- sr-users - kamailio.org

Kamailio stops processing the calls - restart fixes it.
by Muhammad Danish Moosa 01 Mar '23

01 Mar '23

Hi, I have a very simple proxy (stateful) where kamailio acts as a proxy between 2 endpoints. Everything works fine for weeks and suddenly I see kamailio stops responding. From pcap I can see kamailio is not proxying the session progress and bombarding invites to one endpoint without any reason. Even that invite was stripped on the Body part. Restarting kamailio fixes it immediately. Unfortunately I could not take bt full yet. Version is kamailio 5.5.3 (x86_64/linux) 473cef configuration is very simple , routing is based on tm.t_relay ( based on KEMI). Any help will be welcome. Danish -- Muhammad Danish Moosa

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Kamailio v5.6.4 Released
by Daniel-Constantin Mierla 27 Feb '23

27 Feb '23

Hello, Kamailio SIP Server v5.6.4 stable release is out. This is a maintenance release of the latest stable branch, 5.6, that includes fixes since the release of v5.6.3. There is no change to database schema or configuration language structure that you have to do on previous installations of v5.6.x. Deployments running previous v5.6.x versions are strongly recommended to be upgraded to v5.6.4. For more details about version 5.6.4 (including links and guidelines to download the tarball or from GIT repository), visit: * https://www.kamailio.org/w/2023/02/kamailio-v5-6-4-released/ RPM, Debian/Ubuntu packages will be available soon as well. An additional note to say that call for speakers and registration for Kamailio World Conference 2023 are now open, more details at: * https://www.kamailioworld.com 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 - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

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Kamailio MS Teams TLS Issue
by iliusha.md＠gmail.com 27 Feb '23

27 Feb '23

Hello, We have one Kamailio Instance connected with MS Teams (based on this instruction: https://skalatan.de/en/blog/kamailio-sbc-teams), which worked fine for a while until recently we noticed that calls from teams are not working anymore. When I looked through the logs I found that Microsoft cannot establish a TLS connection to our server because of the cipher: TLS accept:error:1408A0C1:SSL routines:ssl3_get_client_hello:no shared cipher (sni: sbc.example.com - domain is obfuscated). Certificate is valid, the configuration is below: [server:default] method = TLSv1.2+ verify_certificate = no require_certificate = no private_key = /usr/local/etc/kamailio/certs/example.net/sbc1-teams_example_net.key certificate = /usr/local/etc/kamailio/certs/example.net/sbc1-teams_example_net.crt server_name = sbc1-teams.example.net ca_list = /usr/local/etc/kamailio/certs/sectigo_ca.pem #ca_list=/etc/ssl/certs/ca-bundle.crt [client:default] method = TLSv1.2+ verify_certificate = no require_certificate = no private_key = /usr/local/etc/kamailio/certs/example.net/sbc1-teams_example_net.key certificate = /usr/local/etc/kamailio/certs/example.net/sbc1-teams_example_net.crt ca_list = /usr/local/etc/kamailio/certs/sectigo_ca.pem #ca_list=/etc/ssl/certs/ca-bundle.crt We use a certificate from Sectigo, but I've tried with Let's Encrypt - and it's the same. Any idea what could be the reason?

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Releasing v5.6.4
by Daniel-Constantin Mierla 27 Feb '23

27 Feb '23

Hello, I am considering to release Kamailio v5.6.4 (out of branch 5.6) next week (likely on Monday or Tuesday, Feb 27/28, 2023). If anyone is aware of issues not yet on the bug tracker, report them there asap in order to have a better chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

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Kamailio RPC configuration
by sadik.oualla.mohamed＠gmail.com 24 Feb '23

24 Feb '23

Dear Kamailio community, I am a newbie in Kamailio and I am currently working on a project that involves communicating with Kamailio using Python. I have been exploring the Kamailio documentation and I have come across RPC, JSONRPCS..., which I believe can help me achieve my goal. However, I am not sure about the steps to follow in the Kamailio configuration file to create my own script to communicate with Kamailio using Python (the script is on an other machine on the same LAN). I would be grateful if someone could guide me through the process. I have already set up Kamailio (builed it from the source code 5.5.5) and Python on my machines and I have basic knowledge of Kamailio configuration. I just need some guidance on how to set up the RPC module in Kamailio and how to create a Python script to communicate with it. I found that I can use RPyC library of python. I would appreciate any help or suggestions from the Kamailio community. Thank you in advance for your time and assistance. Best regards, Wild Coder.

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OT: MS Teams source IP address ranges
by Jon Bonilla (Manwe) 24 Feb '23

24 Feb '23

Hi Sorry for the OT but I think here's the place where I an find a lot of Ms teams integrations I've been working on MS teams direct routing integration for PekePBX. It works. I guess I've done it as everybody else, using Henning's guide as base and extending it for multitenant setup (thanks Henning!) What I've realized is that the source IP address of calls coming from MS are not always matching dispatcher hosts. Sometimes they come from another source IP and failover to the dispatcher hosts when they receive no response. That makes some of the calls to have an additional latency Searching in the MS doc I see that they document these nets as source of their signaling: 52.112.0.0/14 52.120.0.0/14 But I've seen IP addresses outside of this range as source. In this blog https://erwinbierens.com/microsoft-teams-direct-routing-ip-addresses/ The ranges are listed as 52.112.0.0/16 52.113.0.0/16 52.114.0.0/16 52.115.0.0/16 52.120.0.0/16 52.121.0.0/16 52.122.0.0/16 52.123.0.0/16 which looks better but scares me out. Having no auth is it secure to bind so many ranges to MS? Do you use anything else than certificate verification for these calls? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

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Dispatcher failover when SRV failover is on?
by Jon Bonilla (Manwe) 23 Feb '23

23 Feb '23

Hi Again, fighting with MS Teams but this time it's a kamailio question. I have MSteams hosts in the dispatcher for OPTIONS and for destination. Acording to MS you need to try sip.pstnhub.foo then sip2.pstnhub.foo and sip3.pstnhub.foo 2 | sip:sip.pstnhub.microsoft.com;transport=tls | 0 | 1 2 | sip:sip2.pstnhub.microsoft.com;transport=tls | 0 | 2 2 | sip:sip3.pstnhub.microsoft.com;transport=tls | 0 | 3 I was going to implement dispatcher failover as usual ds_select_domain (alg 8) with failure routes but checking the SRV of the first domain I see it already includes 4 hosts: host -t srv _sips._tcp.sip.pstnhub.microsoft.com _sips._tcp.sip.pstnhub.microsoft.com has SRV record 10 0 5061 sip.pstnhub.microsoft.com. _sips._tcp.sip.pstnhub.microsoft.com has SRV record 20 0 5061 sip2.pstnhub.microsoft.com. _sips._tcp.sip.pstnhub.microsoft.com has SRV record 30 0 5061 sip3.pstnhub.microsoft.com. _sips._tcp.sip.pstnhub.microsoft.com has SRV record 40 0 5061 sip4.pstnhub.microsoft.com. So... If I have dns srv failover enabled in kamailio... what would be the point of the dispatcher failover? In case I only want failover in case 408 or 5XX wouldn't the tm module do it for me? cheers, Jon -- PekePBX, the multitenant PBX solution https://pekepbx.com

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What does l_username in UAC do?
by Дилян Палаузов 23 Feb '23

23 Feb '23

Hello, the uac module-registrations have the fields l_uuid - user for Contact: in REGISTER (before @). Described at https://kamailio.org/docs/db-tables/kamailio-db-5.6.x.html#idm9037 as “Local unique id used to build and match contact addresses.”. Described at https://www.kamailio.org/docs/modules/devel/modules/uac#idm700 as “local unique user id, e.g.,: 12345678” l_username - Described at https://kamailio.org/docs/db-tables/kamailio-db-5.6.x.html#idm9037 as “Local username”. Described at https://www.kamailio.org/docs/modules/devel/modules/uac#idm700 as “local user name, e.g.,: test ”. For me there is no difference between “local unique id” (as for l_uuid) and “local username“ (here). l_domain - used for Contact: in REGISTER (after @), described at https://kamailio.org/docs/db-tables/kamailio-db-5.6.x.html#idm9037 as “local domain” When is l_username used, what is it purpose? Does it solely serve as possible filter criterion for the RPC commands: uac.reg_info l_username abc uac.reg_enable l_username abc uac.reg_disable l_username abc, and uac.reg_unregister l_username abc Do I see correctly, that via UAC-RPC an unREGISTER can be triggered, but there is no way to trigger a REGISTER over UAC-RPC? Thanks for the clarifications Дилян

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change default 477 response code per case
by Patrick Karton 22 Feb '23

22 Feb '23

Hello , i would like to know if there is a way to change the default 477 response code generated by kamailio in some cases. for example if kamalio generates 477 because its not able to reach a peer i would like to change it to another code before to forward it. Thanks.

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Call History
by Anthony Wittig 21 Feb '23

21 Feb '23

Hello, I'd like to start tracking a call history/log as soon as I receive a call. I'd like the customer to be able to see that there's an incoming call to one of their numbers in real time. I'm thinking of using the RABBITMQ Module to manually emit an event from my main route when I receive an INVITE. Is that a bad idea? What if the broker isn't located on the same host (latency/availability)? I don't want to negatively affect the call processing. I was looking at the Dialog Module and it has three event routes: * event_route[dialog:start] * event_route[dialog:end] * event_route[dialog:failed] It's really close to what I want, but I'd like to capture the dialog before it starts (Unconfirmed or Early). Maybe another option is to scan the Dialog table with an external process and emit events from there. Has anyone had success in creating a realtime call history? Is there a better way to go about doing so? Thanks! -- Anthony Wittig Mango Voice Developer

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Info: Registration to Kamailio World 2023 is open!
by Daniel-Constantin Mierla 21 Feb '23

21 Feb '23

Hello, a quick note to the community forums to announce that the registration for the next Kamailio World Conference (June 5-7, 2022, in Berlin, Germany) is now open! The event returns to an in-person conference format, at the same wonderful location in the city center of Berlin! More details are available at: * https://www.kamailioworld.com/k2023/registration/ Call for presentations is also open, there is already a group of very interesting submissions, we aim to publish details about them in the near future. Looking forward to meeting many of you in Berlin! Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio World Conference - June 5-7, 2023 - www.kamailioworld.com Kamailio Advanced Training - Online - March 27-30, 2023 - www.asipto.com

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topos breaking route set on hairpin calls?
by Benoît Panizzon 20 Feb '23

20 Feb '23

Hi Having this situation while elaborating topos in lab. * Kamailio Registration Instance with topos enabled where customers are registered. * Kamailio Core Instance with Dialogue enabled and handling interconnections. Calls in transit from customer to IC work flawlessly, topos does what it should, hides and restores headers needed on the core. But I have run into an issue with customer calling another customer on same registrar instance. To generate a CDR with diaglogue and for decision where to route the calls to (registrar is kept simple and only handles registrations and locations), those calls are also routed via core. This works as long as PRACK is disabled. When PRACK is enabled and CPE B is requesting 100rel, the PRACK send from CPE A has no route set (it was removed by topos). Therefore the PRACK only makes it to the core who then does not know where to route that PRACK. It looks like topos on the REG only restores ONE Route pointing to the core (maybe because the 2nd one pointed to itself?) header when restoring them. I have attempted to spam more record_route() here and there in the config of the core and the registrar to try to force creation of route headers, but failed. Is this a known limitation of using topos? -- 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 ______________________________________________________

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Re: OT: MS Teams source IP address ranges
by Henning Westerholt 20 Feb '23

20 Feb '23

Hi Jon, you could of course try to add some heuristic, in the end the used IPs are a smaller sub-set. But to just quote from the official docs, it seems to be pretty clear: "You need to open ports for all these IP address ranges in your firewall to allow incoming and outgoing traffic to and from the addresses for signaling." Cheers, Henning ________________________________ From: Jon Bonilla (Manwe) Sent: Monday, February 20, 2023 15:49 To: Henning Westerholt; Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] OT: MS Teams source IP address ranges El Mon, 20 Feb 2023 13:06:30 +0000 Henning Westerholt <hw(a)gilawa.com> escribió: > Hello, > > correct me if I am wrong, but the second group of addresses ("/16") is > included in the first group ("/14"), right? > > Regarding the large IP scope, this is the way Microsoft designed it, > unfortunately. There is not that much what you can do about. > > Right. Didn't check the masks. they all were 16 in my mind :) So, if I add those ranges in the permissions table my only way of avoiding spoofed calls is certificate validation? I'm already doing that and also checking the company domain and ms account email address. Just wanted to be sure that nothing "essential" was missing. -- PekePBX, the multitenant PBX solution https://pekepbx.com

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sendig 4 byte udp nat pings
by Bernd Krueger-Knauber 18 Feb '23

18 Feb '23

Hi, maybe I'm stupid, but I can not find out how to send 4 byte UDP NAT packets to the sip server. loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 15) modparam("nathelper|registrar", "received_avp", "$avp(s:rcv)") I'm a absolute beginner with kamailio and the page about the nathelper Module is not really helpful. At the moment no pings are sendet. But I receive 4 byte packets from the sip server. After 15 minutes my calls are cancelled because of the missing pings. Without proxy, the phone send the pings directly to the sip server and the call stays alive. When I experiment with some other nathelper settings, kamailo start results in errors. Any help would be appreciated. Best regards, Bernd

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enable topology hiding on one side or dynamic ip in topoh module Contact and Via Header
by Patrick Karton 17 Feb '23

17 Feb '23

Hello , im using kamailio with two interfaces external and internal. i need a way either to : * enable topoh hiding only when outgoing interface is external ( mask contact and Via ip only when ougoing interface is external * or if it is not possible to enable it only in one direction.i want to know how to configure dynamic(for example in xavp or avp) ip to put in Contact and Via when topoh is enabled. i see that the 'mask_ip' parameter of topoh module is a string. so we can not set a dynamic value here unfortunately. Thanks

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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.

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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.

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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

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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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