Just tellin!, no, asterisk 22 (has it even be released yet) and definitely not been fully tested by Sangoma. What’s the hurry, why do you need 22 now! now! now! there is not even a Changelog published, especially as macros and chan_sip disappear farther into the mist?
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Installing FreePBX on Asterisk 22 using install script?
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Exciting News: FreePBX 17 is Now Generally Available!
pssssst ted
sshh dont tell FreePBX , seems they call it a protocol too, sshhhh
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Exciting News: FreePBX 17 is Now Generally Available!
I never seen any compatibility issue with a SIP provider and PJSIP.
Are you sure a SIP provider uses SIP or PJSIP?
There is several SIP servers based on other drivers outside PJSIP. It should work.
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FreePBX 17 Dashboard Error: Unable to configure networking service: systemd-networkd conflict
What cloud provider are you using?
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No Audio when call gets forwarded to a cell phone
Thank you for the explanation. I removed the ip that was set up as the External Address and removed Force Answer and now it seems to be working fine. Thanks, again.
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Exciting News: FreePBX 17 is Now Generally Available!
You want to define what you mean by supported? Because this statement is not factual. You are aware there is an entire base of users that use Asterisk standalone, right? Getting a PBXact systems doesn’t mean you get Asterisk support. It means you get support from the FreePBX team who manages/develops PBXact. Remember that while both projects are under the Sangoma umbrella they are not the same project and are run completely independent of each other.
I mean you have more support from the Asterisk division if you got Switchvox instead of PBXact because Switchvox is the Asterisk solution vs PBXact which is a FreePBX solution. Heck, Switchvox got rid of chan_sip, app_macro and other crap years ago while FreePBX has been clinging to it.
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Can't dial outbound calls
All FreePBX wiki content has been migrated to Sangoma Documentation including for the Extension Routing module
https://sangomakb.atlassian.net/wiki/spaces/PG/pages/32572684/Extension+Routing
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Keep CallerID after being forwared from queue
Hello all,
When we answer a call form a queue and then forward this call to an extention the number of the agent is being displayed.
Is it possible to keep/retain the orgiginal CallerID after being forwarded from a queue?
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Exciting News: FreePBX 17 is Now Generally Available!
Then in the future when someone reads this forum here and wants to go to the URL they are screwed. Why would you want to make it harder for someone else to access a resource that helped you?
I tried. He has a large number of IP addresses blocked from posting. I don’t know if the few comments on there were ringers or whether he had it open for posts for a few days then blocked the rest of the Internet but that’s the facts.
The only thing that has changed is the asterisk development group isn’t maintaining it any longer someone else is. And Asterisk.org states this as well - they say on their site that they don’t have a problem if someone else wants to continue to maintain chan_sip - which is happening, by the way. Asterisk was written with the concept that channel drivers are plug-ins specifically for this reason.
It is like how Microsoft has been pushing vendors of hardware for years to write their own device drivers instead of depending on Microsoft doing it for them.
It’s important to have documentation written for newbies, very important. So important in fact that 24 years ago I wrote and published a book for FreeBSD that was targeted at newbies. However, it disrespects people when you give them wrong information in an effort to make it more understandable. They need the information presented more simply so they can understand it, yes, but it needs to be the CORRECT information. Otherwise they gain a flawed understanding of how something works and then proceed to embarrass themselves when they get on other forums and ask questions and get frustrated when their flawed understanding causes them to configure something wrong.
Even the experts have flawed understandings and when there are complex issues in play it can be very hard to write out what is going on in a way that communicates the understanding properly. My recent discussion on rtp udp port ranges if you look it up for example. But it is critically important to continue trying to get it right.
I’ve been corrected plenty of times in the past with my flawed understandings just as I’ve called out other people with theirs. It isn’t fun when you are certain something works a particular way then discover the entire time you had the wrong assumption of how it worked and you were just lucky. But if you aren’t willing to stop, and get past the emotion, then go dig into the standards and practices and learn about your misconceptions and change them, you won’t ever progress to guru level.
Now, it is true that many IT techs really only want to learn “just enough to get by” so they can go home and spend their evenings playing Warhammer instead of reading technical journals. I’ve hired (and fired) many like this. This is, unfortunately, increasing in this industry and it’s a large part of why “the cloud” was invented because that allows companies to use mediocre IT techs who fall flat on their faces when confronted with anything interesting, as they can just dial an 800 number for a guy in India who RTFM’s to them for an hour then eventually remotes into their desktop and does it for them.
But don’t you think it’s sad to just sort of float through life never understanding why things work and just sort of living life wrapped in cotton wool? Expecting that meat is manufactured in the back of a supermarket and comes on Styrofoam trays, that everyone is treated equally by police, that electricity is something that is dug up out of the ground and squeezed through those little slots on the wall that you put plugs into? Anyway, I do - which is why I continue to call out misinformation when I see it.
I don’t have anything against the guy who posted his blog just because he’s not as far along in his understanding I just don’t want him misleading others is all, and accepting corrections to his posts is the responsible thing for someone publishing anything. I would NEVER have gotten my book published by Addison-Wesley if I had the attitude that it was OK to just publish wrong information. Nowadays, blogging and so on has replaced technical book publishing for a lot of things, but just because it’s someone’s personal blog does not mean it’s OK for them to block anyone from correcting them when they make a mistake. They still have the same responsibility I did 24 years ago.
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Audio being received but not sent?
Hi all,
I’m relatively new to VOIP so sorry for in advance for what is hopefully a simple question…
I’ve got a basic FreePBX V17 install going on a Pi. At the moment I’m just trying to get internal calling working - I haven’t configured anything to interface it with the outside world so to speak.
I’ve made 2 extensions and am using MicroSIP as my client. The problem I’m having is that I can’t hear any audio on the other end when I talk. I can hear audio coming from Asterisk - E.G. when the call goes to voicemail I hear the greeting, but no audio appears to being sent - only received. Voicemail hangs up while I record a message, presumably because of some kind of silence detection functionality.
All devices are on the same network. I’ve verified my mic settings on the clients are all correct (I’ve tried multiple computers). I also have an old ATA (Linksys PAP2) which works with my current provider, but experiences the same issue as MicroSIP when I reconfigure it to point to my FreePBX install.
Does anyone have any pointers for things I could try?
Many thanks,
Ben.
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Exciting News: FreePBX 17 is Now Generally Available!
I can define what I mean by supported and I can define what is generally accepted as “supported” by the industry but I can’t define what the poster in that blog meant. I can certainly define what he IMPLIED by supported since he wrote it!
In the industry “supported” means the customer can pick up a telephone when something doesn’t work, call an 800 number and it will be fixed for them. This is why complex systems like PBXes are sold through resellers because the PBX vendor knows that even if they build a complete turnkey system - like Switchvox - and not only supply the PBX but supply the phones as well and maybe even supply the SIP trunks - there’s still the ethernet network to contend with, ethernet switches, ports, etc.
The reality is that “support” is “sold/pushed” as a marketing term that simply does not really exist in real life. The marketing and sales guys will IMPLY - as the blog poster did - that their “supported” system that you are paying a yearly service contract for will always be fixed for you if something goes wrong - even though if anything truly knotty comes up you have a 99% chance if you -for example- call Cisco TAC - that when they run out of ideas they will start blaming something else that they aren’t responsible for. Heck they might even blame the electricity in the wall. So even the “paid support” has limits to it.
But the blog poster was writing a DIY guide and by definition DIY is supported only by the person doing the DIY - that is, it’s COMPLETELY unsupported if you use the generally accepted definition of support in the IT industry. So it’s rather silly of him to go on and on about this and that being unsupported when he’s writing a guide for people who wouldn’t even pay for a modicum of support - support as defined by the computer industry.
What Sangoma and Asterisk and FreePBX have discovered with “skunkworks” projects like FreePBX is three critical things:
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Many people out there will start with “unsupported” stuff like FreePBX then migrate to “more supported” stuff later that then turns them into paying customers.
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Sometimes the skunkworks users will invent things that are good product ideas that can be reworked into commercial products. (like, EPM for example)
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The skunkworks users are great testbeds for changes because they will loudly complain when you break something during changing it but because they are getting it for free they will tolerate your changes and work with you to get the changes fixed - even though they may complain the entire time. The paying customers will also loudly complain when you do this - but unlike the freeloaders, they aren’t particularly tolerant and won’t give you very long to figure it out before they make the inevitable conclusion that they aren’t getting acceptable value out of what they are paying you for and go to someone else.
Because of those 3 things they are making money and as long as those things remain true they will continue with the FreePBX skunkworks project. And they will continue doing this balancing act of trying to give just enough support to keep the free users happy, but not so little that they will give up in disgust or fork the project or whatever, and not so much that it destroys the incentive to migrate to the “more supported” products, PBXact or Switchvox.
I’ve spent my career operating in that grey area - I’ve gotten tons of value out of Open Source operating systems and was running MINIX and later 386BSD Unix long before Asterisk even existed, wrote a book on it the same year Asterisk was written. I even wrote a chapter in my book about how wrong that Microsoft was (at the time) with their scorched Earth policy against what they perceived as their competitors - OSS being #1 - and was later vindicated when the Microsoft board quietly convinced Bill Gates to leave, and vindicated even more when Microsoft did it’s mea-culpa about this and created it’s Microsoft Loves Linux marketing campaign and came out with Azure Linux.
So yeah, I get it. My statement IS factual when you define “support” the way that the blog poster did - dial-an-800-number way. It isn’t a statement on what “support” means in OSS or in the FreePBX project or Asterisk, which is far more nuanced. Indeed, the main reason I was outraged was the poster’s misuse of “support” It really takes a ton of gall to be knocking something for being “unsupported” while you are writing a support page that is needed precisely because you believe the level of support on the product you are writing about is inadequate!!!
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Keep CallerID after being forwared from queue
How are you forwarding it? If you use a SIP attended transfer, this is indistinguishable from a new outgoing call until the transfer is completed, at which point you need Send PAI to enabled for the number to be updated on the callee phone. I believe that feature code transfers do propagate the caller ID.
It is propagated on a SIP unattended transfer, but some phone implement these as an, automatically completed, attended transfer.
If you are calling an external number, anti-spoofing measures open a whole new can of worms.
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FreePBX is sending channel name and CID to callerID
All the highlighter info is passed to the caller ID along with the ext number (114).
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Can't dial outbound calls
Is there a way to identify the malfunctioning module?
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Installing FreePBX on Asterisk 22 using install script?
I don’t need it now. I need to know where it and FreePBX is headed. And rather than ask and get someone’s biased and likely incorrect opinion, it’s better to just download it and build it and see for myself. That’s kind of the point of having all of this code open source, after all.
That’s pretty funny considering that there’s been no changes at all in that regard at least with chan_sip, the differential patch that adds it back in still works and the developer who maintains that has not changed his statement that he will continue to keep it working, chan_sccp also still works with it.
Macros is a different story but I have to assume the people who depend on those are either recoding, or continuing to use the asterisk version selector command to ride out the older Asterisk versions until they get a round tuit, or possibly are just a few years away from retirement and stalling it out so after they retire they can turn to whatever former employer is still dependent on Asterisk macros and give them a big FU, your problem now, buddy.
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Cisco 7941G gets a "Registration rejected"
dicko, as I made it clear the phone (and, likely, Asterisk) is obviously not using the non-mod-2 ranges of 10000-20000 and 16384-32766 as a directive to block incoming rtp that is not within that range, but rather as a directive to restrict the starting port it will select for rtp streams (not rtcp). And in that case, the range does not need to be mod 2 since it would only determine the source port for rtp - because as you point out, the port for rtcp is always +1. The fact it works proves this. The code is obviously ignoring that directive for rtcp.
If the phone and asterisk was refusing to accept rtp outside of it’s defined ranges - nothing would work at all since as Tom pointed out lots of things have different “media port range” settings yet all clearly work together.
These directives are antiques left over from an era where NAT devices were not RTP-aware and all you could do was port forward 5060 and some rtp range you defined for your phone and your Asterisk system and add your “public” IP address into various other configurations in your phone and PBX and cross your fingers and hoped it all worked. Which often, it didn’t.
That is why I told the OP to remove his public IP address from his phone’s config. Now, if he’s using public SIP trunks he might possibly be in a situation where he needs that. But he needs to get the phone registered in, and he needs to get extension-to-extension calling working, first. And in any case he likely has a modern router that knows about SIP so even if he is using SIP trunks the NAT in the router will handle any rewriting needed.
Unlike you I don’t see no value in discussing all this. Even some modern SIP phones manufactured a month ago have these antique configuration directives in them. I certainly wouldn’t go buy hundreds of old 7941Gs for a new installation (someone else might, lol) but it’s useful to know that you could walk into a company like Dunder Mifflin Paper that had 50-100 of these phones on people’s desks and drop in a brand new FreePBX system and support them all with chan_pjsip.
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Cisco 7941G gets a "Registration rejected"
Again you miss the point, if a non mod 2 range is blindly chosen for RTP range and the same one used for 'unblockin’or ‘routing’ back to Asterisk a number of calls will have problems, and the smaller that range is the higher proportion, there are no idiot checks in Asterisk to correct a non mod2 range so choosing a mod2 range for asterisks makes sense and confuses no-one.
One in 5000 calls with problems will hardly be noticed, but on in 3 almost certainly would, so I pointed it out.
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FreePBX is sending channel name and CID to callerID
You might try moving through each of the trunks and routes and extensions; and one-by-one, start saving them - so a bit more than reviewing them - in order to force push new values in to the database. Sometimes new defaults are added in new versions of the software and the old portions of your configuration need placeholders for the new values.
Or, you might try re-making new trunks and routes, with a copy of the old ones in a separate window/tab so you can copy-pasta important parts, then delete the old ones.
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Sangoma Talk lost connection to PBX reconnecting
Don’t think the bare metal vs virtual has anything to do with it. If you upgrade to Asterisk 18 does it also break?
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No Audio when call gets forwarded to a cell phone
I am not sure that removing the External Address is the answer here either. If you are behind a NAT then you need to specify the actual External IP here that will get NATed as that will allow the SIP packets to be built properly by asterisk.
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