I’m no more capable than you - it’s just us chickens. Click on the “Issues” tab at the top of the page and fill out an issues ticket. The developers will look at it and adjust, as required, the procedures on the Wiki.
↧
CentOS 7 install failure
↧
Custom IVR with JSON
For the purposes of discussion here, FreePBX is nothing more than a GUI for Asterisk (I know, but for this, it’s OK) so the installation of this module is almost nothing more than the regular Asterisk install.
If there are “contexts” to install, you’ll need to add them to the appropriate “whatever _custom.conf” files (e.g., extensions_custom.conf) instead of the ‘base’ Asterisk conf files. Other than that, it should be pretty much straight Asterisk.
↧
↧
Asterisk does't start at startup
No, If you want to do something like that, add “fwconsole start”. Starting Asterisk naked like that is only going to cause pain and confusion.
↧
Changing Context and CallerID in sip_additional.conf
If all you are trying to do is modify caller ID name we did something similar using the asterisk phonebook under the Admin section.
https://wiki.asterisk.org/wiki/display/AST/Function_DB_EXISTS
Put 7322 & John Six for the Number and Name Asterisk Phonebook fields respectively.
exten => s,n,Set(NAMEOVRD=${DB_EXISTS(cidname/${CALLERID(number)})})
This will set the ${DB_RESULT} variable to the CID Name in Asterisk Phonebook. You could then set your ${CALLERID(name)} to ${DB_RESULT}.
What was cool about this was you could grant users access to just this module to keep the names up to date. No additional “coding” when making changes. The phonebook also supports .csv import/exports for mass changes and uploads.
Sangoma support here:
↧
Custom IVR with JSON
This is an Asterisk module, and requires that Asterisk be re-compiled. If you’re using the FreePBX distro there is no supported way to compile Asterisk. You are probably better off using a normal AGI file.
↧
↧
CentOS 7 install failure
Ticket submitted.
↧
Voice pjsip quality issue
I tried to record conversations and the quality is so perfect that is like listening an mp3 music file, no way to compare with endpoint quality. what does this mean?
↧
Play an audio message when I pick up the phone's handset
… unless your channel driver supports it. Chan_SIP and PJ_SIP definitely can’t do this, but other (stupider) channel drivers (like DAHDI and Chan-SCCP-B) can actually interact with the phone once it goes off-hook. Of course, both of these drivers rely on the phone to be as dumb as a bag of hammers, so it’s going to depend.
If you were willing to wait until after the call is dialed, then you could probably implement something that would work across the spectrum, but for most people - @dicko absolutely has the right answer.
BIG NOTE: I would not recommend abandoning working SIP phones to adopt any 20-year-old technology so that you can get something like this working.
↧
Voice pjsip quality issue
That the trunk quality (server to server) is good, but the leg from the server to the phone is where the problem is.
Remember, Asterisk is a back-to-back user agent, so calls from extension A to phone B are always actually two calls - from to the server and one from the server to the remote phone. Since the quality is good at the server, the problem is likely to be transcoding on the fly of the codecs in use on the extensions.
↧
↧
Custom IVR with JSON
I know its not the same thing, but we’ve been using Shell in the dial plan, with a cURL and pearl commands to refine the JSON result to just what we need. It works very well with our FreePBX distro without having to install any additional dependencies. All in just one line of the dial plan.
↧
Voice pjsip quality issue
thanks for your clarification, is something I can do or check to improve this?
↧
Odd messages in mailog
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
↧
BLF hints over IAX2 trunk to remote system
Hi, I’m in the same position but don’t know how/where to actually load up the php script. I’m guessing this is pretty basic stuff…I just haven’t had to go this route in the past. Any help would be appreciated. Thanks.
↧
↧
Asterisk 13.21.0
Hi, we did yum install sangoma-devel but still are unable to get version 13.21.0
any ideas why?
↧
Destination voicemail in ring group changed automatically
It seems that the destination in one of our ring group automatically changed from the “unavailable message” to the “no message” for that particular extension without anyone doing anything to modify that. How can we prevent this from happening and why would it happen to begin with?
↧
DAHDI Analog Ports Missing - Digum X400 A8B
During an upgrade from 11 to 14 in January 2018 the dahdi board, a Digum x400m A8B, ceased working. I found a post about a driver issue, kmod-wanpipe and wanpipe, so I did what was suggested and rolled the driver back to 7.0.20-9.sng7. This solved the problem and has been working fine.
At some point this past Friday the dadhi board quit working again. I was not notified until today. I have applied all upgrades, from the GUI and also o/s upgrades. The dadhi board still reports no ports. I ran several downgrades which did not help. The last downgrade I tried was from 7.0.20.13-1.sng7 which should have taken it down to 7.0.20-9.sng7 but that failed stating “Only Upgrade available on package: Kmode-wanpipe-7.0.21-1.sng7.x86_64” and the same for wanpipe.
How can I downgrade to 7.0.20.13-1 or better yet, why can’t I run the latest version of wanpipe an have the board work?
↧
Cannot provision S205 or S405's
Really banging my head against this, need some help.
We have all polycoms, finally got my hands on some sangoma phones and now i cant get them to provision.
Using option 66 for the polycom, https provisioning.
They are in the same VLAN as others and everything.
↧
↧
Best phones to use with freepbx
Is there a suggestion for phones that are very easy to update firmware, very reasonable on the used market, and very easy to configure to work with freepbx?
I purchased some cisco 7960 phones this past weekend and I have no idea where to begin to check if the firmware on them will work or not, in attempts to update them, I found I cant download the firmware from cisco as I dont have a service contract with them.
So confused where to go/start.
↧
Global time condition day/night toggle
I know each time condition has a day/night number. Is it possible to create a global number where all time conditions, in this case multiple offices, multiple time conditions, can be activated all at the same time? If the switchboard/front desk, wants to activate all time conditions, before 5 pm, is there a way to do that globally, w/o hitting a BLF for every time condition?
The customer wanted that ability at the time group, but that module doesn’t have that ability.
IDEAS???
↧
Best phones to use with freepbx
I would use this as your guide:
https://wiki.freepbx.org/display/FPG/EPM-Supported+Devices
Certified for the “easiest” experience.
OPINION: If you are not very experienced with FreePBX in general, it can be nice to have the Sangoma phones.
- The phones come with support. One stop (Sangoma) for troubleshooting your whole experience (FreePBX + Endpoints).
- Many of the integrated commercial modules are free for the phones (like endpoint manager).
↧