Good afternoon
I have the following error in the application:
[rtjson_routing.c:701]: rtjson_update_branch(): no json for routing
This error occurs regardless of whether there is high demand or not and the application sending the data correctly.
Can you help me with this?
Felipe Nunes
Hi,
we want to execute ds_select from an xhttp:request event route. The idea
is to be able to get all active targets of a dispatcher set so our
monitoring can check that via http which would be convenient for us.
So are there any reasons why ds_select wouldn't work in an xhttp:request
even route?
Regards
Christian Berger
--
Christian Berger - berger(a)sipgate.de
Telefon: +49 (0)211-63 55 55-0
Telefax: +49 (0)211-63 55 55-22
sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois
Steuernummer: 106/5724/7147, Umsatzsteuer-ID: DE219349391
www.sipgate.de - www.sipgate.co.uk
Hello everyone!
Is there a way to define a listen port range for TCP/TLS? Or any macro I
can use to create the LISTEN commands? And finally is there a performance
penalty for having multiple (2k) listening ports at the same time?
I am in a strange situation where I need to register multiple credentials
on a single SIP trunk, for each registration I need to use a
different local port otherwise the trunk will simply overwrite the previous
registrations. Furthermore, when I place a call on that trunk I have to be
consistent with the port I used for registration for that
specific credential. So I am looking into explicitly defining the port to
be used per credential, UACREG allows me to use the contact_addr/socket for
the registration part and for placing of outbound calls I can use $fsn to
force the correct socket.
Best regards,
Joao
Hello,
I am using Kamailio pcscf with ipsec.
I have one ipsec listen address which is used by the UE to connect to the PCSCF and one other listen address on port 5060.
During the registration until the ipsec is not established, the UE sends the requests to the ipsec listen address and the pcscf is using the other listen address to forward the sip request to the ICSCF. That's is ok for me.
But when the ipsec is established after the 200 OK REGISTER, the UE sends a SUBSCRIBE but the pcscf forwards the SUBSCRIBE to the SCSCF using its ipsec listen address instead of the other listen address. I tried to use $fs to force the socket but without success.
Do you know how to force the source IP address to be used on the pcscf?
Thanks for your support.
Regards,
Anthony
Hello,
After upgraded from kamailio 5.7.5 to 5.7.6, when did start kamailio
process, got the following error:
WARNING: db_mysql [km_my_con.c:179]: db_mysql_new_connection(): opt_ssl_ca
option not supported by mysql version (value (null)) - ignoring
After downgrade to 5.7.5 dind't happened.
(we don't use mysql ssl connection).
Hello Kamailio Community,
I am currently working on a large scale VoIP deployment and would love to get insights on the best practices for scaling Kamailio. Our current setup handles a moderate volume of calls,,, but we’re expecting a significant increase in traffic soon. Specifically.., I am looking for advice on:
-Optimizing Kamailio’s configuration for high availability and performance.
-Efficiently managing load balancing and failover.
-Recommended hardware specifications for large-scale operations.
-Common pitfalls to avoid when scaling up.
Any tips or resources you could share would be greatly appreciated. If you have experience with similar projects,,, I’d love to hear about your setups and any challenges you faced.
Thank you!
Maria
https://www.igmguru.com/cloud-computing/aws-developer-certification-trainin…
Hello Kamailio Community,
I am currently working on a large scale VoIP deployment and would love to get insights on the best practices for scaling Kamailio. Our current setup handles a moderate volume of calls,,, but we’re expecting a significant increase in traffic soon. Specifically.., I am looking for advice on:
-Optimizing Kamailio’s configuration for high availability and performance.
-Efficiently managing load balancing and failover.
-Recommended hardware specifications for large-scale operations.
-Common pitfalls to avoid when scaling up.
Any tips or resources you could share would be greatly appreciated. If you have experience with similar projects,,, I’d love to hear about your setups and any challenges you faced.
Thank you!
Maria
<a href="https://www.igmguru.com/cloud-computing/aws-developer-certification-trainin…">aws</a>
Hi all
version 5.7.6 Debian 12
I'm having issues with a kamailio server not sending 200 to some OPTIONS
requests.
It does send 200 to some subscribers but not 100% of the time to 100% of subs.
I have enabled debug=3 in that server and this is what I found:
If the log says that it found existing tcp connection and it's reusing it I
can't see the 200ok in a sngrep capture. Example:
<core> [core/receive.c:126]: sip_check_fline(): first line indicates a SIP
reply
<core> [core/tcp_main.c:1722]: _tcpconn_find(): found connection by id: 2
<core> [core/tcp_main.c:2614]: tcpconn_send_put(): send from reader (360359
(18)), reusing fd
<core> [core/tcp_main.c:2845]: tcpconn_do_send(): sending...
<core> [core/tcp_main.c:2881]: tcpconn_do_send(): after real write: c=
0x7fb0faa893d8 n=384 fd=14
<core> [core/tcp_main.c:2882]: tcpconn_do_send(): buf=#012SIP/2.0 200
Keepalive#015#012Via: SIP/2.0/TCP
10.35.190.105:5060;branch=z9hG4bK15ebf764;rport=55894;received=1.1.1.186#015#012From:
"Unknown" <sip:contesta01@10.35.190.105>;tag=as5311c3a7#015#012To: <sip:
1.1.1.138>;tag=054524a74bcccf444a0f08f67c5b7657.d07b5401#015#012Call-ID:
15dcd38c46f915576ded132a11d8ff4c@10.35.190.105:5060#015#012CSeq: 102
OPTIONS#015#012Server: aaa SBC#015#012Content-Length: 0#015#012#015#012
<core> [core/receive.c:531]: receive_msg(): request-route executed in: 302 usec
On the other side, if kamailio doesn't find an existing tcp connection (at
least nothing in the logs about that) I can see the 200 ok in sngrep. Example:
<core> [core/receive.c:126]: sip_check_fline(): first line indicates a SIP reply
<core> [core/receive.c:531]: receive_msg(): request-route executed in: 295 usec
All subs use tcp. And the kamailio.cfg is very simple regarding OPTIONS:
Sanity, check maxfwd and sl_send_reply(200).
Any hints?
cheers,
Jon
--
PekePBX, the multitenant PBX solution
https://pekepbx.com
Hi everyone,
I'm trying to configure drouting module but i'm not sure how to configure dr_gateways table to use domain names and TLS instead of IPs
How should I do it?
In dispatcher module is by using sips:domain but when I configure it like this I have problems to send invites (maybe I'm doing something wrong).
Does anyone know how to configure it?
Thanks in advance for your help
Samuel Moya Tinoco
Departamento de Sistemas y Redes
Móvil: (+34) 606985997
smoya(a)vivelibre.es<mailto:smoya@vivelibre.es>
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