@snaplink wrote:

Hello!

I have a question about Templates in endpoint manager. Our company is roughly 50 users and almost everyone’s current phone (Allworx) system has a different BLF setup.

After playing around with the system am I wrong that each user will need their own template? I can’t seem to find where to change the BLF’s or Lines on a per handset method without going to the actual phone IP address and using the yealink configuration. Is there a way to edit by extension or another way other than giving each user their own template?

We are only going to use T46G’s and T46S’s in our environment

Thanks!

Posts: 3

Participants: 2

Read full topic

@its_errick wrote:

After updating system, when making outbound calls no ringback is heard. I have Trunk dial option set to r.

Currently on:
Google Voice/Chan Motif 13.0.3.2
Core 14.0.6

I’ve tried rolling back and using motif on edge track 13.0.4 deleting and adding google voice settings back in between changes with no luck. Calling out is still silent.

A side issue that started happening when ring back tone stopped was when inbound calls come in, i had it set to ring desk phone and external number (cell phone) at the same time. Now the desk phone rings once and calls the cell phone only, but now with no ring back tone.
This last part may be a separate issue.

Posts: 7

Participants: 2

Read full topic

@vespino wrote:

Is it possible to pass the original caller id to an external number that is part of a ring group (XXXXXXXXXX#)?

Posts: 5

Participants: 3

Read full topic

@ArtsNCrash wrote:

Wasn’t sure where to post this so if I’m in the wrong place, let me know.

FreePBX 14.0.2.14
Asterisk 13.20.0
Cisco SPA303 Software ver 7.6.2c

Initially, I could not get Call Pickup to work at all. I ended up enabling and disabling the option in Feature Codes and that seemed to set things right. I can use a Soft Phone to pickup my two SPA303 extensions.

The issue is that every time I try to pickup with *8 from the SPA303, I get back “Forbidden” on the display. If I try **ext I get “Invalid Number” on the display. When I watch the Asterisk CLI line, there is no activity.

So I’m thinking this has something to do with the Dial Plan on the SPA303. I’ve tried several times to input a dial plan value that should cover this but no luck. One would think adding the following would work but it doesn’t.

*x|**xxx

I only have three digit extensions.

This is the default dial plan on the SPA303

(*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)

Any thoughts would be great. Or references would be awesome as well. I’ve been all over the net without success.

Posts: 1

Participants: 1

Read full topic

@Luke1982 wrote:

I’ve encountered the problem where voicemail starts telling ‘you have one…’ and then hangs up. I know this problem is common when audio files are missing, so I checked the logs. Sure enough:

WARNING[21259][C-000003fb] file.c: Unable to open vm-INBOXs (format (ulaw)): No such file or directory

So I searched the entire system for that file (vm-INBOXs.ulaw), but no luck, only ‘vm-INBOX.ulaw’. So I just cp’ed it to duplicate the file into ‘vm-INBOXs.ulaw’ and now it’s working again. Still I’d like to know how this could happen. Anyone any ideas?

Setup is FreePBX 13 on a CentOS 6 VPS.

Posts: 2

Participants: 2

Read full topic

@jlsatt4 wrote:

I am receiving the following error in red:

"Some SSL/TLS Certificates have been automatically updated. You may need to ensure all services have the correctly update certificate by restarting PBX services "

It appears the lets encrypt certificate properly renewed. Browser access shows a current green lock. I checked the System Admin HTTPS settings, etc. Everything LOOKS correct. I even restarted the server.

Could someone please point me in the correct direction to correct this?

Thank you!
John

Posts: 4

Participants: 2

Read full topic

@bbb333 wrote:

Hello,

Do HAVE to pay the $25 to set up VPN? In the instructions that i followed, the VPN option is not showing up, but i am being asked if i would like to upgrade to sys admin pro…
Let me know, Thanks!

Posts: 2

Participants: 2

Read full topic

@Link wrote:

Just wanted to share a strange issue I am trying to wrap my head around and can’t figure it out for the life of me and was hoping someone has shared an experience that is similar to mine.

Scope of Issue: Currently when making an inbound call into my test SIP trunk it is getting a ring back to an extension when trying to receive the call it is met with a active timer that the call has initiated but the extension within my PBX still continues to ring and the call then fails but yet my extension still continues to ring until I forcefully reject on my physical extension. Outbound calling works as intended. I do have a pcap of this instance but didn’t know how to share it within this post.

Any help would be appreciated as I feel this simply might be a silly setting I am missing.

Thank you,

Posts: 3

Participants: 3

Read full topic

@GSnover wrote:

We have a client that we just moved their system offsite into the cloud (well, really our cloud) and they were having CONSTANT phones dropping off - PJSIP mostly remedied that (they stopped becoming unreachable) but when you would call in and hit a ring group, some of the phones in the ring group would not ring - it was driving me crazy - we tried adjusting things like qualifyfrequency, played with the firewalls, and even switched out some phones (they had some REALLY old Polycom’s) and nothing helped - just switched them to TCP and the problem has COMPLETELY disappeared! I am surprised to say the least - has anyone else seen this? Had bad experiences with using all TCP?

I think it’s a miracle, but probably only because I have been fighting this for days.

Anybody have any horror stories about using TCP instead of UDP?

Posts: 3

Participants: 3

Read full topic

@netphoneusa wrote:

I have a client where ALL incoming calls go to ring group # 651

My client would like to use a feature code to change all incoming calls to ring group 670

Each ring group has 5 or 6 extensions.

They have 10 DID’s that all point to ring group 651

She just wants to enter a code that will change to 670 and then a code to change back to 651

ANy ideas?

Posts: 3

Participants: 2

Read full topic

@snaplink wrote:

I’m so close to completing the configuration of my test FreePBX system but running into a few snags it seems at the end.

I have a single DID with flowroute that I picked up for testing purposes. I originally used SIP registration and was ultimately able to make and receive calls. After chatting with Flowroute they suggest using Host based route and disabling the SIP credentials. They were able to talk me through setting that up on their end of things but now I can’t get it going in FreePBX. Calling the DID now returns an error of “You have reached a non working number”

I’m getting a little confused with where I need to enter the Tech Prefix. I deleted the existing Trunk and started fresh with creating a new one. According to a document on the Flowroute support site it said I only need to enter it in the Outbound Dial Prefix field of the Dialed Number Manipulation Rules section. I did that and have no other info in the new Trunk except Trunk Name, Outbound CallerID, clicked Submit, Apply Changes and even restarted FreePBX like the document suggested.

https://support.flowroute.com/SIP_Trunking_and_Voice/Getting_Started/Statically_route_your_phone_number_to_a_host_system_for_inbound_calls

Has anyone done this successfully? What am I missing?

Posts: 2

Participants: 2

Read full topic

@enuro12 wrote:

I purchased CRM 25 year license in november. I did a fresh install of FreePBX and activated it. Then updated. It bombed out.

I tried resetting the hardware key in my account, but my new install just wont pickup the CRM modules? I understand i only get to do it twice and i’ve done it once, so i dont want to hose it.

Posts: 1

Participants: 1

Read full topic

@steve_pbuk wrote:

I have activated voicemail on an extension and set a password:

image.png762x659 37.8 KB

If I dial *97 it asks for the password.

However, if I press the voicemail button on a Sangoma S500 phone it lets me straight into the voicemail app.

Anyway I can password protect this?

Posts: 1

Participants: 1

Read full topic

@mbrady wrote:

When trying to discover a Vega 60 FXO or FXS gateway within FreePBX Vega Management module the discovery returns no results. Trying to manually add a vega gateway fails as well.

It fails with message “Failed to connect to ssh://admin:admin@x.x.x.x:22, transport not supported”

Vega Gateway Module log contains the same error as above.

Both the gateway and FreePBX instance are on the same subnet. I’ve tried entering the correct subnet info and trying to discovery the gateway, I’ve also tried to directly add the gateway in FreePBX. In both cases, I’ve tried the default gateway credentials of admin/admin and I’ve tried creating a new password on the gateway first, then trying to discover/add it. - Doesn’t give any different results.

Also tried the latest gateway GA versions 11 and 12

All results in the same transport error.

From the FreePBX instance, I can ping the gateways and ssh into them as well.

Current PBX Version:14.0.2.14
Current System Version:12.7.4-1804-1.sng7
Python 2.7.5 (default, Aug 4 2017, 00:39:18)
[GCC 4.8.5 20150623 (Red Hat 4.8.5-16)] on linux2

Looking in the source code, I can find the “transport error” output line but I don’t know python well enough to troubleshoot further. Has anybody else seen this before?

Posts: 1

Participants: 1

Read full topic

@caoquocai wrote:

Dear all,

I installed a free-PBX (version 14.0.2.14) + Digium Gate 2G402F (4 port) + two PSTN providers

Free-PBX server IP: 192.168.1.10
Digium gateway has two IPs: 192.168.1.11 and 192.168.1.12

I have two extensions: 6XXX and 7XXX
I need to configure outbound calls for extensions with: 7XXX through port 1 on Digium gateway and 6XXX through port 2 on Digium gateway. so I created two trunks in freePBX to Digium gateway:

first trunk:
trunk name: 6XXX
type=peer
secret=password
host=192.168.1.11
defaultuser=user
context=from-trunk

and second trunk:

trunk name: 7XXX
type=peer
secret=password
host=192.168.1.12
defaultuser=user
context=from-trunk

I created outbound route in freePBX and saw that extensions 6XXX go through trunk name: 6XXX and extensions 7XXX go through trunk name: 7XXX

I also created two call routing rules on Digium gateway. the outbound calls on trunk 6XXX go through port 1 and 7XXX go through port 2.

but I have issue, all outbound calls go through port 1 on Digium gateway.

Please help on this

Posts: 4

Participants: 2

Read full topic

@mkaye wrote:

this dashboard window has not updated for almost a month now
I am running v14 in a Hyper-V on Windows Server
everything is up-to-date

Uptime isn’t correct either - says last reboot was a month ago, but rebooted server yesterday
everything else seems OK

mark

Posts: 3

Participants: 2

Read full topic

@matthewljensen wrote:

I’m trying to set up the endpoint manager to configure our gxp 2170s. I haven’t been able to get option 66 working, so I’ve had to go into each phone to configure the config server as the freepbx server. This works fine. I type in the ip and it grabs the config file. But after that, I’m not able to give it any more configs. The config changes the server address to include the tftp://, and as far as I can tell, this is preventing it from grabbing any more updates. Is there a way to not include this prefix in the config? I know on another system I manage, epm doesn’t seem to add that prefix, so maybe I can fix this by downgrading epm. What are your thoughts?

Posts: 3

Participants: 2

Read full topic

@SilverNodashi wrote:

Please help.

We have an old analogue Panasonic KD-TDA100D with 20x analogue phones in the office. We’re moving some users to a new office. The two building have a good wireless link to share internet and network services like printers, servers, etc. The extensions on the Panasonic PBX run from 101 to 127 (some extensions are inactive). 10 of the staff need to go to the new building.

So I installed FreePBX 14.0.1.24 on a new server in the new building and we’re going to purchase some new SIP phones, and also setup something like Linphone or Zoiper on some of the users’ mobile phones. I have the extensions of the users moving into the new building setup in FreePBX and on their mobile phones (have not decided on SIP phones yet) and this works very well.

Then I have some Grandstream HT503, Grandstream HT802 and HT704 ATA gateways to extend their extensions to the new building via the VPN.

I don’t have access to the Panasonic PBX, nor can we replace it at this stage. I setup FreePBX and the Grandstream HT-503 FXO as per this article

The problem is, when I attempt to dial one of the existing extensions on the Panasonic PBX, using the FXO port, I get an engaged signal - on all the numbers I tried to phone, even though I know the phones are not engaged.

When calling 102, which still remains on the Panasonic PBX, from 112, which is a newly created extension in FreePBX, I get the following on the Asterisk console:

<— SIP read from UDP:192.41.100.31:5062 —>
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 197.184.153.199:5160;branch=z9hG4bK4e177ce2;rport=5160;received=192.41.100.240
From: “Mariska” sip:114@197.184.153.199:5160;tag=as17d893a0
To: sip:102%40112-1@192.41.100.31:5062;tag=270990891
Call-ID: 1dcd9e4a1ce2320f7a0d290c61d1992f@197.184.153.199:5160
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V2.0A 1.0.15.5 chip V2.2
Warning: 399 GS “All channels are in use”
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
— (11 headers 0 lines) —
– Got SIP response 486 “Busy Here” back from 192.41.100.31:5062
Transmitting (NAT) to 192.41.100.31:5062:
ACK sip:102%40112-1@192.41.100.31:5062 SIP/2.0
Via: SIP/2.0/UDP 197.184.153.199:5160;branch=z9hG4bK4e177ce2;rport
Max-Forwards: 70
From: “Mariska” sip:114@197.184.153.199:5160;tag=as17d893a0
To: sip:102%40112-1@192.41.100.31:5062;tag=270990891
Contact: sip:114@197.184.153.199:5160
Call-ID: 1dcd9e4a1ce2320f7a0d290c61d1992f@197.184.153.199:5160
CSeq: 102 ACK
User-Agent: FPBX-14.0.1.24(13.19.1)
Content-Length: 0

---

-- SIP/114-1-00000035 is busy

== Everyone is busy/congested at this time (1:1/0/0)
– Executing [s@macro-dialout-trunk:32] NoOp(“PJSIP/114-00000161”, “Dial failed for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 17”) in new stack
– Executing [s@macro-dialout-trunk:33] GotoIf(“PJSIP/114-00000161”, “0?continue,1:s-BUSY,1”) in new stack
– Goto (macro-dialout-trunk,s-BUSY,1)
– Executing [s-BUSY@macro-dialout-trunk:1] NoOp(“PJSIP/114-00000161”, “Dial failed due to trunk reporting BUSY - giving up”) in new stack
– Executing [s-BUSY@macro-dialout-trunk:2] PlayTones(“PJSIP/114-00000161”, “busy”) in new stack
– Executing [s-BUSY@macro-dialout-trunk:3] Busy(“PJSIP/114-00000161”, “20”) in new stack
== Spawn extension (macro-dialout-trunk, s-BUSY, 3) exited non-zero on ‘PJSIP/114-00000161’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, 102, 7) exited non-zero on ‘PJSIP/114-00000161’
– Executing [h@from-internal:1] Macro(“PJSIP/114-00000161”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“PJSIP/114-00000161”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,3)
– Executing [s@macro-hangupcall:3] ExecIf(“PJSIP/114-00000161”, “0?Set(CDR(recordingfile)=)”) in new stack
– Executing [s@macro-hangupcall:4] NoOp(“PJSIP/114-00000161”, " monior file= ") in new stack
– Executing [s@macro-hangupcall:5] AGI(“PJSIP/114-00000161”, “attendedtransfer-rec-restart.php,”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
– <PJSIP/114-00000161>AGI Script attendedtransfer-rec-restart.php completed, returning 0
– Executing [s@macro-hangupcall:6] Hangup(“PJSIP/114-00000161”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 6) exited non-zero on ‘PJSIP/114-00000161’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/114-00000161’
– PJSIP/114-00000161 Internal Gosub(crm-hangup,s,1) start
– Executing [s@crm-hangup:1] NoOp(“PJSIP/114-00000161”, “Sending Hangup to CRM”) in new stack
– Executing [s@crm-hangup:2] NoOp(“PJSIP/114-00000161”, “HANGUP CAUSE: 17”) in new stack
– Executing [s@crm-hangup:3] ExecIf(“PJSIP/114-00000161”, “0?Set(__CRM_VOICEMAIL=)”) in new stack
– Executing [s@crm-hangup:4] NoOp(“PJSIP/114-00000161”, “MASTER CHANNEL: 1523829540.407 = 1523829540.407”) in new stack
– Executing [s@crm-hangup:5] GotoIf(“PJSIP/114-00000161”, “0?return”) in new stack
– Executing [s@crm-hangup:6] Set(“PJSIP/114-00000161”, “__CRM_HANGUP=1”) in new stack
– Executing [s@crm-hangup:7] AGI(“PJSIP/114-00000161”, “sangomacrm.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
– <PJSIP/114-00000161>AGI Script sangomacrm.agi completed, returning 0
– Executing [s@crm-hangup:8] Return(“PJSIP/114-00000161”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘PJSIP/114-00000161’
– PJSIP/114-00000161 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=
Really destroying SIP dialog ‘1dcd9e4a1ce2320f7a0d290c61d1992f@197.184.153.199:5160’ Method: INVITE
[2018-04-15 23:59:12] NOTICE[20338]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request ‘REGISTER’ from ‘“Server Room” sip:144@192.41.100.240’ failed for ‘192.41.100.160:5060’ (callid: aa15fbe-89f29be1-13603d7c@192.41.100.160) - No matching endpoint found
[2018-04-15 23:59:12] NOTICE[8340]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request ‘REGISTER’ from ‘“Server Room” sip:144@192.41.100.240’ failed for ‘192.41.100.160:5060’ (callid: aa15fbe-89f29be1-13603d7c@192.41.100.160) - No matching endpoint found
[2018-04-15 23:59:12] NOTICE[8340]: res_pjsip/pjsip_distributor.c:649 log_failed_request: Request ‘REGISTER’ from ‘“Server Room” sip:144@192.41.100.240’ failed for ‘192.41.100.160:5060’ (callid: aa15fbe-89f29be1-13603d7c@192.41.100.160) - Failed to authenticate

The FreePBX Server IP is 192.41.100.240. The Grandstream FXO is on 192.41.100.31

fxo1.jpg1366x768 104 KB

fxo2.jpg1366x768 96.1 KB

fxo3.jpg1366x768 92.5 KB

Posts: 4

Participants: 2

Read full topic

@cphtdk wrote:

Hi,

We had to “upgrade” to the commercial endpoint manager as the OSS was no longer getting regular updates.

I’m able to provision the phones OK, but they (Snom 720) ignore the custom dialplan, referenced within the basefile.

</phone-settings>

Reverting to the old files backed up from the OSS version, the phones load the dialplan file OK.

Brgds
Tom

Posts: 1

Participants: 1

Read full topic

@bajramia wrote:

HI All,
I have purchase Appointment Reminder i have configured I have issue is calling before the day start time the status is Not Running and still making calls is wakening the customers at 05:00 AM when it should not call at that time the start time is 09:00 till 17:00 M-F and Saturday and Sunday is from 11:00 till 17:00

Thank you

Posts: 2

Participants: 2

Read full topic