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Upgrade 13 to 14?
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FreePBX Distro dahdi channels not configured
Hello,
I have a FreePBX Distro newly installed, and connected to a Xorcom Astribank to handle traffic from 7 FXO channels. I worked with Xorcom Support to recognize the dahdi channels into the system, but Asterisk does not show the channels when I use the “dahdi show channels” command within asterisk. Some online help I found says this issue usually means there are no channels configured in the chan_dahdi.conf file.
Any help with this issue is greatly appreciated.
I’ve configured the analog hardware using the FreePBX GUI Dahdi Configuration module, under analog hardware. Here is a screen shot:
The file chan_dahdi_groups.conf appears to have the group configuration I configured using the Dahdi Configuration Module. Here is the relevant part of the file:
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;signalling=fxs_ks
context=from-analog
group=1
channel=>15signalling=fxs_ks
context=from-analog
group=2
channel=>16signalling=fxs_ks
context=from-analog
group=3
channel=>17signalling=fxs_ks
context=from-analog
group=4
channel=>18signalling=fxs_ks
context=from-analog
group=5
channel=>19signalling=fxs_ks
context=from-analog
group=5
channel=>20signalling=fxs_ks
context=from-analog
group=5
channel=>21
I use the context from-analog because I have each analog line mapped to a DID.
The output of my lsdahdi:
### Span 1: XBUS-00/XPD-00 “Xorcom XPD [usb:X1080142].1: FXS” (MASTER)
1 FXS
2 FXS
3 FXS
4 FXS
5 FXS
6 FXS
7 FXS
8 FXS
9 Output
10 Output
11 Input
12 Input
13 Input
14 Input
### Span 2: XBUS-00/XPD-10 "Xorcom XPD [usb:X1080142].2: FXO"
15 FXO (battery)
16 FXO (battery)
17 FXO (battery)
18 FXO (battery)
19 FXO (battery)
20 FXO (battery)
21 FXO (battery)
22 FXO RED
### Span 3: XBUS-00/XPD-20 "Xorcom XPD [usb:X1080142].3: FXO"
23 FXO RED
24 FXO RED
25 FXO RED
26 FXO RED
27 FXO RED
28 FXO RED
29 FXO RED
30 FXO RED
The output of asterisk -rx “dahdi show channels”
Chan Extension Context Language MOH Interpret Blocked In Service Description
pseudo default default Yes
What am I missing to get asterisk to configure the dahdi channels?
Thank you
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FreePBX Distro hangs at install
Lack of support for modern processors in CentOS 6.6 is a major stumbling block now if you’re trying to install v13 on newer hardware.
It was something I wasn’t aware of when I first encountered it a year or so ago.
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Zulu presence not working?
This is not currently in Zulu 3.0. Feature requests are the best way.
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Cannot backup FreePBX 13 to FTP Server - failed to open stream: Value too large for defined data type
Thanks for the pointer Dave!
Huh… so what in the world do I do about that? Not sure how I’m able to change the name or anything else for that matter. Hoping someone has a definitive idea.
Rick
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Appointment Reminders
This topic was automatically closed 24 hours after the last reply. New replies are no longer allowed.
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Protocol 'tftp' is not enabled. Please enable in System Admin module
whoops message - I can’t get into GUI to enable TFTP
How do you use CLI to enable TFTP in System Admin Module?
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Outbound Route - Time Group Assigned
I was looking at a FreePBX/Asterisk 14 box for a friend that had mentioned they couldn’t make outbound calls.
I looked at thier outbound routes and saw the Time Group Assigned lit up as green. Thinking the problem was that they had a time condition conflict with thier outbound route, I clicked the edit button.
I noticed the Route was set to permanent and there were no time groups setup in the time group module.
I deleted the route, made a new outbound route and added the trunks, but the new outbound route lit green right away.
Any ideas on what we might be missing?
Thanks!
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Transferring a call to a prerecorded message
Hi All,
I have been poking around for some solutions to a challenge my team has brought up. When our teams call out to vendors, we leave a lot of voicemails. We would like to be able to have our callers transfer the call to a prerecorded message when we come upon having to leave messages. Is there a feature or app that can assist with this? Or some sort of solution that wont take a whole lot of administration from my end?
Thanks!
Adam
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Problem with CID when transferring calls
How about you resolve your noticeable errors first
apparently does not exist in the innertubes.
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Protocol 'tftp' is not enabled. Please enable in System Admin module
edited endpoint_templates table with MySql, - changed all templates protocols to HTML
Now GUI starts
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9000 Extension Architecture
If you have ever been in the position to handle more than a very few hundred concurrent SIP calls, you will just NOT use a B2BUA, wrong call on every level.
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Calls not hanging up
Additional info.
When the PBX should hangup, CLI shows:
ERROR[2211]: pjproject:0 <?>: sip_endpoint.c Error processing packet from ProviderSIP-IP:5060: Missing required header(s) (PJSIP_EMISSINGHDR) CSeq [code 171050]:
SIP/2.0 400 Bad From Header
Via: SIP/2.0/UDP LocalPublic-IP:5060;rport=5060;branch=z9hG4bKPj0e43e49a-e2e5-4baa-87c3-6ea0546b2b11
Content-Length: 0
Still cannot get to the bottom of this.
Any hints please?
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Calls not hanging up
Breiefly change to chan-sip and see if that still pertains.
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Calls not hanging up
This is the first thing I tried.
I have tried again, just in case and the same situation.
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Asterisk phone book used to cache caller id entries but no more?
In previous versions of FREEPBX I set up CallerID superfecta to check the Asterisk phone book first then go to OpenCnam. In previous versions it would add the caller’s ID to the asterisk phone book that way the next time that caller would call it would never go to opencnam saving a few cents.
Since I upgraded to 14, it no longer caches the results in the asterisk phone book. I can only enter manually
Any ideas?
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Calls not hanging up
It seems it might be from the SIP provider’s end.
I setup another PJSIP account from another provider and this one does not exhibit the problem, calls being hangup cleanly.
As a side note, the problematic account also shows:
WARNING[8267][C-0000000e]: chan_sip.c:22961 func_header_read: This function can only be used on SIP channels.
While the 2nd test account that works, does not show that error.
Now I just need to get the provider to fix this somehow…
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LetsEncrypt on New Installation ( Direction that work all the time )
After several attempts on Getting Lets Encrypt to work on PBXact and FreePBX on SNG7 I finally got it to work with a grade of A on SSLLABS.com instead of an F.
1.) Install SNG7-1710 on machine
2.) At first login - Do Not Activate the deployment yet. — do yum update
3.) After that is complete reboot system
4.) login and Activate the deployment
5.) fwconsole --edge ma upgrade certman
5.) SSH into the box and delete the following directory /etc/asterisk/keys/_account/ ( just the _account directory
cd /etc/asterisk/keys/
rm -r _account
6.) In Sysadmin Pro if you have it - goto DNS and setup your FQDN
7.) Proceed with the LetsEncrypt procedure on issuing your certificate.
Hope this helps anyone with those annoying errors… Tried everything else and just gave up and it looks like this has been working flawlessly for me.
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Phone going off hook incorrectly: S700 with Plantronics Savi 740 on EHS
@Overkill Thanks for the update we have ten s500s and all of our users are asking for headsets with Bluetooth / PC connectivity. I do hope that you or @qwell gets some news from Plantronics sometimes soon:thinking: as for now I have delayed this project/purchase of more s700’s to next year (might have to look at other options). If you think it would help I will email Plantronics support with a detailed description of the issue that we are having, with the hopes that another squeaky screw will urge them to push an update to their firmware.
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Transferring a call to a prerecorded message
The broadcast dialer looks like it would work. Take a look
https://wiki.freepbx.org/plugins/servlet/mobile?contentId=5537932#content/view/5537932
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