sr-users May 2021

sr-users@lists.kamailio.org

57 participants
69 discussions

New MariaDB server "limiting" Kamailio CPS
by Stephen Lokker 01 Jun '21

01 Jun '21

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:

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How To setup Provider SIP trunk with registration
by zak ast 31 May '21

31 May '21

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.

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Architecture - your opinion is welcome and your experience too
by Daian Conrad 28 May '21

28 May '21

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

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No Audio For Outbound Calls
by Kashish Raheja 28 May '21

28 May '21

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

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kamailio: split initialized call to two (three or four or five) numbers
by Фёдор Измайлов 28 May '21

28 May '21

Hello, I have question regarding of call forking on kamailio. Maybe somebody can help me. For example I have table like +--------------+---------------+--------------+---------------+--------------+ | First number | Second number | Third number | Fourth number | Fifth number | +--------------+---------------+--------------+---------------+--------------+ | A number | B number | C number | --- | --- | +--------------+---------------+--------------+---------------+--------------+ Table 1: Extra_numbers or separated by comma +--------------------+-----------------------------+ | First_number | Additional_numbers | +--------------------+-----------------------------+ | Primary number | 2nd,3rd number | +--------------+-----------------------------------+ Table 2: Extra_numbers +--------------------+---------------------------------+ | First_number | Additional_numbers | +--------------------+---------------------------------+ | Primary number | 2nd number,3rd number,4 number | +--------------------+---------------------------------+ Table 3: Extra_numbers If kamailio get the call (INVITE) to some number/extension (for example, "1st number"), then in routing I request (select `Additional_numbers` from `Extra_numbers` where First_number='1 number';) from the mysql database and there I check, if "Additional_numbers" exists in the database. Then I add "Additional_numbers" to the routing, and generate INVITE to "Additional_numbers" and 1 number. (always) set r-uri ($ru) to first destination (2 number) set r-uri ($ru) to second destination (3 number) set r-uri ($ru) to Third destination (1 number) set r-uri ($ru) to n destination How could this be done? Who can help with routing? -- Best regards, Fjodor Izmajlov

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Setting the prefix in Json formatted logs
by Abdirahman Osman 27 May '21

27 May '21

Hello everyone, Tried to use the log-engine to push json formatted logs, here is a what kamailio is sending, my question is, How can I set the kamailio prefix? (232) { "kamailio.level": "DEBUG", "kamailio.module": "tm", "kamailio.file": "timer.c", "kamailio.line": 634, "kamailio.function": "wait_handler", "kamailio.callid": "", "kamailio.logprefix": "", "message": "finished transaction: 0x7f232236b1c0 (p:0x7f232216ae58/n:0x7f232216ae58)" } Thank you! Abdirahman

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kamailio: split initialized call to two (three or four or five) numbers
by Фёдор Измайлов 27 May '21

27 May '21

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Body formats for filter_body()
by Duarte Rocha 27 May '21

27 May '21

Hello all, A couple years ago the function filter_body() from textops would only work with "application/sdp" if the string on the body was "Content-Type: application/sdp\r\n". It would fail if there wasn't a space after ":" and it would fail if there was present "\r\n\r\n" instead of "\r\n". Is this the actual behaviour of the function or has it been changed to also identify SDP in the formats mentioned above? Best Regards,

2 1

Collecting stats from Kamailio.cfg using Kamailio_exporter Prometheus
by Hisham Ahmad Jan 26 May '21

26 May '21

Hi, I'm using Kamailio version 5.3.7 with DISPATCHER module and I would like to integrate my kamailio with a kamailio_exporter for monitoring using Prometheus server. Is it possible to collect data such as the number of total calls (inbound, outbound) and plot time-graphs using those stats? If so, what modules/functions would I need apart from the Dispatcher module to be able to have efficient results? Any degree of help will be appreciated. *Hisham Jan* *VoIP Engineer* *0322-6398775*

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not able to display number in SCA module
by Aatif Shaikh 26 May '21

26 May '21

Hello All, I am using sca module to handle SLA presence and I am using sca.update_appearance RPC command to update the display name. kamcmd sca.update_appearance sip:215@voice.example.com 3 held KELVIN after executing the above command I am getting the KELVIN name on my Polycom phone display but I want to also update the display number so is there any way to update the display number for SLA presence using the SCA module

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sr-users May 2021