.168.30.13:11152, SSRC 0, 0 p, 0 b, 0 e, 60 ts --------- Port GW.PRIVATE.IP:10001 <> 192.168.30.13:11153 (RTCP), SSRC 0, 0 p, 0 b, 0 e, 60 ts --- Tag 'as541b1e61', created 1:00 ago for branch '', in dialogue with 'as6d12caea' ------ Media #1 (audio over RTP/AVP) using unknown codec --------- Port GW.PUBLIC.IP:10000 <> PEER.SIP.TRUNK:28216, SSRC 0, 0 p, 0 b, 0 e, 60 ts --------- Port GW.PUBLIC.IP:10001 <> PEER.SIP.TRUNK:28217 (RTCP), SSRC 0, 0 p, 0 b, 0 e, 60 ts Best regards, Istvan

2 2

Kamailio and CDR
by Yu Boot 10 Apr '19

10 Apr '19

Is there any other way to retrieve CDRs from Kami other than dialog/acc modules and writing CDRs to syslog only? Thanks.

6 13

Help request in regards to Kamailio
by Anas ZAHHAF 10 Apr '19

10 Apr '19

Dear, I am writing to ask for help, I found your Email after I've subscribed to Kamailio.org. I am a beginner in Linux environment ( I worked on Windows for 20 years), it would be very helpful if you can help me installing Kamailio on a computer running Debian and how to use the load balancing module so that I can split the charge between 2 asterisk servers, by sharing any useful tutorial or document or even by Teamviewer if this doesn't disturb you. Please help me, all the tutorials I find on internet aren't made for beginners. I will be awaiting your reply and excuse me again if I disturb you. Best regards, -- *Anas ZAHHAF* *105, rue des Moines75017 PARIS* *+33(0)7.66.31.31.85* azahhaf(a)zaion.ai <niluse(a)zaion.ai> | www.zaion.ai ᐧ

2 1

change failure status if reason is q.850 cause=31
by Techinnovation 10 Apr '19

10 Apr '19

Hello, How can I change the status where Reason: q.850;cause=31. ? This code change status for all where we receive status 480 to 404. Is it possible to this code also to add second condition - if reason is like "q.850;cause=31" failure_route[MANAGE_FAILURE] { ... if (t_check_status("480")) { t_reply("404","Unallocated number"); exit; } Need to modify this massage. SIP/2.0 480 Temporarily Not Available. Allow: UPDATE,REFER,INFO. Call-ID: 6b11b153405a5103330697b774b97f5a@192.168.2.2:5060. Contact: <sip:192.168.3.3:5060>. CSeq: 102 INVITE. From: "380668375820" <sip:380668375820@192.168.2.2>;tag=as445242c0. Reason: q.850;cause=31. Server: Cirpack/v4.76 (gw_sip). To: <sip:9052946032@192.168.3.3:5060>;tag=00-07929-6e4a3079-75447e2a5. Content-Length: 0. Thanks. BR, Aleksandr

2 1

Question regarding retcode
by Andrew Chen 09 Apr '19

09 Apr '19

Hey all, Question regarding the type of retcode. I ran a curl command and $retcode returned 200. I check against that variable if it's 200 or not. I tried to check 200 as a string and integer but never matches so I ran these lines of code: xlog("L_INFO", "[$ci][$mi][CSeq $cs] Validating meeting ($rU). Result: $var(result) Return code: $retcode \n"); if(is_int($retcode)) { xlog("L_INFO", "[$ci][$mi][CSeq $cs] retcode is an integer \n"); } if(typeof("$retcode", "str")) { xlog("L_INFO", "[$ci][$mi][CSeq $cs] retcode is a string \n"); } None of these xlog outputed. How do I check for $retcode in if or switch statement? Thanks -- 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.*

2 2

Centos 7 opensuse repo issues for 5.2
by Jeffrey Gray 09 Apr '19

09 Apr '19

Good afternoon, We are unable to get kamailio 5.2 to install using the recommended opensuse repos for Centos 7. Is there any way to resolve the following errors: Thanks very much for your assistance! using the following repo setup in /etc/yum.repos.d/ [home_kamailio_v5.2.x-rpms] name=RPM Packages for Kamailio v5.2.x (CentOS_7) type=rpm-md baseurl=http://download.opensuse.org/repositories/home:/kamailio:/v5.2.x-rp… gpgcheck=1 gpgkey=http://download.opensuse.org/repositories/home:/kamailio:/v5.2.x-rpm… enabled=1 Resolving Dependencies --> Running transaction check ---> Package kamailio.x86_64 0:5.2.2-3.el7.centos will be installed --> Finished Dependency Resolution Dependencies Resolved ============================================================================================================================================================================= Package Arch Version Repository Size ============================================================================================================================================================================= Installing: kamailio x86_64 5.2.2-3.el7.centos home_kamailio_v5.2.x-rpms 5.6 M Transaction Summary ============================================================================================================================================================================= Install 1 Package Total download size: 5.6 M Installed size: 25 M Downloading packages: kamailio-5.2.2-3.el7.centos.x8 FAILED ================- ] 2.3 MB/s | 1.7 MB 00:00:01 ETA http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.2.x-rpms/C…: [Errno -1] Package does not match intended download. Suggestion: run yum --enablerepo=home_kamailio_v5.2.x-rpms clean metadata Trying other mirror. Error downloading packages: kamailio-5.2.2-3.el7.centos.x86_64: [Errno 256] No more mirrors to try. If we run the suggested clean metadata command, then the error changes to: sudo yum -y install kamailio Loaded plugins: fastestmirror Loading mirror speeds from cached hostfile * base: mirrors.advancedhosters.com * epel: mirror.cogentco.com * extras: mirrors.advancedhosters.com * updates: mirrors.advancedhosters.com Resolving Dependencies --> Running transaction check ---> Package kamailio.x86_64 0:5.2.2-3.el7.centos will be installed --> Finished Dependency Resolution Dependencies Resolved ============================================================================================================================================================================= Package Arch Version Repository Size ============================================================================================================================================================================= Installing: kamailio x86_64 5.2.2-3.el7.centos home_kamailio_v5.2.x-rpms 5.6 M Transaction Summary ============================================================================================================================================================================= Install 1 Package Total download size: 5.6 M Installed size: 25 M Downloading packages: Delta RPMs disabled because /usr/bin/applydeltarpm not installed. kamailio-5.2.2-3.el7.centos.x8 FAILED http://download.opensuse.org/repositories/home%3A/kamailio%3A/v5.2.x-rpms/C…: [Errno 14] HTTPS Error 416 - Requested Range Not Satisfiable Trying other mirror. Error downloading packages: kamailio-5.2.2-3.el7.centos.x86_64: [Errno 256] No more mirrors to try.

3 3

Upstream ignores reply, because To-tag doesn't match?
by Sergiu Pojoga 09 Apr '19

09 Apr '19

Hi ppl, Scenario: invite from upstream is t_relayed to a client gateway. After `fr_inv_timeout` occurs, in a failure route I simply t_reply with "503 - Service unavailable", then exit(). However, the upstream provider simply ignores this reply, call doesn't hang up. This doesn't happen when the client's gateway generates identical 503 or other negative replies. I suspect this is happening because the To-tag in the Kam generated 503 reply doesn't match with the To-tag that previously was forwarded from the client's gateway to the upstream provider in the `180 Ringing`. Any suggestions what can be done about it? 2019/04/06 10:42:46.127177 65.39.xxx.xxx:5060 -> 208.72.xxx.xxx:5060 SIP/2.0 180 Ringing Via: SIP/2.0/UDP 208.72.xxx.xxx:5060;received=208.72.xxx.xxx;rport=5060;branch=z9hG4bK1982422573 Record-Route: <sip:65.39.xxx.xxx;lr=on;did=915.8af1> From: "SERGIU" <sip:514XXXXXXX@208.72.xxx.xxx>;isup-oli=00;tag=1975755942 To: <sip:514XXXXXXX@65.39.xxx.xxx>;tag=as2ab54180 Call-ID: DID-28270826(a)208.72.xxx.xx CSeq: 477023 INVITE Supported: replaces, timer, path Contact: <sip:1514XXXXXXX@65.39.xxx.xxx:5060> Content-Length: 0 2019/04/06 10:42:51.072516 65.39.xxx.xxx:5060 -> 208.72.xxx.xxx:5060 SIP/2.0 503 Service Unavailable Call-ID: DID-28270826(a)208.72.xxx.xx Via: SIP/2.0/UDP 208.72.xxx.xxx:5060;rport=5060;branch=z9hG4bK1982422573;received=208.72.xxx.xxx From: "SERGIU" <sip:514XXXXXXX@208.72.xxx.xxx>;isup-oli=00;tag=1975755942 To: <sip:514XXXXXXX@65.39.xxx.xxx>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8e82 CSeq: 477023 INVITE Server: KAM Content-Length: 0 Thanks, --Sergiu

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modifying via header from public IP to private IP
by Narasimhaiah, Vinod Maralenahalli 09 Apr '19

09 Apr '19

Hi , I have a cloud server, when I make call from a sip phone (registered with kamailio), in the INVITE header I am seeing the via header with public IP, is there any way that I can modify via header to send only the private IP. The calls are landing on another sip gateway, they will allow only the private IP. Please let me know, is it possible to replace. I tried all the replace functions but it is not working. Thank you vinod.M.N -- ***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.***

1 0

modifying via header from public IP to private IP
by vinod mn 08 Apr '19

08 Apr '19

Hi , I have a cloud server, when I make call from a sip phone (registered with kamailio), in the INVITE header I am seeing the via header with public IP, is there any way that I can modify via header to send only the private IP. The calls are landing on another sip gateway, they will allow only the private IP. Please let me know, is it possible to replace. I tried all the replace functions but it is not working. -- Thanks and regards Vinod.M.N

3 2

How to Register sip trunk in kamailio without username and password?
by Mostafa Moradi 08 Apr '19

08 Apr '19

Hi, I'm mostafa moradi from iran, I Have installed kamailio, and I have a sip trunk without username and password, I want to register the sip trunk in kamailio with UAC Module, but this module only accepts sip trunk with user and password, How Can I Register sip trunk in kamailio without username and password? Thank you in advance Best regards,

4 3

Re: [SR-Users] Timeout between 100 & 183
by Alex Balashov 05 Apr '19

05 Apr '19

On Fri, Apr 05, 2019 at 06:23:16PM -0400, Alex Balashov wrote: > route { > ... > > # Initial INVITE handling: > > if(is_method("INVITE")) { > # Set fr_inv_timer initially to 5 seconds, so that > # if we do not receive an answer or at least one non-100 1xx > # response, trigger a failure_route and try the next gateway > # or whatever. > > t_set_fr(50000); Sorry, this would have been a lot more pedagogically useful if it had read: t_set_fr(5000); -- 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/

1 0

UAC authentication challenge response using KEMI/app_ruby
by Andrew White 05 Apr '19

05 Apr '19

Hi all, I’m continuing on my quest to use app_ruby for my logic - it’s working very well so far! The last major hurdle I’ve come across is UAC response to WWW challenge. On my outbound path, the correct URIs for From and To are set based on provider requirements, and the INVITE is sent using t_relay(): def ksr_route_to_trunk() KSR.info("Route to trunk") if KSR::TM.t_is_set("failure_route") < 0 then KSR::TM.t_on_failure("ksr_failure_manage_trunk") end # Relay KSR::PV.sets("$fu", "sip:#{$uacs[trunk]['r_username']}@#{$uacs[trunk]['r_domain']}") KSR::PV.sets("$fn", '') new_uri = "sip:61400123123@#{uacs[trunk]['r_domain']}" KSR.info("New URI is #{new_uri}") KSR.seturi(uri) KSR::UAC.uac_replace_to('', uri) KSR::TM.t_relay() exit end I’ve stripped it back, but this function is great and works perfectly. An INVITE is sent to the correct place, and the trunk receives the INVITE. It then responds with a 401/WWW-Authenticate. My failure route is called fine. Below is the stripped down code: def ksr_failure_manage_trunk() exit if KSR::TM.t_is_canceled() > 0 KSR.info("Hit trunk failure manage route") KSR::PV.sets("$avp(auser)", 'user') KSR::PV.sets("$avp(apass)", 'pass') KSR::PV.sets("$avp(arealm)", 'test.local') KSR.info("AVPs: #{KSR::PV.gete("$avp(auser)")} #{KSR::PV.gete("$avp(apass)")} #{KSR::PV.gete("$avp(arealm)")}") if KSR::UAC.uac_auth() then KSR.info("UAC authed, relaying") KSR::TM.t_relay() else KSR.info("UAC NOT authed, no relay") end exit end Also the AVP values are set (I believe) correctly: modparam("uac", "auth_username_avp", "$avp(auser)") modparam("uac", "auth_password_avp", "$avp(apass)") modparam("uac", "auth_realm_avp", "$avp(arealm)") This throws the following complaint, and doesn’t send a new response back to the trunk: Apr 5 02:55:22 ip-10-0-0-2 /usr/local/sbin/kamailio[26870]: ERROR: {1 102 INVITE 3a84da15272da66e527a79144c48516c@10.0.0.20:5060} tm [t_fwd.c:1728]: t_forward_nonack(): no branches for forwarding Apr 5 02:55:22 ip-10-0-0-2 /usr/local/sbin/kamailio[26870]: ERROR: {1 102 INVITE 3a84da15272da66e527a79144c48516c@10.0.0.20:5060} tm [tm.c:1619]: _w_t_relay_to(): t_forward_noack failed I don’t understand branches very well, however after some digging around, I appended KSR::COREX.append_branch() in before the t_relay() in the failure. This stops the error and now we send an ACK and a new INVITE, but still without the expected authorisation header. I feel like my lack of understanding about branches is the issue here - I seem to be sending an INVITE, but not adding the auth header that I suspect is being generated somewhere! Thanks for your help. ________________________________ Andrew White - Director uConnected Email: andrew(a)uconnected.com.au Web: www.uConnected.com.au

2 2

Statistics of received replies
by Duarte Rocha 04 Apr '19

04 Apr '19

Greetings, Does Kamailio have statistics like the sl.stats but for received replies instead of locally generated ones? I want to be able to look at how many "491" replies i'm receiving, for example. Best Regards, Duarte Rocha

2 1

Kamailio v5.0.8 Released
by Daniel-Constantin Mierla 04 Apr '19

04 Apr '19

Hello, Kamailio SIP Server v5.0.8 stable release is out. This is a maintenance release of the old stable branch 5.0, that includes fixes since the release of v5.0.7. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.0.x. Deployments running previous v5.0.x versions are strongly recommended to be upgraded to v5.0.8 or even better directly to v5.1.x or v5.2.x because v5.0.x is getting out of the official maintenance time frame. Important note: this is the last planned release from branch5.0. For more details about version 5.0.8 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2019/04/kamailio-v5-0-8-released/ RPM, Debian/Ubuntu packages will be available soon as well. Note: the latest stable branch is 5.2, at this moment with its latest release v5.2.2. See more details about it at: * https://www.kamailio.org/w/kamailio-v5-2-0-release-notes/ Many thanks to all contributing and using Kamailio! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

1 0

Kamailio-error reading: Connection reset by peer
by Mangesh Kharade 04 Apr '19

04 Apr '19

Hi Team, We are facing issue with kamailio application , we are not able to communicate to kamailio application getting below error. Can you please suggestion us how to resolved this issue issue, it production servers and we did not get any clue to resolve it. Error1:- Apr 2 06:53:54 urcmcs /usr/local/sbin/kamailio[2240]: ERROR: <core> [tcp_read.c:299]: tcp_read_data(): error reading: Connection reset by peer (104) Apr 2 06:53:54 urcmcs /usr/local/sbin/kamailio[2240]: ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading Apr 2 06:54:12 urcmcs /usr/local/sbin/kamailio[2244]: ERROR: <core> [tcp_read.c:299]: tcp_read_data(): error reading: Connection reset by peer (104) Apr 2 06:54:12 urcmcs /usr/local/sbin/kamailio[2244]: ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading Apr 2 06:54:17 urcmcs /usr/local/sbin/kamailio[2244]: ERROR: <core> [tcp_read.c:299]: tcp_read_data(): error reading: Connection reset by peer (104) Apr 2 06:54:17 urcmcs /usr/local/sbin/kamailio[2244]: ERROR: <core> [tcp_read.c:1326]: tcp_read_req(): ERROR: tcp_read_req: error reading Error1:- Apr 2 06:54:26 urcmcs /usr/local/sbin/kamailio[2179]: INFO: <script>: WebSocket connection from ###################:3008 has closed Apr 2 06:54:26 urcmcs /usr/local/sbin/kamailio[2179]: INFO: <script>: WebSocket connection from ###################:3009 has closed Apr 2 06:54:26 urcmcs /usr/local/sbin/kamailio[2179]: INFO: <script>: WebSocket connection from ###################:3010 has closed Apr 2 06:54:26 urcmcs /usr/local/sbin/kamailio[2179]: INFO: <script>: WebSocket connection from ###################:3011 has closed Apr 2 06:54:26 urcmcs /usr/local/sbin/kamailio[2179]: INFO: <script>: WebSocket connection from ###################:3012 has closed Please let us know if you need more help from our end. Thanks and Regards, Mangesh Kharade Consultant - Microsoft Azure | Managed Services [cid:image003.png@01D15FFB.27B1AA80] Pune IT Park, 34 Aundh Road Bhau Patil Marg, Pune 411020 Tel: +91.20.30204000 | Extn. 9694 VOIP: +1.713.353.7464 Mobile: +91.9168181166 Mangesh.kharade(a)silicus.com<mailto:Mangesh.kharade@silicus.com>| www.silicus.com<http://www.silicus.com/> The information in this email is confidential and may be legally privileged. It is intended solely for the addressee and access to it by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, distribution or any action taken or omitted to be taken based on it, is strictly prohibited and may be unlawful.

2 2

Releasing Kamailio v5.0.8
by Daniel-Constantin Mierla 04 Apr '19

04 Apr '19

Hello, I am considering to release v5.0.8 to mark the end of official maintenance for branch 5.0 on Thursday, April 4. If no major regression will be introduced, this will be the last release from branch 5.0. Afterwards, the stable branches maintained will be 5.1 and 5.2. In case you have in mind some specific commits that should get into 5.0 branch, then reply and list them to be sure they are not missed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

2 2

Kamailio & asterisk integration
by ayoub krari 04 Apr '19

04 Apr '19

Good morning, I want to Loadbalance calls equally to asterisk servers, i have already tried with opensips but didnt work for me, i want to try kamailio now, i have tried with the tutorial of " Kamailio 4.0.x and Asterisk 11.3.0 Realtime Integration using Asterisk Database" but i got some errors and packages missing during the installation and the configuration, my Linux distro is Debian 8.7.0 and i’m using Asterisk v 13, i want really to implement Kamailio , any one can help me with steps or anything , Thanks in advance !

2 1

dispatcher "failover feature disabled"
by David Villasmil 03 Apr '19

03 Apr '19

Hello guys, In my config i have: # ----- dispatcher params ----- modparam("dispatcher", "db_url", DBURL) modparam("dispatcher", "table_name", "dispatcher") modparam("dispatcher", "flags", 2) modparam("dispatcher", "dst_avp", "$avp(AVP_DST)") modparam("dispatcher", "grp_avp", "$avp(AVP_GRP)") modparam("dispatcher", "cnt_avp", "$avp(AVP_CNT)") modparam("dispatcher", "dstid_avp", "$avp(dsdstid)") modparam("dispatcher", "attrs_avp", "$avp(dsattrs)") # PVs for hashing modparam("dispatcher", "hash_pvar", "$fU@$ci") # PVs to store results when calling ds_is_from_list modparam("dispatcher", "setid_pvname", "$var(setid)") modparam("dispatcher", "attrs_pvname", "$var(attrs)") # method to probe the gateways modparam("dispatcher", "ds_ping_method", "OPTIONS") modparam("dispatcher", "ds_ping_from", " sip:dispatcher@proxy.voice-proxy.singlecommand.com") modparam("dispatcher", "ds_ping_interval", 10) modparam("dispatcher", "ds_probing_threshold", 1) modparam("dispatcher", "ds_ping_reply_codes","class=2") modparam("dispatcher", "ds_probing_mode", 1) # size of hash table storing data for call load dispatching, power of two modparam("dispatcher", "ds_hash_size", 10) # expiration time in seconds to remove the load on a destination if no BYE was received modparam("dispatcher", "ds_hash_expire", 3600) # expiration time in seconds to remove the load on a destination if no 200 OK for INVITE was received # and state updated with ds_load_update modparam("dispatcher", "ds_hash_initexpire", 60) modparam("dispatcher", "ds_hash_check_interval", 30) But i'm getting on startup: ds_warn_fixup(): failover functions used, but required AVP parameters are NULL -- feature disabled What avp am I missing? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

2 3

Re: [SR-Users] Adding presentity_uri prefix for multi-tenanted presence/dialoginfo configuration
by Rhys Hanrahan 03 Apr '19

03 Apr '19

Hi All, I managed to solve my need to re-write NOTIFY packets with help from this thread: https://www.mail-archive.com/sr-users@lists.kamailio.org/msg08088.html This seems to work well. I am re-writing the packets as shown below… I guess my only question is, is this relatively safe/smart to do? I basically copied what our standalone Asterisk PBXes are doing in my captures, so I guess so? But I’m not confident enough to say it won’t cause some weird edge case issues. I can only say it works fine with the Snom and Yealink handsets I am testing with. I’m just concerned because it feels a bit like a hack. For SUBSCRIBE I re-write: * Request URI and To header from 101@ sent by the handset to tenant101@ (to match the SIP username of the handset - tenant101@) For NOTIFY I re-write: * From and Contact header to remove the tenant prefix, to go back to 101@ to match the BLF key configuration on the handsets. Any advice would be appreciated on if this is a bad idea, and there’s a better way. Thanks! Rhys Hanrahan Chief Information Officer Nexus One Pty Ltd E: support(a)nexusone.com.au<mailto:support@nexusone.com.au> P: +61 2 9191 0606 W: http://www.nexusone.com.au/ M: PO Box 127, Royal Exchange NSW 1225 A: Level 12 227 Elizabeth St, Sydney NSW 2000 From: sr-users <sr-users-bounces(a)lists.kamailio.org> on behalf of Rhys Hanrahan <rhys(a)nexusone.com.au> Reply-To: "Kamailio (SER) - Users Mailing List" <sr-users(a)lists.kamailio.org> Date: Tuesday, 26 March 2019 at 8:56 pm To: "sr-users(a)lists.kamailio.org" <sr-users(a)lists.kamailio.org> Subject: [SR-Users] Adding presentity_uri prefix for multi-tenanted presence/dialoginfo configuration Hi Everyone, I’ve recently just started working with Kamailio – thanks everyone for this amazing software. Like many people, I’m in the process of trying to put Kamailio in-front of Asterisk to allow it to scale out, and my plan is for Kamailio to take over registrations, usrloc and presence/dialoginfo, as we’ve had issues where handsets are failing to get BLF updates/notifys, so I am hoping a couple of Kamailio boxes can scale better in this regard. DMQ is particularly exciting in how it will allow me to build a truly distributed platform. But I’m struggling to get presence to work the way I need it to. Our environment is multi-tenanted, but we do not (and can’t really) use multi-domains. Instead, we prefix the SIP usernames with the “tenant name” such as tenanta101 and tenantb101. My problem is that all the BLFs are configured for 101@PBX with no tenant name in the User part of the URI so that internal dialing and call pickup will work. Because in Asterisk we use “subscribecontext” this hasn’t been a problem in the past – Asterisk knows for subscriptions coming from that handset that it belongs to that tenant’s context in the dialplan and they are “isolated”, so having handsets subscribe as 101@ in their request URI was never a problem. Of course, with Kamailio I don’t have subscribecontext, and my main issue is that the “presentity_uri” being stored is 101@ while each of the SIP accounts of the handsets are registered as tenenata101@, and as such, no NOTIFYs are sent by Kamailio because it thinks that there are no watchers for the presentity_uri. If I change the SIP account username to 101 to match the BLF key, NOTIFYs are sent as expected. But then this breaks call pickup and internal dialing using the BLF keys. I would rather handle this in Kamailio than in Asterisk and having to re-configure the BLF keys for hundreds of handsets. I need to do something like: * When a SUBSCRIBE comes in, I need to prefix presentity_uri with the tenant name so the subscription changes from 101@ to tenanta101@ in the active_watchers table. * When Kamailio generates a NOTIFY, the notify would be built based on presentity_uri and would come out as tenanta101@, but the phone’s BLFs are configured for 101@ so I would need to *remove* the tenant prefix before sending out the notify. Is there a good, or recommended way to handle this scenario? Maybe something entirely different to changing the presentity_uri? I’ve written the following config to re-write the request URI and To field in any incoming subscribe requests, and successfully got the active_watchers table to store a presentity_uri containing my tenant prefix. But the problem I am having is that I can’t figure out how to modify the NOTIFY packets before they are sent – and looking at cfgtrace, it seems I might not be able to? My “standard” presence + dialoginfo configuration was taken from here as a starting point: https://kb.asipto.com/kamailio:presence:k43-blf Here is what I have so far in terms of trying to make my modifications – any guidance would be greatly appreciated. It feels like there’s probably a better way to do this than re-writing critical headers like To and Request URI? @@ -466,6 +467,8 @@ request_route { # authentication route(AUTH); + route(REWRITE_PRESENCE); + # record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); # Presence server processing route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return; if(is_method("SUBSCRIBE") && $hdr(Event)=="message-summary") { # Asterisk is our voicemail route(TOASTERISK); # returns here if no voicemail server is configured sl_send_reply("404", "No voicemail service"); exit; } #!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; } if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if(is_method("SUBSCRIBE")) { # See REWRITE_PRESENCE - this should have been executed before we get here. handle_subscribe(); t_release(); } exit; #!endif +route[REWRITE_PRESENCE] { + if (is_method("SUBSCRIBE")) { + xlog("Re-writing subscribe to include tenant prefix\n"); + # The default presentity_uri needs to be prefixed with + # the tenant name + # So re-write the To header + route(TENANTINFO); + # Now grab the tenant name. + $var(subscribe_ru) = "sip:" + $var(tenant_name) + $rU + "@" + $rd; + xlog("Re-writing SUBSCRIBE To header as: $var(subscribe_ru)\n"); + insert_hf("To: $var(subscribe_ru)\r\n", "From"); + $ru = $var(subscribe_ru); + # Force change immediately: + # http://www.kamailio.org/wiki/tutorials/faq/main#why_changes_made_to_headers… + msg_apply_changes(); + } else if (is_method("NOTIFY")) { + xlog("Re-writing NOTIFY to remove tenant prefix\n"); # This code block does not get executed right now. + # We are storing the presentity_uri with a tenant prefix + # handsets do not expect this, only 101@, 102@ etc... In their BLF configs + # So this tenant prefix must be removed when building the reply. + route(TENANTINFO); + $var(notify_ru) = "sip:" + $(rU{s.substr,$(var(tenant_name){s.len}),0}) + "@" + $rd; + xlog("Re-writng NOTIFY to use: $var(notify_ru)\n"); + insert_hf("To: $var(notify_ru)\r\n", "From"); + $ru = $var(notify_ru); + # Force change immediately: + # http://www.kamailio.org/wiki/tutorials/faq/main#why_changes_made_to_headers… + msg_apply_changes(); + } +} Thanks for your time. Rhys Hanrahan Chief Information Officer Nexus One Pty Ltd E: support(a)nexusone.com.au<mailto:support@nexusone.com.au> P: +61 2 9191 0606 W: http://www.nexusone.com.au/ M: PO Box 127, Royal Exchange NSW 1225 A: Level 12 227 Elizabeth St, Sydney NSW 2000 [ttp://quintus.nexusone.com.au/~rhys/nexus1-email-sig.jpg]

1 0

uac_replace_from/to, msg_apply_changes and Re-INVITE
by Henning Westerholt 03 Apr '19

03 Apr '19

Hello, I am currently debugging a strange issue related to the usage of uac_replace_from/to() after msg_apply_changes(). It works all fine, but in a Re-INVITE case it inserts the wrong headers for the 100 - Trying case. The calle side gets then a reply with wrong From/To. This happens all the time with this particular cfg. I already checked for the usual issues: - msg_apply_changes() called several times - msg_apply_changes() called after transactions created/async processing etc.. Any ideas what else could be the problem? The msg_apply_changes() is needed for some SDP filtering logic in the cfg. Thanks and regards, Henning -- Henning Westerholt - https://skalatan.de/blog/ Kamailio services - https://skalatan.de/services

3 8

TOPOS Call-ID masking
by Cinthia Leung 02 Apr '19

02 Apr '19

Greetings, Call-ID masking is not available, although the modparam is there. Just wondering if there are any plans to get that in soon. Thanks. Cindy

2 2

ENUM for CNAM dips
by Spencer Thomason 02 Apr '19

02 Apr '19

Hi all, I’m interested in extending the ENUM module to add support for draft-ietf-enum-cnam-08 CNAM queries using ENUM. I’m looking for high level guidance as to the best way to implement this and any feedback on naming of exported functions and parameters. Maybe something like enum_cnam_query("pvar", "destination", [,"suffix" [,"service"]]) ? Best regards, Spencer

2 1

Info: Kamailio World 2019 – Participation Grants
by Daniel-Constantin Mierla 02 Apr '19

02 Apr '19

Hello, we continue our tradition to give away 3 free passes to students and underrepresented people at Kamailio World Conference (www.kamailioworld.com) – thanks to the sponsors, we are able to offer them in 2019 as well. The passes cover the entire registration fee for all three days and the social networking events – costs with travel and accommodation have to be covered by the participants. Anyone enrolled in an academic activity (including master and PhD programs) as well as underrepresented people can apply for one of the passes via email to me or to: * conference [at] kamailioworld.com or by contacting us via web form at: * https://www.kamailioworld.com/k07/contact-us/ Applications done until April 15, 2019, will be selected by April 18, 2019. Afterwards, if there are still available passes, they will be allocated in the first come first served policy. The details for most of the presentations were published, the full agenda is expected to be ready in a matter of a few days. You can see more details at: * https://www.kamailioworld.com/k07/schedule/ Looking forward to meeting many of you in Berlin, at Kamailio World 2019! It’s going to be another great event about open source real time communications! * https://www.kamailioworld.com/k07/registration/ Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - May 6-8, 2019 -- www.kamailioworld.com

1 0

Port 5060 omitted in Record-Route
by Lợi Đặng 02 Apr '19

02 Apr '19

Hi all, i was using kamailio 4.2.1 located in 2 networks > listen = tcp:MY_ADDR:5060 advertise MY_ADDR:5060 > listen = tls:MY_ADDR:5061 advertise PUBLIC_NAT_ADDR:5061 when the call made from the inside network to out side, running `record_route()` resulted in 2 Record-Route headers added (enable_double_rr=1) Record-Route: PUBLIC_NAT_ADDR:5061;transport=tls;lr > Record-Route: MY_ADDR;transport=tcp;lr That was totally fine omitting the port in the first Record-Route when using tcp (or udp) on the first realm, but when i start switching to tls, it caused trouble Record-Route: PUBLIC_NAT_ADDR:5061;transport=tls;lr > Record-Route: MY_ADDR;transport=tls;lr The client is then told to send ACK/BYE to `MY_ADDR;transport=tls` located at `MY_ADDR:5061` as per rfc3263, then the call would failed. I had another try with `record_route_preset("PUBLIC_NAT_ADDR:5061;transport=tls", "MY_ADDR:5060;transport=tls");`, it really did add what i want with explicit 5060 port on RR, `ACK/BYE` travel on the correct path, but `loose_route()` only consumes the local `Route` header (it should consume 2). So my assumption is to stick with `record_route()` function to make `loose_route()` work properly. I tried using another port on the local realm, e.g: 5062 and the port is explicitly added to the Record-Route header `MY_ADDR:5062;transport=tls;lr` So is `5060` couldn't be explicitly added to the inbound Record-Route, or i just missed something? Any help will be appreciated. P/S: I also tried 4.4.7 and it still omit my 5060 port in the RR. rgds, Loi Dang Thanh Phone : +84. 774.735.448 Email : loi.dangthanh(a)gmail.com

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Compressing SIP messages
by Igor Olhovskiy 01 Apr '19

01 Apr '19

Hi! Was looking on some way to compress SIP signalling between internal servers. But found out, there are no some pre-defined or recommended mechanism of doing this. Yes, I found gzcompress module, but it will help only in case of big body, but not whole message itself. Like in some cases you can have 4-5 RR headers, which is already a lot of info. What I googled is SigComp Compact Header form As I found, SigComp is really not widely-used method and it support along opensource SIP servers/PBXes is not documented. For Compact Header form - is there any way of using Kamailio to compress message this way? Maybe also there is a way of having some intermediate proxy-layer software, that will gzip/gunzip SIP signalling in realtime? All of this actually cause info in SIP/SDP grows rapidly, unfortunately MTU size is not.

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presence_dialoginfo module not sending publish on dialogue change
by Richard Nunley 01 Apr '19

01 Apr '19

To whom can be of assistance, I’m am working towards enabling presence on my Kamailio proxy but have hit a bit of a brick wall. I’m seeing dialogues being created and persisted to the database, subscriptions are being handled and rows are being written to the active_watchers table but no PUBLISH is ever being received by the server to kickoff the sending of NOTIFY to the active watchers. Further, I’m never seeing a row in the watchers table though I don’t know that this isn’t expect behavior. I’ve attached my current config for review. Any help will be greatly appreciated! Thank in advance!

3 2

Use Kamailio as forward proxy between two networks
by Lauritz 01 Apr '19

01 Apr '19

Hello, I am trying to implement the following network setup using Kamailio and RTPProxy, but was not able to find any recent documentation or blog posts with hints: > local_net1 : local_net2 > 10.250.10.* : 10.250.11.* > : > (1) (2) : > +-------------+ +--------------+ +--------------+ > ! !.11 .12 ! Proxy !.21 ! Debian Host !.22 > ! Starface !---------------! + Kamailio !<<--------------! + Linphone !.23 > ! ! ! + RTPProxy ! ! ! > +-------------+ +--------------+ +--------------+ > eth0 : eth1 > : > : > As you can see, the proxy has two interfaces and its purpose is to forward traffic from the client's network to the Starface's network. Has anyone experience to share or got any other hints where to start? Thank you very much for your help :) Kind regards, Lauritz

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Kamailio Managed Cloud
by Alex Balashov 01 Apr '19

01 Apr '19

For immediate release: ATLANTA, GA (1 April 2019)--Following its long tradition of forward-thinking innovation in open-source Real-Time Communications (RTC), the leadership of the Kamailio SIP server project (formerly known as OpenSER) has launched the Kamailio Managed Cloud (KMC). Alex Balashov of Evariste Systems and Fred Posner of The Palner Group were among several key North American stakeholders on hand to discuss it further. As Posner told us, "Kamailio Managed Cloud is a new, best-of-breed managed Kamailio hosting environment. We provide elasticity by leveraging human capital. Using our patent-pending Orchestration Turk system, our managed cloud technicians will purchase your SuperMicro bare-metal server, known as a KamBox, drive to the data centre, and rack-and-stack it on demand -- all in response to a SOAP API call! And when you want to remove your on-demand KamBox, just issue a 'delete' operation and we will decommission the hardware -- live!" "Dedicated KamBox colocation really puts the Management in Managed Cloud," emphasised Balashov. "But that doesn't mean we keep the customer out of the loop. We are using the most modern XMLHttpRequest and 'AJAX' techniques to provide a real-time dashboard with KamBox status to all users, though it will not work with Internet Explorer 6 -- we're addressing that." "Anyway, there are numerous other cloud platforms out there, many of which rely on virtualisation and automated orchestration and the like. It really takes the human touch out of the equation and gives the short shrift to quality, while also taking away jobs. And besides, how can a so-called 'virtual' machine be truly carrier-grade?" Kamailio Managed Cloud launched in two principal regions today -- US-EAST-1, located in Jacksonville, Florida, USA, and EU-CENTRAL-1, based in the booming tech hub of central Europe and birthplace of OpenSER: Berlin, Germany. Unnamed sources tell us that strategic expansion is planned for several other regions, including Flint, Michigan, USA. The mechanics of Kamailio Managed Cloud have invited questions about how other elastic cloud resources are managed, more especially elastic storage. According to KMC senior management, storage is provided via a mechanism developed by ASIPTO StorageWorks. ASIPTO chief Daniel-Constantin Mierla was on hand at the KMC launch to explain: "It's very easy. When a customer requests more space with a simple and lightweight SOAP call with just a little custom XML namespaces, then depending on the region, Fred or I receive an SMS on our smartphones from our advanced PHP backend and drive to the data centre to insert another hard drive into one of the KamBox's several drive bays. We are using the latest in advanced mobile communications, SMS, to ensure fastest response time. Of course our service level is comprehensive and includes Managed Partition Table Labelling and Managed LVM Insertion. Also there is optional Managed Filesystem Formatting for a complete portfolio of managed services. ASIPTO StorageWorks moves at the speed of global e-business. Just text us your desired filesystem layout and partitioning scheme. It is really that easy!" When asked about business focus, marketing turnaround veteran Henning Westerholt, Executive Vice President of Business Development for KMC responded: "Our target customer is a socially conscious Millennial-run corporation or start-up who see value in the unique blend of managed services and an artisanal, hand-crafted, organic approach that is compatible with social justice and takes a people-centric view." Asked to elaborate upon his vision of social justice, Westerholt added: "Virtualisation, orchestration and auto-scaling take the people out of the equation, and Kamailio is, in the end, about people, as communication in general is about people. What would it be without the people? Moreover, these mechanisms take away gainful opportunities from hard-working university graduates. Kubernetes, Docker Swarm, VMWare, and KVM contravene the social policy objective of full employment." At press time, Posner, who also heads up KMC Labs North America, was said to be lending his pioneering business vision and knack for seeing around technology trend corners to intensive work on next-generation, bleeding-edge functionality for KMC. While Posner was tight-lipped about the specifics, he did state that the 'dispatcher' module is slated to be deprecated in favour of a "human load-balancing approach to SIP requests that is both cloud-native and eco-conscious". Industry analysts agree that protocol-aware load-balancing strategies are complicated and that the application of human judgment to the problem represents a fertile field for dynamic innovation and investment. Reports from regional press confirm that KMC Labs are in the process of building a fullfilment centre for this innovative offering, said to be branded "Central Switchboard", at an undisclosed site in Bangladesh, where thousands of "Load Balance Operator" jobs have been advertised in and around the Dhaka market. -- 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/

1 0

db_redis and user
by Joan Salvatella 01 Apr '19

01 Apr '19

Hello, We are working on a project that's using two active-active Kamailio 5.2 instances that redirect traffic to several Asterisk 15.4 instances. Kamailio is acting as a registrar and should store the location information of the registered terminals. The backend database is db_redis. Phone terminals get registered and the information gets looked up from the subscriber schema in redis (That's cool) but we are having problems with the locations. This is the related code: modparam("db_redis", "keys", "location=entry:ruid&usrdom:username,domain&timer:partition,keepalive") if(ds_is_from_list("1", "2", "sip:$fd:5080")) { record_route(); route(INVITE_ASTERISK); exit; } route[INVITE_ASTERISK] { xlog("L_INFO", "[R-INVITE-RESOLVER:$ci] Entering ASTERISK INVITE resolver\n"); if (!lookup("location")) { sl_send_reply("404", "User Not Found"); return; } route(INVITE_POSTROUTE); } route[INVITE_POSTROUTE] { rtpproxy_manage(); t_on_reply("MAIN_REPLY"); if(!t_relay()) sl_reply_error(); } When looking up the location schema this errors appear: 9(39) WARNING: db_redis [redis_dbase.c:1098]: db_redis_perform_query(): performing full table scan on table 'location' while performing query 9(39) WARNING: db_redis [redis_dbase.c:1101]: db_redis_perform_query(): scan key 0 is 'expires' 9(39) WARNING: db_redis [redis_dbase.c:1101]: db_redis_perform_query(): scan key 1 is 'expires' 9(39) ERROR: <core> [db_ut.c:92]: db_str2int(): Unexpected characters: [-03-29 18:16:36] 9(39) ERROR: <core> [db_val.c:76]: db_str2val(): error while converting integer value from string 9(39) ERROR: db_redis [redis_dbase.c:1062]: db_redis_convert_row(): Failed to convert redis column 'expires' to db value 9(39) ERROR: db_redis [redis_dbase.c:1231]: db_redis_perform_query(): Failed to convert redis reply for row 9(39) ERROR: db_redis [redis_dbase.c:1243]: db_redis_perform_query(): failed to perform the query 9(39) ERROR: db_redis [redis_dbase.c:1829]: db_redis_query(): failed to do the query 9(39) ERROR: usrloc [udomain.c:981]: udomain_contact_expired_cb(): db_query failed Any hints? Thanks and regards,

3 2

Kamailio v5.2 kamctl alias add problem
by Igor Potjevlesch 01 Apr '19

01 Apr '19

Hi kamailios, I meet a problem with the " kamctl alias add" command in the 5.2 version, I saw that the module used to add an alias had changed, and now the command result with an error : [kamailio]# kamctl alias add XXXX@XXXX sip:XXXXX@XXXX { "jsonrpc": "2.0", "error": { "code": 500, "message": "Not enough parameters or wrong format" }, "id": XXXXX } I read the function who return this message " static void ul_rpc_add(rpc_t* rpc, void* ctx)" in the ul_rpc.c in the usr_loc module. I tried to pass the command with two more parameters : # kamctl alias add 0175951068(a)trunk7.bornsip.fr sip:0175951068@trunk7.bornsip.fr NULL NULL { "jsonrpc": "2.0", "result": { }, "id": 29200 } And this time it works correctly, but I saw thaht the test prameters was in the field "path" , do you know why we have to indicate a path ? Are we obliged to indicate this two more parameters ? What is the utility of this field ? Thank you in advance, Igor

2 2

where to find the cfg languaje for program on it?
by PICCORO McKAY Lenz 01 Apr '19

01 Apr '19

recently i wrote a portion of my cfg file but i found that i based my code in other sources and results of try/tach experiments i cannot found the documents for the languaje of the core i only found the link: https://www.kamailio.org/wiki/cookbooks/devel/core but in that link there's no references about languaje per see only about how to write portions of the file and sections.. maybe i wrong or the document are wrote in different way? by example how to evaluate the output of a custom function/section? Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

2 1

drop before exit and send ok reply first can be done?
by PICCORO McKAY Lenz 01 Apr '19

01 Apr '19

h have this for a custom rule agains spoecific messages: sl_send_reply("200", "OK"); drop(); exit; my question are if that rules are correct: i want to drop but send firts to client a ok response.. i mean client receive "all are ok dont worry" and then drop the rest packets.. that's correct? Lenz McKAY Gerardo (PICCORO) http://qgqlochekone.blogspot.com

3 2

kamailio and rtpengine
by David Villasmil 01 Apr '19

01 Apr '19

Hello guys, i'm trying to configure rtpengine, but when starting kamailio (with an empty table) i'm getting ```build_rtpp_socks(): no more pkg memory for rtpp_socks``` Can anyone help? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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sr-users April 2019