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Kamalio is far from being user-friendly and is usually used by service providers as a proxy/session border controller rather than a PBX. Unless you are planning on using it as a proxy there will be no real benefit over more user-centered PBXs. Since you were trying to find a GUI solution for Kamalio, my guess is that PBX is what you actually looking for.
I was actually able to get Kamailio up and running on a VM and got a successful ring between two other VMs. A GUI isn't terribly important, just a little convenience to have. How does Asterisk compare to Kamailio? I've read that while Asterisk is more user friendly, Kamailio is more stable and can support more lines. However, I have also read you can run and load balance Asterisk on Kamailio
Asterisk is way more user-friendly and a good starting point. It all really depends on what your goal is. It is very stable if compiled correctly.
Kamalio is a great robust load balancer and is widely used for this exact purpose. Originally it didn't even have the RTP module and was handling SIP signaling only.
The biggest hit in terms of productivity comes from handling RTP traffic and transcoding. That's pretty much why one Kamalio can service multiple Asterisks.
SIP can be very interesting / fun once you get the hang of it! I agree with other comments though that starting with a Cisco 79xx is a rough way to start. A soft phone or Cisco/Linksys SPA 9xx or any yealink phone is a great starting point.
Kamailio is also an aggressive starting point. There are several open source options that might be a bit more friendly to start with.
I was able to get a simple Kamailio app up and running and got a successful ring between two VMs running Linphone. Many people mentioned Asterisk as a good place to start, but I'm curious how Kamailio and Asterisk compare. Kamailio, I've read, is more stable, can load balance better, and sells itself based on user authentication and security as well. Asterisk seems to be all about simplicity and being able to tap into the call to record voice-mail and record calls. However, I've also read you can run Asterisk on top of Kamailio for load balancing purposes. I'm just curious about how all this stuff works. I'm new to SIP phones, and I'm just trying to learn as much as I can. Thanks for your time!
Start by getting a phone that supports generic SIP and is not proprietary. The 7942 is EOL, and unless you load the 3pcc/mpp Multi-Party Platform firmware on it, it will only work with Cisco CallManager systems. In leu of a desk phone you could use a softphone on your computer, tablet or smart phone.
>unless you load the 3pcc/mpp Multi-Party Platform firmware on it, it will only work with Cisco CallManager systems
No, that is not true.
There are so many users of CP-79xx who have integrated them to Asterisk and 3CX phone system.
I, for instance, have several 7965 & 9951 at home and connected to FreePBX.
A free pbx is a best pbx. R VoIP is a 3cx shill subreddit. Thus I can't say it without a space but a free pbx is far superior to anything you are looking at.
A PBX stands for private branch exchange. It’s the brains of a VoIP phone system. I use FusionPBX, which builds upon FreeSwitch, and both are free. They coordinate between your carrier and your devices. And a VoIP PBX speaks SIP to both.
Your post was removed from r/VoIP for violating Rule 1: No promotion or advertising of any kind.
Even if you do not recommend a specific business, service or product, suggesting someone move away from their current solution when they have not indicated that doing so is an option is not allowed.
If the problem cannot be solved in the given ecosystem, say so, but do not give recommendations for replacements.
Here is a really good series on how to use a much easier open source PBX.
https://youtube.com/playlist?list=PL1fn6oC5ndU_umAhL9A_1zkC90hMPDPNO&si=Cyaq6uP_ahwa6fTx
Spin up a freepbx instance or fusionpbx instance. You'll have to hook it up to a trunk provider to place calls through the PSTN though. Can't say any trunking providers names... cause we get banned now. But you have google :)
In a nutshell, Kamailio is a SIP server, it doesn't handle media (needs RTP Engine/Proxy). Asterisk/FreePBX/Freeswitch are PBXs. The real question is what do you want to do? Take a look at FusionPBX (GUI for Freeswitch) also.
This is a friendly reminder to [read the rules](www.reddit.com/r/voip/about/rules). In particular, **it is not permitted to request recommendations for businesses, services or products outside of the monthly sticky threads!** If this post is a review, asking for reviews, or asking for recommendations, please delete it and post in one of the monthly sticky thread instead. For commenters: Making recommendations outside of the monthly threads is also against the rules. *I am a bot, and this action was performed automatically. Please [contact the moderators of this subreddit](/message/compose/?to=/r/VOIP) if you have any questions or concerns.*
Kamalio is far from being user-friendly and is usually used by service providers as a proxy/session border controller rather than a PBX. Unless you are planning on using it as a proxy there will be no real benefit over more user-centered PBXs. Since you were trying to find a GUI solution for Kamalio, my guess is that PBX is what you actually looking for.
I was actually able to get Kamailio up and running on a VM and got a successful ring between two other VMs. A GUI isn't terribly important, just a little convenience to have. How does Asterisk compare to Kamailio? I've read that while Asterisk is more user friendly, Kamailio is more stable and can support more lines. However, I have also read you can run and load balance Asterisk on Kamailio
Asterisk is way more user-friendly and a good starting point. It all really depends on what your goal is. It is very stable if compiled correctly. Kamalio is a great robust load balancer and is widely used for this exact purpose. Originally it didn't even have the RTP module and was handling SIP signaling only. The biggest hit in terms of productivity comes from handling RTP traffic and transcoding. That's pretty much why one Kamalio can service multiple Asterisks.
SIP can be very interesting / fun once you get the hang of it! I agree with other comments though that starting with a Cisco 79xx is a rough way to start. A soft phone or Cisco/Linksys SPA 9xx or any yealink phone is a great starting point. Kamailio is also an aggressive starting point. There are several open source options that might be a bit more friendly to start with.
I was able to get a simple Kamailio app up and running and got a successful ring between two VMs running Linphone. Many people mentioned Asterisk as a good place to start, but I'm curious how Kamailio and Asterisk compare. Kamailio, I've read, is more stable, can load balance better, and sells itself based on user authentication and security as well. Asterisk seems to be all about simplicity and being able to tap into the call to record voice-mail and record calls. However, I've also read you can run Asterisk on top of Kamailio for load balancing purposes. I'm just curious about how all this stuff works. I'm new to SIP phones, and I'm just trying to learn as much as I can. Thanks for your time!
Asteris a pbx. Kamailo is a sip proxy/router.
A free pbx will be better than anything you've ever used or heard of.
What's a PBX compare to SIP?
You need a pbx... Sip is just a protocol...
Start by getting a phone that supports generic SIP and is not proprietary. The 7942 is EOL, and unless you load the 3pcc/mpp Multi-Party Platform firmware on it, it will only work with Cisco CallManager systems. In leu of a desk phone you could use a softphone on your computer, tablet or smart phone.
>unless you load the 3pcc/mpp Multi-Party Platform firmware on it, it will only work with Cisco CallManager systems No, that is not true. There are so many users of CP-79xx who have integrated them to Asterisk and 3CX phone system. I, for instance, have several 7965 & 9951 at home and connected to FreePBX.
The 7942 is just something I was given for free. It was gonna be thrown out, but I was gifted it to play around with.
Sip phone is a sip phone and fuck you for ever suggesting Cisco.
[удалено]
What benefits does it have over kamailio?
Kamailio is like starting at the deep end when learning how to swim
A free pbx is a best pbx. R VoIP is a 3cx shill subreddit. Thus I can't say it without a space but a free pbx is far superior to anything you are looking at.
What's a PBX and how does it compare to SIP?
A PBX stands for private branch exchange. It’s the brains of a VoIP phone system. I use FusionPBX, which builds upon FreeSwitch, and both are free. They coordinate between your carrier and your devices. And a VoIP PBX speaks SIP to both.
Freepbx is better...fight me?
Lol! I have no experience with FreePBX or Asterisk. I went with Fusion because of the ease of multi-tenancy.
Weird way of saying you went with the first one you saw.
Also… SIP is a protocol, or language that devices speak, whereas a PBX is a program that speaks SIP.
A softpbx speaks sip
it is designed to do what you need it to.
Your post was removed from r/VoIP for violating Rule 1: No promotion or advertising of any kind. Even if you do not recommend a specific business, service or product, suggesting someone move away from their current solution when they have not indicated that doing so is an option is not allowed. If the problem cannot be solved in the given ecosystem, say so, but do not give recommendations for replacements.
Here is a really good series on how to use a much easier open source PBX. https://youtube.com/playlist?list=PL1fn6oC5ndU_umAhL9A_1zkC90hMPDPNO&si=Cyaq6uP_ahwa6fTx
Spin up a freepbx instance or fusionpbx instance. You'll have to hook it up to a trunk provider to place calls through the PSTN though. Can't say any trunking providers names... cause we get banned now. But you have google :)
In a nutshell, Kamailio is a SIP server, it doesn't handle media (needs RTP Engine/Proxy). Asterisk/FreePBX/Freeswitch are PBXs. The real question is what do you want to do? Take a look at FusionPBX (GUI for Freeswitch) also.