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Upgrading a Hyper-V FreePBX Machine to Sangoma-7 - You CAN get there from here!

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Better than the angry call from the customer because something doesn’t work - like last night at 5:15 as I was walking out the door - Machine I had migrated and I thought I had gotten everything right - the migration lost their custom MOH (which I still had, but didn’t know about until the complaint) and their Announcement that got switched to saying they were closed (A Pizza Place - So Friday night is KILLER).

Both were fairly quick to fix, but once again, our reputation for reliability takes a hit.

Yes this method takes a while, but the box comes back at 100% - No other solution comes close!


Upgrading a Hyper-V FreePBX Machine to Sangoma-7 - You CAN get there from here!

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I migrated last week using the script for the first time, I realized that MOH wasn’t moved over, it was another 15 seconds of work… Had in mind to file a bug report (or check if there is already)

I must say, that the migration was faster then expected, and the easiest PBX migration I ever had (they have no commercial modules)

Client didn’t notice a thing wrong or not working.

We just have to put together a list, or perhaps add to the documentation a list of everything which is currently not being moved over by the script.

How to: Upgrade FreePBX13 to FreePBX14 on Hyper V - Guide

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And another point which was just discussed over in another thread, MOH is also not being migrated.

Predial-hook for specific Outbound Route

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Thanks for the pointer on that, I’ve managed to use that. For benefit of future people finding this thread, my predial-hook now looks like this

[macro-dialout-trunk-predial-hook]
exten => s,1,GotoIf($["${OUTBOUND_ROUTE_NAME}" != "Test_Route"]?noMatch)
 same => n,Read(TICKET,"goodbye",,,3,5)
 same => n,Set(CHANNEL(hangup_handler_push)=out_ticket_hangup_handler,s,1) ;
 same => n,AGI(csm_number_check.php,${TICKET})
 same => n(noMatch),NoOp(Not a ticket related call)

Problem outbound and inbound :

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I think i have found :stuck_out_tongue:

Current PBX Version:

14.0.3.1

Current System Version:

12.7.4-1804-2.sng7

Problem outbound and inbound :

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Well, have you upgraded the core module under module admin?
Just upgrade all modules to the latest.

S705 Freepbx 14 looses registration

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LetsEncrypt as I cannot use a Comodo certificate at all with the Sangoma phones. That’s another issue I am having Sangoma phones won’t work with Comodo Certificate and Yealinks won’t work with LetsEncrypt. Polycom phones work with phone if there on firmware 4.0.12 and above.

Paging without the BEEP

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Do you still have your call trace when you were on v14.0.4? If so please open a BUG ticket here https://issues.freepbx.org/ so that they can fix the 14 branch. You can reference my ticket so they can hopefully and easily put the fix back in. Otherwise the bug will persist.

Thanks!


Disk i/o errors on FreePBX distro

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For what it’s worth, I have been using FreePBX or it’s forbears for longer than most, I can assure you that I always deploy a completely open-source system that does not need registration, licensing or any ‘commercial’ modules with almost no lack of function, If you need commercial support , please feel free to use it as it makes it easier for newbies and adds function (at a cost) , complaining that the Open-source parts of FreePBX are in any way not working or likely to go away is just plain wrong.

Your backup tarball contains whatever you chose to put in it, the mysql dumps, the web server and the etc/asterisk files and anything else you arbitrarily added to your backup schema , you should be careful restoring any part of it if you suspect that the original system was in any way ‘Broken’ ( as yours apparently was) and it is very easy to be particular as to what you want to restore.

JM2CWAE

Zulu mobile connectivity issue

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0x7faae8ed7d40 – Strict RTP learning after remote address set to: 76.23.183.29:53036
[2018-09-15 21:08:44] ERROR[14071]: pjproject:0 <?>: icess0x7faae8eef5d8 …Error sending STUN request: Network is unreachable
[2018-09-15 21:08:44] ERROR[24304]: pjproject:0 <?>: icess0x7faae8eef5d8 …Error sending STUN request: Network is unreachable
[2018-09-15 21:08:44] ERROR[24304]: pjproject:0 <?>: icess0x7faae8eef5d8 …Error sending STUN request: Network is unreachable
[2018-09-15 21:08:44] ERROR[24304]: pjproject:0 <?>: icess0x7faae8eef5d8 …Error sending STUN request: Network is unreachable
[2018-09-15 21:08:44] ERROR[24304]: pjproject:0 <?>: icess0x7faae8eef5d8 …Error sending STUN request: Network is unreachable

this is the errors I’m receiving

Zulu UC 3 and Mobile

SIP Trunk rejects and then won't reconnect

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This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.

Configuring a Panasonic KX-UDS124 DECT setup

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Hello

I am new here and also new to FreePBX. I successfully got FreePBX working with a Snom320 extension (calls and everything works fine). Now I am trying to get a Panasonic KX-UDS124 Multi-Cell DECT system with a KX-UDT121 handset connected to my FreePBX. Note that the basic Air setup is working fine, i.e. the handset is registered to the UDS124.

My local freepbx is reachable inside my network through the host name freepbx.local

These are the basic voip settings of the UDS124 (base station):

Handset settings:

My extension (202) is CHAN_SIP and listening on Port 5160 (UDP). All configs are applied.

I tried various settings, but am always getting either DNS error or a 403 response.

Has anyone gotten a Panasonic UDS124 setup working with FreePBX and would be willing to share their config?

Avaya 96x1 extended Features

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It looks like, that the NOTIFY is causing this issue.

I ran the Avaya phone and the software phone in parallel under the same phone number. Then I dropped a call from Softphone.
This NOTIFY message came then for both phones. Both phones then had also the “=” on the display.

This is the only entry with the IP address of the Avaya phone. The softphone has the IP address .20,
.16 is my FreePBX

I start now another try with 14.5.0. I saw that you also use this version.

— (7 headers 0 lines) —
Reliably Transmitting (no NAT) to 192.168.178.19:5060:
NOTIFY sip:303@192.168.178.19;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.178.16:5060;branch=z9hG4bK09580239
Max-Forwards: 70
From: sip:303@192.168.178.16;tag=as2a9f4571
To: sip:303@192.168.178.16;tag=1018db5b9cdd425b9ce664_F303192.168.178.19
Contact: sip:303@192.168.178.16:5060;transport=TCP
Call-ID: 3_71e60c5107b5b9ce47e_S@192.168.178.19
CSeq: 111 NOTIFY
User-Agent: FPBX-14.0.3.13(13.7.2)
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 204

<?xml version="1.0"?> confirmed

Unable to avoid direct media

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First, you need to redo this debug that you originally posted. You have it via a tcpdump, which is great and all but you need to do it via Asterisk’s command console so you can see if and how the packets are hitting Asterisk itself.

Second, DO NOT edit things out of the file. You’re having call/audio issues and you are MASKING the IPs that are being used. We have no idea if the right IPs or information is being sent via this because you’ve removed it.

Third, go actually configure your softphone client that you are testing with or whatever phone you are using so it’s not sending 12 codecs that aren’t being used at all as options to the PBX. Specially when some of those options are at the top of this list and probably not even enabled on the PBX.


Avaya 96x1 extended Features

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I am sorry I have been traveling and have not had a chance to dig in to this

There is code that handles the Notify issue. I will try to look at soon can you post you 2 modified files?

1 NIC - Need two gateways due to local sip

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I am working with Mark on this issue and have been doing some testing of my own. Here is what we have at our office. Two ISPs, with two routers. One router acting as DHCP server at 192.168.0.1/23 and the other at 192.168.0.2/23. there are two Proxy IPs that this SIP provider uses. They are 10.7.32.20 and 10.7.32.36. I created a new file at etc/sysconfig/network-scripts/route-eth0 with the entries like so:
10.7.32.20/24 via 192.168.0.2 dev eth0
10.7.32.36/24 via 192.168.0.2 dev eth0
I restarted the network with service network restart
Then a route -n provided this:

Kernel IP routing table
Destination Gateway Genmask Flags Metric Ref Use Iface
0.0.0.0 192.168.0.1 0.0.0.0 UG 0 0 0 eth0
169.254.0.0 0.0.0.0 255.255.0.0 U 1002 0 0 eth0
192.168.0.0 0.0.0.0 255.255.254.0 U 0 0 0 eth0

I also tried these entries in the route-eth0

ADDRESS0=10.7.32.20
NETMASK0=255.0.0.0
GATEWAY0=192.168.0.2
METRIC0=10
ADDRESS1=10.7.32.36
NETMASK1=255.0.0.0
GATEWAY1=192.168.0.2
METRIC0=11

Again restarting the network and running route -n got the same results. A traceroute to 10.7.32.20 goes out the 192.168.0.1 gateway.

The system I am using to test is a FPBX 14.0.3.13 64 bit
The system firewall is not enabled.
Everything I have read says this should work, but it isn’t. Did I miss something?

Thanks for any ideas

Configuring a Panasonic KX-UDS124 DECT setup

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Never mind, I had the most stupid mistake - had port 5060 in setup instead of 5160 - duh :grin:
Can confirm voice connection. Will try to move on now with config (voicemail, etc.).

No Audio Between 2 Extensions

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Did you ever find the solution? Could you post it?

Problems with Extensions at Remote Location

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If the remote location does not have its own PBX and they are connecting over the VPN to the PBX then this sounds like a firewall/NAT (possible) with the VPN connection and how it is handling new requests from the phones.

So let me ask this, when this happens and the remote location starts to have these issues what is the current resolution? Are you rebooting something? If so, what and at which location?

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