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4 participants
37456 discussions

Calling uac_replace multiple times?
by Benoît Panizzon 17 Feb '23

17 Feb '23

Hey all I have started using the uac module to be able to mirror username changes on both legs of a call. But now I found an issue: Assume this situation: Call IC => Customer IC is e164, Customer wants 'local' numbers. IC From: +41991234567 => Customer From: 0991234567 uac_replace_from("sip:" + $avp(from_local) + "@" + $(fu{uri.host})); Customer is absent, after timeout in failure_route call is dispatched to Voicemail Server, which is configured for e164 numbers: uac_replace_from("sip:" + $avp(from_e164) + "@" + $(fu{uri.host})); But then uac complains: replace_uri(): Already called uac_replace for this dialog Why isn't just the most from version kept and used? -- Mit freundlichen Grüssen -Benoît Panizzon- @ HomeOffice und normal erreichbar -- I m p r o W a r e A G - Leiter Commerce Kunden ______________________________________________________ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 Pratteln Fax +41 61 826 93 01 Schweiz Web http://www.imp.ch ______________________________________________________

2 1

TLS version and cipher from WSS
by Michal Hajek 17 Feb '23

17 Feb '23

Hi, I can log incoming SIPS TLS version and cipher with PV $tls_version and $tls_cipher_info. How can I log these variables on incoming HTTP/WebSocket. Thanks Michal H.

2 2

Identify error in log
by Евгений Мастратов 16 Feb '23

16 Feb '23

Hello colleagues. I'm starting to use Kamailio for educational purposes, as an IMS test zone. Registration of subscribers, calls go successfully, but the following errors are constantly pouring into the log: Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/rvalue.c:1378]: int_intop2(): rv div by 0 Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/rvalue.c:1378]: int_intop2(): rv div by 0 Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/rvalue.c:2381]: rval_expr_eval(): rval expression evaluation failed (735,15-735,55) Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/rvalue.c:2381]: rval_expr_eval(): rval expression evaluation failed (735,15-735,55) Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/lvalue.c:386]: lval_assign(): rval expression evaluation failed (735,15-735,55) Feb 16 14:40:46 ct-pcscf /usr/local/sbin/kamailio[2849]: ERROR: <core> [core/lvalue.c:386]: lval_assign(): rval expression evaluation failed (735,15-735,55) Apparently they don’t carry anything critical, but they clog the log, natively I can’t understand the meaning of the errors. Please tell me what they mean and how to eliminate their cause? If necessary, I will provide the pcscf config.

2 1

uac reg reload frequency
by Dmytro M 16 Feb '23

16 Feb '23

When you do "kamcmd uac.reg_reload" frequently enough, you'll get an error: # kamcmd uac.reg_reload error: 500 - Failed to shift records - check log messages Logs: ERROR: uac [uac_reg.c:416]: uac_reg_ht_shift(): shifting in-memory table is not possible in less than 150 secs What is the reason behind 150 sec limitation for how frequent reloads can be done? Is it configurable in any way? Use case: Remote app needs to disable uac registration. It deletes record in DB and calls kamcmd uac.reg_reload to activate the change. With 150 sec limit it can't apply DB changes immediately. Please advise. Thanks, Dmytro

2 2

ACK on RE-Invite is send to UAC NATed Internal port instead of public port
by Angelo Sipper 15 Feb '23

15 Feb '23

Hi, My setup has: (TLS)UAC 77.1.1.1 -> (TLS)Kamailio 17.17.17.211 (UDP) -> (UDP)Kamailio 17.17.17.181 (UDP) -> Carrier(UDP) Both Kamailio are 4.3 with minor modifications on default cfg, and Topoh and TLS modules are used. I am having an issue on some outbound calls from nated UACs. On these calls our SBC sometimes receives reverse invites from the carrier side within the call. When this happens, my SBC(17.17.17.211) sends the invite to UAC and the UAC sends back 200 OK (SDP). Unfortunately my SBC(17.17.17.211) sends back the ACK not to the correct UAC public port but instead to the UAC internal port making the call to crash with several retries from my SBC to send 200 OK as ACK was never received from UAC! If you look the below traces of reverse invite past within the call, the reinvite has port private 20125 and alias the public port 2314 of UAC but the ACK bellow is send to 20125 and not to 2314 2023/01/26 09:48:30.668634 17.17.17.181:5060 -> 17.17.17.211:5060 INVITE sip:301234567890@77.1.1.1:20125;alias=77.1.1.1~2314~1 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Supported: timer Session-Expires: 600;refresher=uas Min-SE: 90 Content-Length: 297 Content-Disposition: session; handling=required Content-Type: application/sdp Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> v=0 o=Sonus_UAC 14218 13519 IN IP4 14.1.1.30 s=SIP Media Capabilities c=IN IP4 14.1.1.30 t=0 0 m=audio 18600 RTP/AVP 8 18 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:18601 2023/01/26 09:48:30.679743 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0;rport=5060 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Server: Supervoice Content-Length: 0 2023/01/26 09:48:30.740803 17.17.17.211:5060 -> 77.1.1.1:2314 INVITE sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Max-Forwards: 68 Allow: INVITE,ACK,CANCEL,BYE,INFO,PRACK,UPDATE,OPTIONS Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed Supported: timer Session-Expires: 600;refresher=uas Min-SE: 90 Content-Length: 297 Content-Disposition: session; handling=required Content-Type: application/sdp Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> v=0 o=Sonus_UAC 14218 13519 IN IP4 14.1.1.28 s=SIP Media Capabilities c=IN IP4 14.1.1.28 t=0 0 m=audio 13288 RTP/AVP 8 18 101 a=maxptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=sendrecv a=rtcp:13289 2023/01/26 09:48:30.893512 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Content-Length: 0 2023/01/26 09:48:31.026742 77.1.1.1:2314 -> 17.17.17.211:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.90b692fac0d1cc8a2ac460a67a02aec9.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890"<sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 259 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 77.1.1.1 s=3cxPS Audio call c=IN IP4 77.1.1.1 t=0 0 m=audio 9054 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=sendrecv 2023/01/26 09:48:31.093656 17.17.17.211:5060 -> 17.17.17.181:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.78276307977b5b2776919ce1f1e76d49.0 Require: timer Contact: <sip:301234567890@77.1.1.1:20125> To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 INVITE Session-Expires: 600;refresher=uas Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE Content-Type: application/sdp Supported: replaces, timer User-Agent: 3CXPhoneSystem 18.0.6.905 (905) Content-Length: 276 v=0 o=3cxPS 792019937001472 2393157908037634 IN IP4 14.1.1.28 s=3cxPS Audio call c=IN IP4 14.1.1.28 t=0 0 m=audio 13278 RTP/AVP 8 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=sendrecv a=rtcp:13279 2023/01/26 09:48:31.254591 17.17.17.181:5060 -> 17.17.17.211:5060 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.181;branch=z9hG4bK7b5d.48f49235d45a0f6f9efdad9d6d8224bd.0 From: <sip:300987654321@sip.mydomain.com>;tag=gK0e80b4b9 To: "301234567890" <sip:301234567890@sip.mydomain.com>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 69 Content-Length: 0 Route: <sip:17.17.17.211;lr;ftag=07ece97c;vsf=AAAAAFpUAgECBwAACnAGAW5FUF4ZXUdDUUhEX1hRUC5ldQ--;vst=AAAAAAAAAAAAAAAAAAAAAABCXkYAQkdeVEJDQVtSVBRQRTYw;did=5fc.669;nat=yes> Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> 2023/01/26 09:48:31.260901 17.17.17.211:5060 -> 77.1.1.1:20125 ACK sip:301234567890@77.1.1.1:20125 SIP/2.0 Via: SIP/2.0/UDP 17.17.17.211;branch=z9hG4bK7b5d.e8fb34e79623843d516ae77257014e2d.0 Via: SIP/2.0/UDP 17.17.17.181;rport=5060;branch=z9hG4bK7b5d.48f49235d45a0f6f9efdad9d6d8224bd.0 From: <sip:300987654321@17.17.17.211:5060>;tag=gK0e80b4b9 To: "301234567890" <sip:MyTrunk@77.1.1.1:20125>;tag=07ece97c Call-ID: 1q5nE37RGZ8a8ChI6rqShw.. CSeq: 29227 ACK Max-Forwards: 68 Content-Length: 0 Contact: <sip:atpsh-b5-62dfd3a3-cd5ed-1d354@17.17.17.181> Any help will be much appreciated. Thank you! Regards, Angelo

2 5

SCA Presence : Record-Route Filed Not Update in sca_subscriptions Table
by Vinayak Makwana 14 Feb '23

14 Feb '23

Hello all, I am using the sca module in kamailio script.so when i subscribe with my ip phone for SCA Presence i get all the information in a sca_subscriptions table for the first time. When I reboot the device I get a new record-route header in trace but It does not update *record-route *Filed in sca_subscriptions table. Can anyone help me out with this issue ? *Here's modparam configuration * modparam("sca", "db_update_interval", 10 ) modparam("sca", "db_url", DBURL_LOCAL) modparam( "sca", "line_seize_max_expires", 30) modparam( "sca", "purge_expired_interval", 1800) -- -- * <https://www.ecosmob.com/itexpo/> * *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.

3 8

SCA Feature:: Record-route query.
by satyaprakash ch 13 Feb '23

13 Feb '23

Hello I have a query regarding the record-route field update in DB for SCA SUBSCRIBE. Here are my test cases *Initial SCA subscription,* We got the below SUBSCRIBE packet in kamailio SUBSCRIBE sip:1234.xyz.com:5060 SIP/2.0 Via: SIP/2.0/UDP kam-ip:5060;branch=z9hG4bK2sansay1081381rdb66 *Record-Route* : <sip:sansay1081381rdb66spid17urcvjtr139807186744464nat0@kam-ip:5060;lr;transport=udp> To: <sip:372427@1234.xyz.com:6060> From: "708" <sip:372427@1234.xyz.com:6060>;tag=CFFBAF19-19A68CBC Call-ID: bd4b50d9384523c14ea05c532d895bf2 CSeq: 2 SUBSCRIBE Contact: < sip:372427@192.168.1.3> Expires: 3600 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER Event: call-info User-Agent: PolycomVVX Accept-Language: en Max-Forwards: 69 Content-Length: 0 also in the sca_subscriptions table record-route header is updated as per Initial SUBSCRIBE record-route. *record_route: <sip:sansay1081381rdb66spid17urcvjtr139807186744464nat0@kam-ip:5060;lr;transport=udp> * But After rebooting the device, we got a new record-route in SUBSCRIBE Packet but in DB Record-route Field not updated after getting an updated record-route in Trace. *Kamailio Trace *SUBSCRIBE sip:1234.xyz.com:5060 SIP/2.0 Via: SIP/2.0/UDP kam-ip:5060;branch=z9hG4bK2sansay1081415rdb48 *Record-Route*: <sip:sansay1081415rdb48spid17urcvjtr139807186744464nat0@kam-ip:5060;lr;transport=udp> To: <sip:372427@1234.xyz.com:6060> From: "708" <sip:372427@1234.xyz.com:6060>;tag=35852031-CF749E20 Call-ID: 41fa031d79e35fae4a61c61984895bf2 CSeq: 2 SUBSCRIBE Contact: < sip:372427@192.168.1.3> Expires: 3600 Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER Event: call-info User-Agent: PolycomVVX Accept-Language: en Max-Forwards: 69 Content-Length: 0 as per logic it should update the record-route field in sca_subscriptions table . but it didn't update it : record_route: <sip:sansay1081381rdb66spid17urcvjtr139807186744464nat0@kam-ip:5060;lr;transport=udp> So can you suggest to us how we can update the record-route field as per SUBSCRIBE's packet new record-route ?

2 1

Issues matching regex
by Nathan Stratton 11 Feb '23

11 Feb '23

I am having an issue matching toll-free. I tested my regex on regex101.com, and it looks good, but it does not match my if statement. Am I doing something wrong? if (method=="INVITE") { xlog("L_INFO", "Checking Routes $rU"); if ($rU=~"^(\+?1)?(8(00|44|55|66|77|88)[2-9]\d{6})$") { xlog("L_INFO", "Tollfree call"); }; }; Logs show "<script>: Checking Routes +18002211212" so I don't get why it does not match. ><> Nathan Stratton Founder, CTO Vocinity, Inc. <https://www.vocinity.com/gabbi>

2 1

kemi equivalent config
by mohsen khashei 11 Feb '23

11 Feb '23

2 1

KEMI Equivalent of script
by mohsen khashei 10 Feb '23

10 Feb '23

Hi I have an existing kamailio config and I want to migrate to kemi. I want to know what is the equivalent of this code in kemi javascript ? $avp(s:contact) = $hdr(Contact); $avp(s:contact_fix) = $hdr(Contact); avp_subst("$avp(s:contact_fix)", "/@([0-9]+).([0-9]+).([0-9]+).([0-9]+)/@$sel(cfg_get.kamailio.ss_ip)/gi"); remove_hf("Contact"); append_hf("Contact: $avp(s:contact_fix)\r\n"); t_on_reply("REG_FIX"); thanks for your attention.

1 0

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