FOOD FOR THOUGHT Help with learning IncrediblePBX13

ballhogg

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Hey Everyone,

I'm trying to get up to speed on IncrediblePBX13. I've been using older versions for some time, and while not really technically savvy regarding the behind the curtains workings of this, I'm fairly comfortable setting up and using IncrediblePBX (and very willing to experiment and learn!). However, there appears to be some pretty major differences between IncrediblePBX11 and IncrediblePBX13. (i.e. I have know idea what chan_pjsip is vs. chan_sip, and the different splash page for monitoring). Normally, I'm just fine playing around with things and learning on the fly. But with this one, I seem to be stuck right out of the gate. I've got a brand new install of centOS 6.8 with IncrediblePBX13-12.2 and IncredibleFax11 installed on a virtualbox vm. I can login to the web interface, and I've set up trunks, inbound routes, and outbound routes just as I have in previous versions of IncrediblePBX. However, I've been unable to test anything because I can't get my x-lite softphone to register to any of my extensions, including extension 701 that was preconfigured during setup.

So, I don't know if there is an issue with my initial setup of IncrediblePBX13, or if something has changed in the way things are done that I'm not addressing. In searching through the forums, I found one reference to issues with registering extensions to IncrediblePBX13 initially (http://pbxinaflash.com/community/threads/unable-to-login-from-ext-701-new-install.18957/#post-119568), but there was no apparent resolution to the problem notated in that post. So I don't know if this is a continuation of that problem or not.

At any rate, if anyone has a suggestion on how to get a device registered, I'm willing to continue playing and learning. Or if anyone has some suggestions on where to go to learn how to use IncrediblePBX13, I'd really appreciate the help.

Thanks in advance for your responses.
 

stanjohn

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Check out Knock and traveling man by default they are a little more aggressive in the latest version. You can add your public ip with the ip-add command.
 

progs_00

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I would also suggest that you made sure your xlite is registered as a pjsip extension. If you need any help post an extract of the cli while your phone is trying to regsiter so we can see what is going on
 

ballhogg

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Check out Knock and traveling man by default they are a little more aggressive in the latest version. You can add your public ip with the ip-add command.
A couple questions regarding this point. First, do I need to explicitly install travelinman or is it automatically installed with the incrediblepbx installer? Second, does the firewall affect ip's on the same subnet as the pbx? I thought it only affected ip's coming from outside the network.
 

ballhogg

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I would also suggest that you made sure your xlite is registered as a pjsip extension. If you need any help post an extract of the cli while your phone is trying to regsiter so we can see what is going on
This is actually one of the things that I'm most curious to learn about. What is pjsip vs. normal chan_sip? I don't know anything about this, in any way.
 

jerrm

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This is actually one of the things that I'm most curious to learn about. What is pjsip vs. normal chan_sip? I don't know anything about this, in any way.
Chan_pjsip is just a newer alternative SIP driver for asterisk. The goal is to eventually replace chan_sip, but that won't be anytime soon. My personal wild a** guess is not before Asterisk 16. Too many kinks to work out still.

From the perspective of the endpoint there should be no difference, they both speak SIP, just use different back end code to do it. In the real world, there seem to be more blf issues and gremlins with chan_pjsip.

From a configuration, programming, API and architecture viewpoint pjsip is superior. The pjsip library is used in many products with a larger user and developer base should progress faster, it's more modular, etc.

From a daily use viewpoint, the only advantage I find for chan_pjsip is the ability to register multiple endpoints to a single extension, but FreePBX12 support of pjsip is incomplete and some things don't work as they should for the multi-endpoint feature.

Bottom line - for IPBX13, I swap the ports around so chan_sip is listening on 5060 and only use chan_pjsip for extensions where multiple endpoints are really needed.
 

philsh

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I have just downloaded and installed current version of IPBX. I also cannot get a softphone to connect to an extension. I cannot see what I am missing. I have done this in the past with older versions of IPBX with no problem. Is these some default that has changed?
 

progs_00

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@jerm

Your last paragraph was the saddest thing about pjsip. Necessary I guess but bad nontheless. Initially I had quite a few issues with pjsip with all my extensions. I had to switch back to "plain" sip which given the port change caused even more trouble. However, I believe that pjsip is currently stable enough to make a safe (more or less) first choice. I currently have all my extensions in pjsip and only a remote one is using sip on its new port.

Guys post some debug info because like this it's just like shooting in the dark
 

philsh

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On the softphone side, is all that is needed: IP address of IPBX, username (extension number ie. 701) and password (secret) ? Is it just those three pieces of information? Thank you.
 

progs_00

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Hi philsh,

Happy to help. Can you please try to register your softphone by creating a new extension (pjsip and all else default)?
 

philsh

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Well, I deleted all the accounts. Rebooted. Created one extension as pjsip and it works! Not sure what I put in wrong. Oh well. Can you direct me for the following? I have an account with Vitelity. I was able to register my account. Outbound works fine. Inbound does not. I created a new Inbound "default" route and have it sent to my extension (702). But I am not sure if I need to "connect" that default route to a trunk ? Thanks.
 

jerrm

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Your last paragraph was the saddest thing about pjsip. Necessary I guess but bad nontheless.
The sad thing is that 2.5 years after Asterisk12 release chan_pjsip is not where it needs to be. In my opinion it's really only reached the "mostly acceptable" level in the past six months or so. We still experience what I call "gremlins" - inconsistent but usually OK behavior with some phones that just don't happen with chan_sip.

I'll be the first to admit folks like myself are part of the problem, not taking time to report and debug the issues because chan_sip is sitting there and stable.

The other issue from an IPBX perspective is the many FreePBX13 pjsip fixes will never be backported to the 12.x tree, and I doubt there will ever be a FPBX13 based IPBX, so why bother with chan_pjsip unless you really need a pjsip specific feature. Use what is the most stable and supported option - chan_sip. I don't see any "pjsip is the future" benefits if there won't be a future at all for the current platform.

When the time comes to change, decide what fits your needs best and go with either another FPBX13 based option or IPBXivo. For our needs, based on where features are now, I would lean toward non-distro FBPX13, even though I feel XIVO is a technically superior base platform. I had hoped we would see something on the FPBX13 front from the @phonebo.cx folks by now.
 

progs_00

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Well, I deleted all the accounts. Rebooted. Created one extension as pjsip and it works! Not sure what I put in wrong. Oh well. Can you direct me for the following? I have an account with Vitelity. I was able to register my account. Outbound works fine. Inbound does not. I created a new Inbound "default" route and have it sent to my extension (702). But I am not sure if I need to "connect" that default route to a trunk ? Thanks.

Great to see we got the first part out of the way.
Now If I got this correctly you created your outbound route and then your trunk with the settings Vitelity gave you. Right?
This the absolute minimum to get the system working.
A default Inbound route (catch-all) can be left as is and the system should work. If you are having trouble with the incoming, it's most definitely an issue with settings. If I recall well, IncrediblePBX has Vitelity already included in the trunks configuration along with specific rules to unblock it inside IPTables. Verify this and afterwards open an ssh connection to your box, type
asterisk -rvvv
and perform an incoming call to your system. This will tell you if packets arrive and we shall see from there. If you need help debugging, post the cli output here and I will try to lend a hand.

@jerm
The sad thing is that 2.5 years after Asterisk12 release chan_pjsip is not where it needs to be. In my opinion it's really only reached the "mostly acceptable" level in the past six months or so. We still experience what I call "gremlins" - inconsistent but usually OK behavior with some phones that just don't happen with chan_sip.

I'll be the first to admit folks like myself are part of the problem, not taking time to report and debug the issues because chan_sip is sitting there and stable.

True. PJSIP had a really slow start but to tell you the truth I don't have any remorse for not helping out. LOL It takes time to debug, resources and a lot of frustration. So I leave this to the professionals.
In the end, we know that SIP needed 10+ years to get to where it is now so we have to give PJSIP some more time. From my end, after some initial frustration, I don't have any particular problem to mention.

The other issue from an IPBX perspective is the many FreePBX13 pjsip fixes will never be backported to the 12.x tree, and I doubt there will ever be a FPBX13 based IPBX, so why bother with chan_pjsip unless you really need a pjsip specific feature. Use what is the most stable and supported option - chan_sip. I don't see any "pjsip is the future" benefits if there won't be a future at all for the current platform.

With this, you hit the nail on the head. Let me start by saying that I HOPE there will never be a FreePBX13 based IPBX. There are numerous reasons why I'm saying this, but it all boils down to this for me: what was a great project that pushed the community forward is now becoming a project that is holding the community back. We owe FreePBX a lot, but I believe it's time for a better, true open source alternative. XiVo seems to be the right answer for now and even if we don't get to see the future in the current platform, we will definitely see it in a new one. So as long as someone finds PJSIP stable enough, why not try it and if things don't work out, it's only a flip of a switch and you are back to good old SIP

When the time comes to change, decide what fits your needs best and go with either another FPBX13 based option or IPBXivo. For our needs, based on where features are now, I would lean toward non-distro FBPX13, even though I feel XIVO is a technically superior base platform. I had hoped we would see something on the FPBX13 front from the @phonebo.cx folks by now.

I believe they started a fork of FreePBX 13 just a couple of months ago, so it's gonna take time. It will be an interesting project, but for me, from an open source perspective, time spent on FreePBX, is time wasted. We need true open source alternatives that the community can work on
 
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