@mattman110 wrote:

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 ?

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@amztransit wrote:

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!

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@VoIPTek wrote:

Hello All,

I still have a few boxes I need to convert to the commercial EPM, but can't at the moment with this server.

I was changing from TFTP to HTTP, this requires you to update your upgrade policy and in Advanced OSS EPM configuration and update the configuration type. When you do it, it appears to save, however I believe it's not really updating the tftp files for cisco spa phones ( at least in my initial testing ).
I tried to write all the configuration files from the device manager, but no luck.

I still see: tftp://10.1.10.10/spa$MA.xml
rather than something like: http://10.1.10.10:84/spa$MA.xml

I believe it's saving the configuration value of the type, but not updating globals.

There is no longer a submit type button, previously called "update globals"

Any suggestions?

Thanks!

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@brk wrote:

I was trying to create a Let's Encrypt certificate. I have the latest updates, which mention the firewall being configured automatically. I have no other external firewalls.

I am getting the following message.

There was an error updating the certificate: openssl_csr_new(): dn: add_entry_by_NID 16 -> (failed)
 New Certificate

Where to look for more details?

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@a12342398 wrote:

Hi, I'm having problems getting around the following problem:

One way audio inbound calls / no audio from remote

1. I can place outbound calls without any issue, but if I receive a call, the other party can't hear my voice...

2. 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

----

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@andersonhaulage wrote:

I have a stable installation. This morning I walked in and everybody was bitching. Phones were down. We could make and receive calls, but had no audio, in or out. internal or external. Just for laughs, I rebooted my endpoint, and it started orking. Sent out a reboot to all phones, and they all started working.
Does anybody have an idea what could cause something like that, and could it be solved without rebooting the phones?

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@flavionyc wrote:

When I access my freepbx 13 via SSH the first thing that loads is the below. Wondering what is wrong?

PHP Warning: include_once(/etc/asterisk/freepbx.conf): failed to open stream: Permission denied in PHP Warning: include_once(/etc/asterisk/freepbx.conf): failed to open stream: Permission denied in /var/lib/asterisk/bin/fwconsole on line 13
PHP Warning: include_once(): Failed opening '/etc/asterisk/freepbx.conf' for inclusion (include_path='.:/usr/share/pear:/usr/share/php') in /var/lib/asterisk/bin/fwconsole on line 13
PHP Fatal error: Class 'Symfony\Component\Console\Application' not found in /var/www/html/admin/libraries/FWApplication.class.php on line 11
[ec2-user@skynet ~]$

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@flavionyc wrote:

I started using freepbx13 and my cisco phone SPA504G is upgrading its firmware too often. I would say 2 times a day. The displays " Upgrading Firmware Please do not unplug power Phone will reboot automatically"

My cisco phone did not have this behavior on my older version of Freepbx

Under Settings>EndpointManager>Cisco>template there a line that says "firmware version = Firmware slot1" I'm wondering if here I should change it to "firmware version = none"

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@ou812 wrote:

I had a system that was FPBX12/Asterisk11 I upgraded the system to FPBX13/Asterisk13. The system works fine just every couple days we can not access the gui, if I do a service httpd restart all is fine for a couple more days, it's been like this for about a month now, this is what I see when we try to access.

Whoops! There was an error. - http___192.168.125.11_admin_config.php.png909x1606 66.3 KB

does something need to be reinstalled.

Regards,

Gary.

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@josephchrz wrote:

Hello i got a new message in asterisk -rvvv when i was monitoring it said == Spawn extension (from-sip-external, h, 1) exited non-zero on Not sure what that is. Can someone help me out please?

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@NUB wrote:

Hello Everybody,

I have a strange Problem with our IVR Module.
Our IVR is perfectly working, meaning when I dial 1 or 2 to get to the setup IVR destination it is working well.
When I dial a IVR Destination which is not setup, I get "we have not received a valid response" which is supposed to be like this. Also when I dial an extension, its getting routed to the extension.
But, when i dial a wrong Number which is longer the 2 digit, the System hangs up, unless I dial this Number and get put a # behind, then I get the write announcement "we have not received a valid response" again.
The Error I get, when I dial a longer Number then 2 digit in the IVR is:

[2016-06-19 19:03:34] WARNING[27115][C-000000d6]: pbx.c:6763 _astpbx_run: Channel 'SIP/CN-TCom-0000011c' sent to invalid extension but no invalid handler: context,exten,priority=from-did-direct,875,1

It was working well in the past and I just came by chance across this Problem when I added new Music to the System and started some Test's ...

Anybody experienced something similar or has any idea where I could look ...

thanks for your help,
Cheers, NUB

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@mydxbtester wrote:

I have a working FreePBX machine using the latest version. All is working fine from within the LAN and I wanted this machine to be accessible from the outside.
I have a static public IP address in place so I wanted to be able to communicate from users within the LAN when I'm on the road.

I'm successful connecting to our FreePBX machine via PPTP VPN with Zoiper but I would like to access it via "normal" communication since I have the static IP address.

Can someone point me to a "How-To" on how I can make this work.

Thanks.

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@tjgertge wrote:

So I've done this on dozens of systems, but this is the first time I've tried to setup HTTPS since the new GUI for HTTPS Setup in the system admin module.

I went to certificate manager, generated a new self signed cert. Went back to the HTTPS Setup in the system admin module, selected my new cert and hit install. Everything looks like it installed properly there.

I try to connect to this PBX using https now and I get the following:

"x.x.x.x normally uses encryption to protect your information. When Google Chrome tried to connect to x.x.x.x this time, the website sent back unusual and incorrect credentials. This may happen when an attacker is trying to pretend to be x.x.x.x, or a Wi-Fi sign-in screen has interrupted the connection. Your information is still secure because Google Chrome stopped the connection before any data was exchanged.

You cannot visit x.x.x.x right now because the website sent scrambled credentials that Google Chrome cannot process. Network errors and attacks are usually temporary, so this page will probably work later."

This used to be pretty straight forward Freepbx 12, so I'm not sure where the disconnect is.

Freepbx: 10.13.66-12
Cert Man: 13.0.22
Sys admin: 13.0.57.8

This is holding up the roll out of two new systems so I need to get this figured out. Thanks.

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@fransh wrote:

I want to have the number of the original caller displayed when the call is transferred via attended transfer. I have set the Send RPID field of both handsets to "send remote party ID header". As well as the RPID field of the general setings to yes. This works for unattended transfer but not for attended transfer. So I have two extensions 30 and 31. A call comes in from 06123456 on extension 30. 30 transfers to 31. After the call has been transferred 30 is displayed in the screen of 31. I want to have 06123456 displays. How can I configure that?

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@bigj4155 wrote:

This phone does not seem to be officially support in the latest end point manager. When I goto network scan it does not return any phones because it doesnt have the CP7960G template only a CP7960 which does work but I have to manually enter mac info ect to register the phones. Anyone have a info on how to get network scan to find a cisco 7960G phone?

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@clarkdv wrote:

I've setup a separate Inbound Route for each of the most popular toll free prefixes, 800, 822, 833, etc. but they're not working.

I just received a call from a 877 number even though I have one of my Inbound Routes set to CallerID Number _877NXXXXXX and Set Destination is Terminate Call - Play SIT Tone (Zapateller)

Did I forget a setting needed to make this work?

Thanks!

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@AndreL wrote:

Hi,

My Inbound route is going to a call flow control.

Until version 13.0.131 was ok but with version 13.0.137 calls are going to blackhole.

How to troubleshoot that situation ?

Thanks.

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@JeffroJones57 wrote:

Just did a fresh install with FreePBX 13 and Asterisk 13 using the latest download as of yesterday 6/20. Internet connection was fine and said it was updating all modules. Logging into the GUI there are over 60 modules disabled pending upgrade. Is this normal? I can manually go in and upgrade each one, but seems like i shouldn't have to do this.
Thanks

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@ladiko wrote:

All my extensions are reported as being in use instead of being busy via AMI action
http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+ExtensionState and https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_PJSIPShowEndpoint and in FreePBX GUI http:///admin/config.php?display=asteriskinfo#peers

Regarding the first link, it should be reported as

• in use if "One or more devices INUSE"
• busy if "All devices BUSY"

I have two phones which register to the same PJSIP-Extension. I therefore set "Max Contacts = 2" and rebooted all and should be able to get "In use" if at least one phone is used and busy if both phones are used. In any case, i get "In Use". I also checked another Extension with "Max Contacts = 1" and one registered phone and should report to be "Busy" AND "In Use", but it doesn't !?

I added the following to /etc/asterisk/pjsip.endpoint_custom.conf and reloaded asterisk, but it still reports in use.

[700]
device_state_busy_at=1

I also tried to add it to /etc/asterisk/pjsip.endpoint_custom_post.conf - same result. The AMI action https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_PJSIPShowEndpoint also reports it to be in use instead of busy

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@KaAzZ wrote:

Hi all,

Since few days , I registered my trunk with my SIP provider , inbound and outbound calls were working quite well but I configured the "inbound DID" with "Any" redirected to my extension/ring group. We want add some numeros and when I try to add my number in the "Inbound DID" , inbound calls don't work. I have a "DID not match" in the Asterisk console.

So , I called my SIP provider to be sure of the format of the num when relayed , so I tried with the international/national format and more ( 0033XXXXXXXXX, 0XXXXXXXXX, even with patterns but nothing is working I can't figure it out )

I hope someone had the same trouble, thanks for reading and have a nice day.

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