Hello
I am using kamailio 5.1 for WebRTC (websocket)
But in the latest version of Google Chrome (85.0) I have a problem, No
sound is transmitted between the source and the destination.
This problem does not exist in Firefox browser
If you have experience in this field, please comment
Hi,
Kindly give me some guidelines on how to convert diameter messages to http
messages when establishing connection between PCSCF and PCF
Kindly help
Thanks,
Pavithra
Hi Everyone
As I am just dipping my foot in the VOIP waters so please bear with me.
Although it may be more appropriate for Freeswitch group I am sure there are
plenty of knowledgeable folks here.
I'm playing with freeswitch in my lab environment and really like what
Kamailio has got to offer. That being said, there are few concepts I need
help understanding.
Consider this ( for hosted services I suppose )
[image: Screenshot 2020-09-16 at 22.21.22.png]
1 Does this topology seem right ?
2 If Kamailio is used as registar and load balancer, does that mean all
domains, extensions must be present on all the FS boxes ?
3 How will auto provisioning work in this case ?
Now, in the below scenario, clients have got FS box deployed on prem
[image: Screenshot 2020-09-16 at 22.28.09.png]
4 Is there any reason to keep the FS farm ? or Kamailio poxying calls to
VOIP providers is enough ?
5 What roles Kamailio could be used for in this case ?
6 Is deploying on prem PBX for a customer a popular solution ?
7 Anyone deployed above on Docker/ Kubernetes ? How's the performance and
the bad bits if any ?
I do realize this is pretty basic stuff for everyone in here, however, I
would appreciate any helpful comments and if there is some
good literature/ links you can recommend, I am all ears.
Regards
Adam
Hi,
Is there any possibility of converting diameter messages to http requests
in IMS. Since I am working in 5g based network, In order to connect pcscf
to pcf, Rx interface has to be established. But PCF is not accepting
diameter messages. It is failing with error "No connected DefaultRoute peer
found for app_id 16777236 and vendor id 10415"
I used IMS Diameter server since it is mentioned as converting diameter
messages to http json messages. But even before calling that diameter
server, It is failing with App_id issue. How it works?
Need to understand, how the diameter messages are working since in 5g ,
everything is based on http rest apis. Is there any possibility of
connecting to it or there is no support in kamailio IMS for 5g. If you have
any idea, please let me know. It will be very useful for me.
Thanks,
Pavithra
Hello everyone.
I've started to have an issue with a kamailio 4.4 instance listening on
multiple network interfaces where it stops processing SIP on one of it's
network interfaces but still continues to work fine on the others. If I
restart kamailio everything starts working fine again.
I'm using the default 'children=8' and one of my guesses on why this could
be happening is that the number of childs processing couldn't be enough. Do
you remember anything else on an issue like this where I should be looking?
Thanks.
--
Nuno Miguel Reis
Departamento de Engenharia Informática
Faculdade de Ciências e Tecnologia
Universidade de Coimbra
I already run into issues with this. If I want to only execute a statement if it’s an INVITE with a request domain of 10.10.*.* how do I format? Below is what I have. I always have to play with it a bit. Looking to get a better understanding. The regular expression tester I’m using states that it’s valid. What am I doing wrong?
if ((is_method("INVITE")) && ($rd =~ "10\.10\..+")) {
# Do something
xlog("fix got a 10.10 address");
}
Thanks
Has anyone been down this path before? We are trying to test this out and the results are pretty promising so far.
I realize the lack of Stored Procedures and Triggers make this untenable for many Postgres based implementations.
Thanks!
~Noah
Hi all,
I have an infrastructure with Kamailio (reg), Asterisks farm, and Kamailio (4.3) proxy which dispatches calls.
Trunk-1 (Vlan_A)
Kam(reg) <> Asterisk Farm (5) <> Kam(proxy) <> Trunk-2 (Vlan_B)
Trunk-3 (Vlan_C)
For each trunk, I have to use the IP of the VLAN interface in "contact header". That means when I need to send a call to Trunk-1 I have to set Vlan_A IP address on the contact header. I have achieved that by rewriting Contact Header.
###
remove_hf("Contact:");
append_hf("Contact: <sip:$fU@$var(recieved)>\r\n", "Call-ID"); #by using db
###
But this causes a problem, Kamailio does not forward the BYE message that comes from Trunk-X to Asterisk (In before, the BYE message had Asterisk IP in R-URI). However, when I use topoh module it forward properly for one trunk. But I can not set multiple mask IP in topoh module.
I'm using the dialog module and I thought it can handle all requests in one dialog to forward to the correct asterisk but it did not.
Is there any suggestion for this case?
And also how many sub-interfaces can Kamailio handle for a good performance? Or do you suggest multiple Kamailio instances instead of multiple sub-interfaces on one server?
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Hello,
I am new in this VOIP field, yet I have to make a decision in using a SIP
proxy that would allow the configuration of multiple conference factories,
make load balancing when creating new conference, and provide presence
information related to active conferences.
Can you please tell me if this is possible with Kamailio, and give me some
reference to a module or some documentation that would relate to that ?
Thank you in advance.
Best regards,
Adrian Tabacioiu
Hi,
I'm rather new to Kamailio and RTPEngine, so I'm looking for some help in
locating any insights, documentation, guides or examples that resemble my
use case.
Currently I've got both running in working order and I can do SIP calls
with both video and audio. Now I'm trying to figure out how to pass through
BFCP as well. From what I understand so far, since BFCP isn't strictly RTP,
it's not something that's outright supported. Please correct me if my
assumption is wrong.
My observation is that "*m=application PORT {UDP,TCP}/BFCP **" SDP lines
doesn't seem to open for traffic as other media lines do. (Obviously
expected if it's not supported)
So I'm really just looking for any suggestions for, or where to start
looking, in getting BFCP support.
I do have the option of BFCP over both TCP and UDP, so if I could somehow
include the originator IP as an added attribute in the re-written SDP
answer, I could then just connect directly, around the proxy. I've yet to
figure out how to do that as well however.
Any pointers are much appreciated.
Regards,
René Hansen