sr-users May 2021

sr-users@lists.kamailio.org

57 participants
65 discussions

by hamed r

hello dears ,i am using presence module and NOTIFY message are working properly. but when i put the database on another server , NOTIFY message will not be sent to subscribers. what is problem?

3 years, 5 months

handle_ruri_alias() and multiple aliases

by Denys Pozniak

Hello! I need help understanding how the *handle_ruri_alias()* function works. *Call-flow:* Upstream operator -> Kamailio Proxy -> Edpoint. The upstream operator in the initial SIP INVITE in the Contact field sends us an alias parameter, in turn, our Kamailio Proxy adds its own too. The Contact after the Proxy looks something like this: *Contact: <sip:10.0.0.115:5060;alias=10.0.0.115~5060~1;alias=3.3.3.3~5060~1>* When the Endpoint sends us a SIP BYE, then the *handle_ruri_alias()* absorbs not the last alias, but the first one, and this leads to the incorrect formation of a $du. The RURI after the Proxy looks something like this: *Request-Line: BYE sip:10.0.0.115:5060;alias=3.3.3.3~5060~1 SIP/2.0* The first thing that comes to mind is a modification of the original Contact field with storing it's within a dialogue ... Is it possible to elegantly solve this problem? -- BR, Denys Pozniak

3 years, 5 months

STIR/SHAKEN tests

by David Villasmil

Hello guys, I want to test secsipid, but i don't yet have the certificate. So i thought i'd create a cert like: openssl req -new -newkey rsa:4096 -nodes -keyout snakeoil.key -out snakeoil.csr openssl x509 -req -sha256 -days 365 -in snakeoil.csr -signkey snakeoil.key -out snakeoil.pem Then i'm simply doing: $var(rc) = secsipid_add_identity("$fU", "$rU", "A", "", " https://somedomain.com/stir/$rd/cert.pem <https://kamailio.org/stir/$rd/cert.pem>", "/etc/kamailio/snakeoil.pem"); if ( $var(rc) ) { xlog("L_ERR", "[STIR/SHAKEN][$ci] Shaken authentication added (SIP Identity Header created)\n"); } else { xlog("L_ERR", "[STIR/SHAKEN][$ci] Failed\n"); } But no matter what i do it silently fails: INVITE d54c2919-39b6-123a-95a7-0e29a5289b8d} <script>: [STIR/SHAKEN][d54c2919-39b6-123a-95a7-0e29a5289b8d] Failed I have debug on 6, but i don't get more info regarding the error. Any ideas? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

3 years, 5 months

Releasing Kamailio v5.4.6

by Daniel-Constantin Mierla

Hello, I am considering to release Kamailio v5.4.6 sometime next week, likely on Wednesday or Thursday (June 2 or 3, 2021). This is the usual heads up notification to see if anyone is aware of issues not yet reported to bug tracker and if yes, do it as soon as possible to give them a chance to be fixed. Cheers, Daniel -- Daniel-Constantin Mierla -- www.asipto.com www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - Online - June 7-10, 2021 (America Timezone) * https://www.asipto.com/sw/kamailio-advanced-training-online/

3 years, 5 months

IMS IPSec does use UDP socket?

by H Yavari

Hi to all, I am interested in the implementation of IPsec module. When I run Kamailio, it is creating processes based on the below formulate: UDP children * IPSEC_MAX_CONN * interfaces (v4 and v6) + (TCP+UD) processes for each interface. For example: children = 16tcp_children=16 IPSEC_MAX_CONN = 25and two interface : IPv6 and V4 Kamailio process count = (16 * 25 * 2 ) + 16 + 16 = 832 But why IPsec module is listening to UDP not TCP? this is an internal connection? Thank you. Regards,Hossein

3 years, 5 months

SDP body, add maxptime attribut

by Victor Velo

Hello, I would like to add the maxptime attribut to the sdp body. How and where should I do insertion? This kamailio is also using RTPEngine, so is it better to do the insertion using RTPEngine? And the source of the SDP is actually FreeSwitch :), maybe there to tweak? Of course, I don't know either way of doing it so please advise. Thanks, Victor

3 years, 5 months

New MariaDB server "limiting" Kamailio CPS

by Stephen Lokker

Good afternoon All, Im hoping for some advise or direction with the below. We are running a Kamailio SBC used for high call volumes and recently decided to upgrade the Database servers as the existing servers were struggling. Both the old and new databases are using MariaDB(version details all below), the problem we face is that since migrating to the new database there appears to a limitation to the amount of DB connections that can be processed or accepted. The SBC functions normally until the CPS starts to exceed around 100cps at which point the SBC stops responding to invites(in bursts which resembles flood prevention). We have checked both the Kamailio logs and the MySQL logs on the DB, both log files have no indication of errors that would cause the problem experienced(the logs attached are seen when the SBC is cut back to use the old DB, not sure they are related to the fault), the connections/processes in mysql do not drastically increase or get stuck either. The fault points to a db connection limit of some kind but all users have no limitations set. If anyone has experienced a similar problem or has any advise it would be greatly appreciated. Some info on the Kamialio and DB versions New DB root@voice-jhb-db01:~# mysql --version mysql Ver 15.1 Distrib 10.3.27-MariaDB, for debian-linux-gnu (x86_64) using readline 5.2 Old DB [root@isadb01 ~]# mysql --version mysql Ver 15.1 Distrib 10.0.36-MariaDB, for Linux (x86_64) using readline 5.1 Kamailio SBC [root@sbc01 ~]# kamailio -v version: kamailio 5.1.4 (x86_64/linux) Kind regards, Stephen Voice Engineer Tel: 2404 Fax: 3468 Email:

3 years, 5 months

How To setup Provider SIP trunk with registration

by zak ast

Hi, I know I can you uacreg to register a SIP trunk as a SIP client with username/password. But how can I set up a SIP trunk as a provider and get the other end (SIP client) to register with username/password. Regards, Zak.

3 years, 5 months

Architecture - your opinion is welcome and your experience too

by Daian Conrad

Hi all, I am studying and improving my understanding of how kamailio works, I have used asterisk for a few years(and starting learning freeswitch too), and I use an application to manage extensions, trunks and other media services. I'm building a new scenario where kamailio is facing the internet and the asterisk(s) are internally on private networks, would like the opinion of you who are more experienced with more complex environments than me. I cannot change the application that uses asterisk for now, due to the effort to develop event monitoring, cdr and other features that I use today. I imagined and working on making kamailio functional by saving the registration of extensions with location on DB, and with the UAC module making the registration of extensions on asterisk(s), replacing the registration address with the address of kamailio, its functional at this point, but data replication with extensions turns things some hard to mantain manually. I think about using the DNS domain for each asterisk and make this forwarded, each asterisk response for a fqdn and its extensions, like ( pbx1.example.com is forwarded to asterisk1, pbx2.example.com to asterisk2) and so on. Read about the dispatcher, rr, htable,carrierroute module to identify the domain and forward based on that , none made me sure to be chosen for the role, however, everyone has the resources to do it. I am very wrong to follow this path, which option in your opinion is "less" painful for a beginner apprentice like me? I think about simple proxy based on domain requests(all messages and dialogs) to asterisk where is responsible for the domain(realm) but, loss the ability to use great security features of kamailio, is what I understand now with the knowledge I have. Sorry and forgive my english, i'm not very good at writing. -- ================ **Daian Conrad** E-mail: daian.conrad(a)gmail.com OpenS Team (DaCoD) Linux user: #248912

3 years, 5 months

No Audio For Outbound Calls

by Kashish Raheja

Hi All, I have set up Kamailio in the following manner: Kamailio (Physical Server: Register to Telecom Operator Carrier SIP trunk) ---> Asterisk Server (on Cloud having public IP) I am successfully able to route the call to Asterisk server on Cloud when I make a call to the number provided by the carrier and there is audio also on both sides. However, when I am making an outbound call from Asterisk server to the number through Kamailio, there is no audio when I pick up the call. I have tried to capture the traces but not able to understand the exact problem here. Note: I am running the RTP proxy on Kamailio server. Any help on why this might be happening? Thanks. Regards Kashish +919413745250

3 years, 5 months

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sr-users May 2021