PIONEERS Incredible PBX Asterisk-GUI

pregomeister

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I just finished installing Incredible PBX Asterisk-GUI on my trusty virtualbox. Status says I'm running Incredible PBX 11.15-04, Asterisk 11.16, Ast-GUI 2.0-r5220. All this is on CentOS 6.6 x86_64. Sweet!! Great work by the entire team!!

I found a small issue I'd like to mention. After I created a 3rd extension 6003, I wanted to add it to the same RingGroup1 which has the preconfigured first 2 extensions 6001 and 6002. I couldn't save my changes. Looking at the asterisk logs, I saw that it couldn't access the file /etc/asterisk/odbc.conf because it was owned by root:root. A quick chown asterisk:asterisk fixed the issue.
 

Sippy

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Just installed Incredible PBX on a zotac box. When I try to edit the outgoing rules, The first few rules at the top will allow the edit screen to pop up, but when I go to the voipms section all I get is a line at the top that says "edit callrules!" and the edit box doesn't show up. I am using chrome browser with Ubuntu. Just tried it with Firefox and am getting the same result. How do I fix it.
 

ou812

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did you try and scroll up, it only appears at the top of the page.

gary
 

bltst2

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pregomeister This is file ownership on odbc.conf is breaking quite a bit. I was not able to add new Calling Rules, but changing the ownership to asterisk:asterisk fixed that.
 

wardmundy

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New versions for CentOS and Ubuntu (not RasPi 2 yet) are now available. These do away with the make menuconfig screen size quirk in favor of a new methodology.

 

sorvani

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No love on a CentOS 7 install yet. I spun up a CentOS 7 cloudatcost vm last night. The install completed, but MySQL and Apache were offline. Did not have time to do more and I needed to reimage it to 6.5 to test something else this morning.
 

smarks

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I have always liked Asterisk GUI and the fact it still works pretty good ever after all these years says a lot.

Anyone willing to join me in sponsoring support for Asterisk 13 and pjsip? I could probably donate some money, vps servers, or time etc.

Any chance Digium might be willing to revive this project or start/revive/contribute to some other GUI project not related to fpbx?
 

smarks

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Justin Hester at Digium published a helpful article just yesterday. Andrew at Frog Central also has done some previous work on this. It's definitely on our radar.

As for Digium's future intentions, who knows? They still appear to be schmoozing it up with Sangoma at least for now.


Thanks for the info. Those links take me to how to's on installing Asterisk 13 and pjsip. What I was referring to is adding config support in the GUI. As I'm sure you know, pjsip requires different config files to be generated by the GUI. Just getting chan_sip and Asterisk 13 working with the GUI probably would not be too big a deal but I don't know for sure. I'm guessing chan_pjsip support is considerably more work.
 

wardmundy

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Some Helpful BASH Scripts

Extract Extension Password
Code:
awk '/\[6001\]/ {p=1}; p; /^secret/ {p=0}' /etc/asterisk/users.conf | tail -1 | cut -f 2 -d "=" | cut -f 2 -d " "

Extract Voicemail Password
Code:
awk '/\[6001\]/ {p=1}; p; /vmsecret/ {p=0}' /etc/asterisk/users.conf | tail -1 | cut -f 2 -d "=" | cut -f 2 -d " "
 

pregomeister

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Thanks for the patch, Ward. I think there are more permission issues in /var/log/asterisk, in particular for the files queue_log and h323_log, both owned by root. In the file /var/log/asterisk/messages, I was seeing entries like this until I changed ownership to asterisk:

ERROR[28885] logger.c: Unable to create queue log: Permission denied

I also see many warnings like these in the same log file:

WARNING[28885] pbx.c: Unable to register extension '864' priority 1 in 'CallingRule_extensions_custom', already in use
WARNING[28885] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior. Please use '_X.' instead at line 1127 of extensions.conf

I have everything stock from the asterisk-gui distribution and things seem to work with a GV trunk and a vitelity one.
 

Jay Deal

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I am trying this out for the first time, runs fine but I cannot find G722 in the codecs section. I checked menuselect in /usr/src/asterisk-11.16.0 and it appears as it was compiled the first time but it does not show up as a choice in the GUI at least my version doesn't have it. Is there a different place where you can enable it?
 

wardmundy

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Jay Deal For extensions, you can manually add g722 in users.conf. Just search for ulaw to find where to put it.

For preconfigured trunks, you can manually add g722 in credentials_trunkname.conf

Don't forget to restart Asterisk.
 

Milton Alvis

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Any supported provisioning for Yealink T-48G and W52P phones yet?
The OSS module does not list these hardware devices as included, yet.
(We have been using our personal cell phones for our business, a workaround for the last 4 months; has sort of worked, with problems.)

I finally got BeagleBone Black running (purchased after the http://nerdvittles.com/?p=8222 article).
I had a several month long delay updating the OEM installed BeagleBone Linux Distro; I could not get the latest image listed at http://beagleboard.org/latest-images to flash the eMMC despite following the instructions many times, very frustrating.
I wasted time worrying the problem was the Samsung 32GB SDHC micro sd card I had purchased for this; this was not the problem. Indeed, the Samsung card writes & reads dramatically faster than a temporary alternative 8GB SDHC micro sd card I purchased & returned, working to follow alternative advice in the p=8222 article.
I finally found that only two of five 2015-03-01 images, as listed in the lower portion of page: http://elinux.org/Beagleboard:BeagleBoneBlack_Debian, will flash the eMMC.
I have attempted to warn others of the misleading instructions on the BeagleBone.org site; without success on the BeagleBone.org site pages so far.

SIP.US (as a big supporter of WardMudy's work & pbxinaflash) offers a routine (in their support page) to add all of a user's SIP trunks and numbers into pbxinaflash.
This worked great, promptly registering our trunks on SIP.US and 8 phone numbers into pbxinaflash on our BeagleBone Black; fantastic service.

However, I have been unable to find or figure out self-provisioning files for the 3 Yealink phones I purchased for our business as SIP phone starters, transitioning away from Grandstream ATAs & a 26 year old (great for its time) Panasonic 6/16 hybrid PBX system which I had purchased, installed & programmed.

Dear Ward, Thank you for all your comments about Sangoma & FreePBX. I was wary and had had difficulty understanding what the issues were. Much clearer now.
 

Stephen Bates

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Ward,

This is indeed an incredible product you have here. I have an issue where It seems the GUI now goes back to asking me to sign in whenever I click anything. Also the Status Page after signing in just shows the Button Bar, and the Incredible PBX Welcome Box, no data otherwise.
Please point me in the right direction here as I am out of ideas.


I have the Centos 6.7 distro from RentPBX

Thank you in advance,

Steve
 

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