↧
Downgrade PhoneApps-Mandatory red MWI light not needed on voicemail app
↧
Conference Menu
Yes that will always work.
The thing Conf Pro offers is it create a IVr for you and on a per conference room basis you decide if you want it included in the conference room.
Also you have have it auto create conference rooms automatically on a per user basis and let that user view and manage the conference room from UCP including kicking, muting and inviting callers into the conference room.
↧
↧
Is there a way to edit Digium phone settings from FreePBX?
We need to change extensions on all of our phones. At first, we thought we could do this from the OSS Endpoint Device List but it's not working how we expected to. The steps we did are:
1. Added the new extensions by importing bulk extensions CSV file.
2. Went to OSS Endpoint Manager, edited the phone and assign it to its new extension.
We thought by clicking "Rebuild" would push the setting on the phone and update it from its old extension to the new extension. Isn't this how it should work?
If that is not how it should work, where can we update all of our phone extensions on the phones via FreePBX?
Another option for us is to access each phone's settings (typing in the phone's IP address on the browser) and just change the extension manually one by one. This is not the fastest way to do it.
If anyone knows the faster way to change phone extensions on the phone settings via FreePBX, we would love to hear your solution to our issue.
↧
Is there a way to edit Digium phone settings from FreePBX?
I don't know how oss epm works but simply rebuilding the configs does nothing. You have to reboot the phones so they can ask for the updated config, assuming they are pointed to your server. The commercial EPM has an option to send a reboot. I think the oss one does too.
tl;dr reboot after rebuilding the phones. Make sure your server is the "configuration server" in the phone"
↧
Cisco BLF Pickup not working?
hello community.
so free pbx and cisco 7970 ip phones, what a ball ake hay.
but i have the phones sip patched, asterisk patched with the cisco patch.
have the phones working, sending and receiving calls, even have the BLF working. go me.
but i cant get the BLF cal pickup to work.
i have followed this guide
//docs.acsdata.co.nz/asterisk-cisco/line-keys-xml.shtml
and this
//wiki.freepbx.org/display/FOP/Cisco
have all the info in and reverent code in the phone xml and the asterisk files.
but when im making a call my blf on another phone goes orange i press this to pick up the call ans get a notify message in asterisk when watching it live with the -rvvvv command.
it says
x-cisco-serviceuri-blfpickup-201' rejected because extension not found in context 'from-internal'
so im figuring that i need to tell asterisk what this x-cisco--------- dose even tho i have
; ----- Group Pickup softkey
exten => pickup,1,Pickup()
same => next,Hangup(normal_circuit_congestion)
in my extensions_custom.conf witch in included in my config file.
i cant seem to figure out what i need to put where, can someone please help me ?
↧
↧
Is there a way to edit Digium phone settings from FreePBX?
We rebooted the phone. It has no effect. We tried to set the phone for our phone server to be the configuration server and used the default port TFTP and it failed connecting to it.
↧
Setting for syslog server?
Hello,
Do the phones have anyway to point to a syslog server? Can it be set by DHCP option 7?
Thanks,
Daniel
↧
CallerID Name over IAX2 trunk - shows local not remote name
Hi,
I have 2 FreePBX systems trunked together via an IAX2 trunk and calling works fine between them. The issue is that the extensions at each location overlap. ie. System A has extension 100 - 150 and System B has extension 100 - 150 as well.
In order to call between systems, one precedes the extension with an 8. ie. on System A, dial 8100 to get to extension 100 on System B and vice versa. This works fine however the callerID Name shown is of the local system and not of the remote system.
Example
System A, ext 100 is Joe
System B ext 100 is Peter
If Joe calls anyone on System B, then the callerID number will be 100 but the callerID name shows as Peter instead of Joe.
I understand it is picking this name up from the local system but this is incorrect so is there anyway of preventing this local lookup fir the IAX trunk and passing the remote callerID name?
I have the routes set as Intra-Company routes and for the trunk to allow any CallerID.
Any thoughts on passing the correct callerID name?
↧
OLD Polycom Phones, Endpoint Manager, legacy_sip.cfg and Paging/Intercom
Lots of other threads on this, but I think I have finally (for me at least) figured out how to make this work and not have updates screw it up - without changing anything in the way FreePBX is set up (Including changing the database like here: http://issues.freepbx.org/browse/FREEPBX-6049)
The root problem is that the OLD Polycoms (301, 430, 501 601 and 4000 - we have LOTS of them in the field - they just don't die!) use legacy_sip.cfg to set Alert Info settings - here is the proper line as of 2016-06-15 and what FreePBX 13 is sending:
<alertInfo voIpProt.SIP.alertInfo.1.value="Alert-Info: Auto Answer" voIpProt.SIP.alertInfo.1.class="4"/>
This is the format that current code in /etc/asterisk/extensions_additional.conf sends to the Polycom phones for Intercom and for paging.
The problem came from someone adding this to the XML for these phones in Endpoint Manager - it's under the Basefile Edit - Ext.cfg for any of the above models:
<SIP
>
<voIpProt.SIP.alertInfo
SIP.voIpProt.SIP.alertInfo.1.class="ringAutoAnswer"
SIP.voIpProt.SIP.alertInfo.1.value="Auto Answer"
>
</voIpProt.SIP.alertInfo>
</SIP>This is what was killing all the other changes I was trying following the other posts - I finally had to resort to Wireshark to see what was (properly) being sent and then looking at the above syntax, and comparing it to legacy_sip.cfg to see that these lines are not properly formatted for the older Polycoms - it was overriding legacy_sip.cfg and therefore Paging/Intercom never worked on these phones - remove that whole section from the Basefile for these models, set legacy_sip.cfg to the above settings and you are (finally) in business.
↧
↧
Downgrade PhoneApps-Mandatory red MWI light not needed on voicemail app
I have moved my voicemail app button to one of the horizontal soft keys, so now I am no longer getting the MWI (except for the somewhat confusing temporary "Voice Mail enabled" message when a voice mail comes in).
↧
Setting for syslog server?
They support it but we have not documented or exposed it yet. The phone does keep a local syslog that can be downloaded from the GUI of the phone.
If you go into the phone GUI and export the config you will see the pcodes for syslog server and you can go into End Point Manager Base File and add your syslog server to apply it to all your phones.
↧
Setting for syslog server?
Thanks, I'll take a look.
↧
Is there a way to edit Digium phone settings from FreePBX?
If I recall correctly digium phones provision with http not ftp
↧
↧
Clearing "You have 2 tampered files" message
Trying to clear this one out. I'm 99% certain it's from when I was adding our SIP trunk provider (Flowroute), and first tried their Asterisk setup guide, later realizing I could do everything in the GUI. I've removed the modifications, but it seems FreePBX still thinks they have been altered. How can I restore the original files, or clear the message, so that it accurately alerts me to any future concerns?
Module: "Core", File: "/var/www/html/admin/modules/core/etc/extensions.conf altered"
Module: "Core", File: "/var/www/html/admin/modules/core/etc/sip.conf altered"
Thank you!
↧
Clearing "You have 2 tampered files" message
fwconsole ma refreshsignaturesmay also try
fwconsole ma upgrade core↧
Clearing "You have 2 tampered files" message
fwconsole ma refreshsignatures
That did it, thank you!
↧
Needing help setting correct trunks in freepbx13
that didnt work so I need some more on this. I have 2 working extension that can call each other just fine but I cant call in lets say from my cell phone to that number or I cant call out from one of the extension.
↧
↧
One way audio inbound calls / no audio from remote
Hi, I'm having problems getting around the following problem:
One way audio inbound calls / no audio from remote
I can place outbound calls without any issue, but if I receive a call, the other party can't hear my voice...
Also the inbound extension seems to ring endless after having picked up an inbound call.
Does anyone have a clue on how to get around this? I'm using freepbx with pfsense and a cisco 7961 phone
My setup:
PFSENSE firewall
(1 LAN & 1 WAN port)
external_sip_servers = all ips of my provider's servers
- portforwardings
wan interface
Rule1
source external_sip_servers udp 5060,5061,5062
destination pbx udp 5060,5061,5062
Rule2
source external_sip_servers udp 10000:20000
destination pbx udp 10000:20000
- AON - Advanced Outbound NAT
wan interface
Rule1
source pbx udp 10000:20000
destination external_sip_servers udp 10000:20000
nat address wan address
nat port *
static port Yes
Rule2
source pbx udp 5060,5061,5062
destination external_sip_servers udp 10000:20000
nat address wan address
nat port *
static port Yes
FREEPBX
Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5061
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: Off
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-13.0.74(11.22.0)
SDP Session Name: Asterisk PBX 11.22.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Trust RPID: No
Send RPID: No
Legacy userfield parse: No
Send Diversion: Yes
Caller ID: Unknown
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: 4294967295
SIP realtime: Disabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Store SIP_CAUSE: No
Network QoS Settings:
---------------------------
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Network Settings:
---------------------------
SIP address remapping: Disabled
Externhost: <none>
Externaddr: (null)
Externrefresh: 10
Localnet: 192.168.1.0/255.255.255.0
Global Signalling Settings:
---------------------------
Codecs: (gsm|ulaw|alaw|g726)
Codec Order: ulaw:20,alaw:20,gsm:20,g726:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: Yes
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 30
RTP Hold Timeout: 300
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: No
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Sub. min duration 60 secs
Sub. max duration: 3600 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Outbound reg. retry 403:0
Notify ringing state: Yes
Include CID: No
Notify hold state: Yes
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: from-sip-external
Record on feature: automon
Record off feature: automon
Force rport: Yes
DTMF: rfc2833
Qualify: 0
Keepalive: 0
Use ClientCode: No
Progress inband: Never
Language: de
Tone zone: <Not set>
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: *97
----↧
Storage space is getting critically high
Hello I am using FreePBX 13.0.131 and I have been receiving alerts every hour on this. I went through so old forums from a couple of years ago and after reading them I have upgraded all of my modules. I went through and checked if I had some recording and backups that I could delete and I don't see anything. Here is the output from CLI
[root@rs-pbx ~]# df
Filesystem 1K-blocks Used Available Use% Mounted on
/dev/sda2 75716712 68616248 3247616 96% /
tmpfs 956732 0 956732 0% /dev/shm
/dev/sda1 289293 26921 247012 10% /boot
[root@rs-pbx ~]# df -h
Filesystem Size Used Avail Use% Mounted on
/dev/sda2 73G 66G 3.1G 96% /
tmpfs 935M 0 935M 0% /dev/shm
/dev/sda1 283M 27M 242M 10% /boot
↧
Unable to direct dial conference/paging extensions
I have my IVR's setup so that I can use direct dial of extensions. It works well for calling individual endpoints, but if I dial a conference or paging extension, the call drops. I'm able to dial these numbers from the endpoints just fine, and our conference room IVR connects to the rooms fine as well. What should I look at first here?
↧