sr-users August 2020

sr-users@lists.kamailio.org

63 participants
82 discussions

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.

4 10

Is Min-SE check broken?
by harneet singh 31 May '23

31 May '23

Hi All, I am facing an issue in understanding how the min_se should be working in kamailio. As per the SST documentation, it seems like if the min_se is configured as 500, then any value of Session-Expires OR MIN-SE if lower than 500, can be responded to by a 422. However, I strangely see the reverse happening. To investigate further, I looked in to the ki_sst_check_min() code in the master, and these seems like a potential issue. Ref Code: Inside ki_sst_check_min(), there is an if condition like below: if (sst_min_se < MIN(minse, se.interval)) { However, shouldn't it be the other way around? ie if (sst_min_se > MIN(minse, se.interval)) { because we need to send 422 if the received value(in INVITE etc) is smaller than the sst configure min_se value? I also found a different documentation, at https://git.sgu.ru/oldssu/ex-opensips/blob/cb9df8d59dbb254a9d862569fd5d11f6… where the check is as below? if (sst_min_se > MIN(minse, se.interval)) { Can someone confirm if this is broken, or my understanding is incorrect? Regards, Harneet -- "Once you eliminate the impossible, whatever remains, no matter how improbable, must be the truth" - Sir Arthur Conan Doyle

2 3

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

REFER from MS teams to Kamailio.
by Yuriy Nasida 04 Dec '20

04 Dec '20

Well, I saw similar questions in the list already but looks like nobody has answer. Please look at REFER below. Kamilio get REFER from MS and sends it to FS node. Next, FS node try to make 3th call for some reason.I expect that FS will not do 3th call and just will connect Alice and Bob itself. 2020/05/14 12:32:00.637027 KAM_IP:5060 -> FS_IP:5060 REFER sip:Alice_number@FS_IP:5060;transport=udp SIP/2.0 FROM: Customer1<sip:MS_TRUNK_NUMBER@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=a860c50a3fb54d08b4e5740fa2dfb3d6 TO: <sip:Alice_number@FQDN_OF_TRUNK:5061>;user=phone;tag=e8ct9S6ty13va CSEQ: 4 REFER CALL-ID: 2c71b2a6669b5343a231e1244b19c945 MAX-FORWARDS: 50 Via: SIP/2.0/UDP FQDN_OF_TRUNK:5060;branch=z9hG4bK10ae.2c42897feca117121a23bf0c8d54cd19.0;i=c VIA: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bK7e3e8998 CONTACT: <sip:api-du-a-euwe.pstnhub.microsoft.com:443 ;x-i=6b68e7aa-f5e2-44ec-9edf-0bacbabfce07;x-c=2c71b2a6669b5343a231e1244b19c945/d/8/b68f86794a8e44d19543f8edbee6b2fc CONTENT-LENGTH: 0 REFER-TO: <sip:Bob_number@sip.pstnhub.microsoft.com:5061 ;user=phone;transport=tls> REFERRED-BY: <sip:sip.pstnhub.microsoft.com:5061 ;x-m=8:orgid:21bc47d3-c050-4292-8234-46f7005b97aa;x-t=fb788ef8-3c4c-455a-8d62-f3c20832c0d3;x-ti=6b68e7aa-f5e2-44ec-9edf- acbabfce07;x-tt=aHR0cHM6Ly9hcGktZHUtYS1ldXdlLnBzdG5odWIubWljcm9zb2Z0LmNvbS92MS9uZ2MvY2FsbG5vdGlmaWNhdGlvbj9kY2k9YzIxMjE3MzEyNTQ2NDk1ZjlhYTcwODliYTkwNGIxZGQ%3D> USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.5.6.2 i.EUWE.4 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY P-ASSERTED-IDENTITY: <tel:MS_TRUNK_NUMBER>,< sip:customer1@m365x587912.onmicrosoft.com> PRIVACY: id X-AUTH-IP: 52.114.75.24 X-AUTH-PORT: 3136 Any advice?

4 11

Add Reason Header to kamailio generated CANCEL
by Alfonso Pinto 18 Nov '20

18 Nov '20

Hi everyone, I've got a specific case: when the inv_fr times out, I need to add a Reason header to the CANCEL generated by kamailio. I've tried to see if I could do it in the onsend_route, but that one is not triggered for the generated CANCEL. I also checked event_route[tm:local-request], but that one isn't triggered either for the generated CANCEL. Is there any way to do it? Or maybe to have any pointer about where to look in the code so I may try to trigger event_route[tm:local-request] for these generated CANCELs? Regards, Alfonso

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Rewrite BYE to Cancel
by Lars Olsson 29 Sep '20

29 Sep '20

hi, I have a Kamailio setup infront of a SIP system that do not handle cancellation of a INVITE correctly. The system sends out a BYE request instead of a Cancel request on non connected dialogs. I am trying to find a way to let Kamailio "translate" the BYE request to a Cancel reqeust for the ongoing INVITE dialog. Alternative if SEMS b2bua can do it, but currently it replies: "not sip-relaying BYE in not connected dlg", and I have not found any obvious way to rewrite it there. Any thoughts. I can not change the behavior of the remote system. Best Regards, Lars

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Dialog - timeout for dlg with CallID
by Ilie Soltanici 28 Sep '20

28 Sep '20

Hello, I'm trying to get CDR working in Kamailio by using the acc and dialog modules. Everything seemed to be working fine - until i noticed that for some of the calls the call duration is 0, even if that call has been successfully established and duration was for about a few minutes. In the Kamailio logs I'm getting such errors: WARNING: dialog [dlg_handlers.c:1649]: dlg_ontimeout(): timeout for dlg with CallID '304bad142b50bb3a7a117816439ea3d5' and tags 'as3adde5c7' '7d28152f-e0e3-4bcf-9d5c-21c3723b95c5' WARNING: acc [acc_cdr.c:230]: db_write_cdr(): fallback to dlg_only search because of message doesn't exist. This error I'm getting at about 2 min after the ACK message for 200 OK. I'm not sure that this is related to the dialog timeout, but below you can see the related configuration for the dialog module: modparam("dialog", "default_timeout", 10800) # 3 hours modparam("dialog", "early_timeout", 180) modparam("dialog", "noack_timeout", 90) Unfortunately, I'm not able to reproduce this issue, as that's happening randomly and just a few times per day. On the SIP Level i didn't notice any strange issues. Any ideas why is that happening? Thank you.

6 34

Kamailio drop calls with Teams
by sip user 22 Sep '20

22 Sep '20

Hi, I'm connecting Teams with kamailio server. From Kamailio to teams I have no problems, but from teams to Kamailio yes. Drop the call.. With ngrep I see that: INVITE sip:1005@CLIENT_IP:55766;transport=UDP;rinstance=d47edf336726e940 SIP/2.0. Record-Route: <sip:FQND_IP;r2=on;lr>. Record-Route: <sip:FQND_IP:5061;transport=tls;r2=on;lr>. FROM: "Javier Gonz..lez Mu..oz"<sip:+324@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=c17bb1eb7f8649d4a89d8d4a876ac32b. TO: <sip:+34560@FQND:5061;user=phone>. CSEQ: 1 INVITE. CALL-ID: c1364913e582553a9a9c2544c3583b0a. MAX-FORWARDS: 69. Via: SIP/2.0/UDP 92.222.217.64;branch=z9hG4bK7bb5.0fbae76a37829205e04279f220a26af2.0;i=a1. VIA: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bKd216a55. RECORD-ROUTE: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>. CONTACT: <sip:api-du-a-euno.pstnhub.microsoft.com:443 ;x-i=b0b53fc5-76ef-4619-9a68-13e0a4eea92d;x-c=c1364913e582553a9a9c2544c3583b0a/d/8/6c25eb3789a14bf188ce6b05b5e27891>. CONTENT-LENGTH: 1091. MIN-SE: 300. SUPPORTED: timer. USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.7.31.1 i.EUNO.0. CONTENT-TYPE: application/sdp. ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY. P-ASSERTED-IDENTITY: <tel:+324>,<sip:EMAIL>. PRIVACY: id. SESSION-EXPIRES: 3600. . v=0. o=- 165103 0 IN IP4 127.0.0.1. s=session. c=IN IP4 52.113.44.8. b=CT:10000000. t=0 0. m=audio 50452 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118. c=IN IP4 52.113.44.8. a=rtcp:50453. a=ice-ufrag:FZTb. a=ice-pwd:yD3P7nr+xNq0VYdv7xpB1F+Y. a=rtcp-mux. a=candidate:1 1 UDP 2130706431 52.113.44.8 50452 typ srflx raddr 10.0.33.240 rport 50 U CLIENT_IP:55766 -> FQND_IP:5060 #2 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP FQND_IP;branch=z9hG4bK7bb5.0fbae76a37829205e04279f220a26af2.0;i=a1. Via: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bKd216a55. Record-Route: <sip:FQND_IP;lr;r2=on>. Record-Route: <sip:FQND_IP:5061;transport=tls;r2=on;lr>. Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>. Contact: <sip:1005@CLIENT_IP :55766;transport=UDP;rinstance=d47edf336726e940>. To: <sip:+34560@FQND:5061;user=phone>;tag=de4e6b45. From: "Javier Gonz..lez Mu..oz"<sip:+324@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=c17bb1eb7f8649d4a89d8d4a876ac32b. Call-ID: c1364913e582553a9a9c2544c3583b0a. CSeq: 1 INVITE. User-Agent: 3CXPhone 6.0.26523.0. Content-Length: 0. U CLIENT_IP:55766 -> FQND_IP:5060 #3 SIP/2.0 200 OK. Via: SIP/2.0/UDP FQND_IP;branch=z9hG4bK7bb5.0fbae76a37829205e04279f220a26af2.0;i=a1. Via: SIP/2.0/TLS 52.114.76.76:5061;branch=z9hG4bKd216a55. Record-Route: <sip:FQND_IP;lr;r2=on>. Record-Route: <sip:FQND_IP:5061;transport=tls;r2=on;lr>. Record-Route: <sip:sip-du-a-eu.pstnhub.microsoft.com:5061;transport=tls;lr>. Contact: <sip:1005@CLIENT_IP :55766;transport=UDP;rinstance=d47edf336726e940>. To: <sip:+34560@FQND:5061;user=phone>;tag=de4e6b45. From: "Javier Gonz..lez Mu..oz"<sip:+324@sip.pstnhub.microsoft.com:5061 ;user=phone>;tag=c17bb1eb7f8649d4a89d8d4a876ac32b. Call-ID: c1364913e582553a9a9c2544c3583b0a. CSeq: 1 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE. Content-Type: application/sdp. Supported: replaces. User-Agent: 3CXPhone 6.0.26523.0. Content-Length: 1067. . v=0. o=3cxVCE 324945090 117647850 IN IP4 . s=3cxVCE Audio Call. t=0 0. m=audio 0 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118. c=IN IP4 52.113.44.8. a=rtpmap:104 SILK/16000. a=rtpmap:9 G722/8000. a=rtpmap:103 SILK/8000. a=rtpmap:111 SIREN/16000. a=fmtp:111 bitrate=16000. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:97 RED/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=rtpmap:13 CN/8000. a=rtpmap:118 CN/16000. a=rtcp:50453. a=ice-ufrag:FZTb. a=ice-pwd:yD3P7nr+xNq0VYdv7xpB1F+Y. a=rtcp-mux. a=candidate:1 1 UDP 213 I never received ACK.. In my configuration: Kamailio.cfg: #!KAMAILIO #!define WITH_TLS event_route[tm:local-request] { if(is_method("OPTIONS") && $ru =~ "pstnhub.microsoft.com") { append_hf("Contact: <sip:FQND:5061;transport=tls>\r\n"); } xlog("L_INFO", "Sent out tm request: $mb\n"); } request_route{ remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) { xlog("L_INFO","$fU is trying to call to $rU con valores $tu\n"); $rU="1005"; } } What I'm doing wrong? I don't understand why not received ACK.. Could anyone help me? Thanks

5 21

CockroachDB and Kamailio
by Noah Mehl 16 Sep '20

16 Sep '20

Has anyone been down this path before? We are trying to test this out and the results are pretty promising so far. I realize the lack of Stored Procedures and Triggers make this untenable for many Postgres based implementations. Thanks! ~Noah

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sr-users August 2020