OK..

Just tried this again and it did work. Great, thanks. I could have sworn I had tried that before though....

Does that status only show how many calls are on a trunk, or does it also show the incoming call information?

Thx

The trunks are all SIP...

-------------------------

SIP/2000]
type=extension
extension=2000
context=from-internal
label=Seemore Phone1
mailbox=2000@default
extenvoicemail=*2000@from-internal

[SIP/2500]
type=extension
extension=2500
context=from-internal
label=Seemore Phone2
mailbox=2500@default
extenvoicemail=*2500@from-internal

[SIP/2501]
type=extension
extension=2501
context=from-internal
label=Seemore Phone3
mailbox=2501@default
extenvoicemail=*2501@from-internal

[SIP/2006]
type=extension
extension=2006
context=from-internal
label=Seemore Phone4
mailbox=2006@default
extenvoicemail=*2006@from-internal

[QUEUE/9000]
type=queue
label=Telemarketer
extension=9000
context=from-internal

[ViaTalk]
type=trunk
label=ViaTalk

[VoipDiscount]
type=trunk
label=VoipDiscount

[FreeDigits]
type=trunk
label=FreeDigits

[CONFERENCE/100]
type=conference
label=Conference 100
extension=100
context=from-internal

Hello,

Yes I see the buttons, and I only have 10 total buttons configured. The trunk buttons just do not show any calling information? The extension buttons work. I have yet to check the conference or que buttons.

Do you see trunk buttons at all? If you do not see them, maybe you hit the 15 button limit on the free version.

Best regards,

--
Nicolás

I have just set up the fop2, and like it. I am having an issue where I don't see the status of trunks.

It is configured properly, I believe. Where would any error logs be?

Thx

Ok thanks for the info.

I thought it would work, and I do the same thing today with the original FOP. My preference is to connect to port 80 thru ssh as well, as opposed to ssl. This way, I only open port 22 and tunnel/encrypt everything thru that port and the PBX interface is not directly exposed to the Internet.

I ask so that I can use this securely via the Internet. I don't want it opened to the public.

I know port 80, or port 443 for web, but the client to server, Is it restricted to port 4445, or are there others.

Also, I would prefer to be able to tunnel the traffic with ssh. II would use ssh and map ports

80 or 443 (without/with SSL)
4445

from my local machine thru ssh to the asterisk server..

any reason this approach would not work?

Hi,

If asterisk is segfaulting because of callevents in sip.conf, then turn it off.. you will loose hold status on fop, but the pbx will be stable :)

Now, I have not seen any segfault due to that sip option. What asterisk version are you running? Maybe updating asterisk will fix the issue...

Best regards,

I am running Asterisk 1.4.21.2

I'll test it a bit more this evening. I am running this as a VM with ESXi 3.5u4, but that shouldn't really matter.

Before I installed FOP2, I just made a snapshot, to make it easier for testing, adding, and removing, etc. Plus, when it gets late at night, I can just move back and forth between the images quickly.

If I redo this, test, and get the segfaults, are there any logs you want, or any debugging you would like me to enable? Just trying to help out...

I followed these instructions :
-----------------------------
/etc/asterisk/sip.conf

callevents=yes

It will send hold/unhold events to FOP2.
----------------------------------
This caused Asterisk to segfault.

I removed the 'callevents=yes' from sip.conf, and no segfaults.

I am running :

PBX in a Flash Version 1.2 Daemon Status

• *******************************************************************
• Asterisk * ONLINE * Zaptel * ONLINE * MySQL * ONLINE *
• SSH * ONLINE * Apache * ONLINE * Iptables * ONLINE *
• Fail2ban * ONLINE * IP Connect* ONLINE * Ip6tables * ONLINE *
• BlueTooth * ONLINE * Hidd * ONLINE * NTPD * OFFLINE *
• Sendmail * ONLINE * Samba * OFFLINE * Webmin * ONLINE *
• Ethernet0 * ONLINE * Ethernet1 * N/A * Wlan0 * N/A *
• *******************************************************************
• Running Asterisk Version : Asterisk 1.4.21.2
• Asterisk Source Version : 1.4.21.2
• Zaptel Source Version : 1.4.12.1
• Libpri Source Version : 1.4.7
• Addons Source Version : 1.4.7
• *******************************************************************

pbx.local on 192.168.6.159 - eth0
CentOS release 5.2 (Final) :64 Bit Kernel: 2.6.18-92.1.10.el5
********************************************************************
Need any additional info?

I'm guessing that your "#exec autoconfig-users-freepbx.sh" is overwriting your users with the freepbx ones try commenting that line out, restart fop2 and then try login to the panel again

I had the same issue.

I'll try this solution as well.