Member
Last active 14 years ago
Hi Nicolas,
After some more experimentation, here are my finding:
Trixbox 2.6 + FOP 2.11
Inbound route straight to extension:
FOP2 shows calling number at extension, Calling number and name shown on pop-up.
Inbound route to queue:
FOP2 shows calling number on all extensions which are queue agents. Shows calling number and name in queue. Shows calling name and number in pop-up.
Trixbox 2.8 + FOP 2.11
Inbound route straight to extension:
FOP2 shows [u:3bfxb89z]called[/u:3bfxb89z] extension number at extension e.g ext 502 shows 502 calling. No pop-up.
Inbound route to queue:
FOP2 shows [u:3bfxb89z]called[/u:3bfxb89z] extension number at extension. Shows [u:3bfxb89z]calling[/u:3bfxb89z] number [u:3bfxb89z]and[/u:3bfxb89z] name in queue. No pop-up.
(Each extension shows own number as calling number e.g. ext 511 shows 511 as calling number, 512 shows 512)
I can only assume that FOP2 must be being given the calling number as it appears in the queue, but why isn't the extension showing the calling number (only shows own extension number calling) and there's no pop-up?
Thanks
Carlos.
Just to add a little more!
I realised that I needed to stop the fop2 server and restart with -X 15 . Silly me!
I've also noticed that internal calls show:
500 Carlos
Line1 device
Line2 inactive
501 Mobile
Line1 501
Line 2 inactive
when calling extension 501 from extension 500
What's the "device" mean?
Carlos
Sorry, forgot to say that netstat -nta gives:
0.0.0.0:4444 0.0.0.0:* LISTEN
0.0.0.0:4445 0.0.0.0:* LISTEN
as I changed FOP1 to port 4444
Carlos
Output of autoconfig-buttons-freepbx.sh is normal.
However /usr/local/fop2/fop2_server -X 15 generates: Can't listen to port 4445
Tried to search for this on forum, but the search seems to be not working at present. Regards this as too common words.
Thanks,
Carlos
Hi,
Yes. Received "Connection to manager OK!"
Carlos
Installed FOP2 2.11 on Trixbox v 2.8.0.5 and am getting no recognition of incoming calls in FOP2. I've used the visual phonebook database as a lookup source for Trixbox and that's picking the correct record to display on the phone. No SQL query originated by FOP2 though. Extension simply shows inactive on both lines.
Tried to compare this to a working setup using Trixbox 2.6 and can't see any difference in my configuration. Any ideas on what's going wrong or how I can investigate further?
Thanks
Carlos.
Thanks Nicolas,
You're correct, there's no appropriate output. Checking the MySQL databases, there's no such table as 'fop2users' within the 'asterisk' database, hence autoconfig-users.sh produces no output other than the button file.
How was that supposed to be added? During the installation of FOP2?
Cheers,
Carlos.
Hi,
Just upgraded to v2.11 and find that I can't get a login as an extension. FOP2 is running on port 4445 and appears to be OK, however I enter the extension number and password on starting the flash client, then it attempts to connect to the server but fails. fop2_server -X 15 gives:
192.168.200.33 => <?xml version="1.0"?>
<!DOCTYPE cross-domain-policy SYSTEM "http://www.macromedia.com/xml/dtds/cross-domain-policy.dtd">
<cross-domain-policy>
<allow-access-from domain="*" secure="false" to-ports="4445" />
</cross-domain-policy>
No flash clients connected
192.168.200.33 <= <msg data="GENERAL|contexto|1|" />
192.168.200.33 => { 'btn': '0', 'cmd': 'key', 'data': 'ZiQmv0sh6uNJNcm7WbgeuLpCg', 'slot': '' }
192.168.200.33 => { 'btn': '0@GENERAL', 'cmd': 'version', 'data': '2.11', 'slot': '' }
192.168.200.33 <= <msg data="1|auth|501|e1a60eeb9f4026059a177f3f3e22bb4e" />
192.168.200.33 => { 'btn': '0', 'cmd': 'incorrect', 'data': '0', 'slot': '' }
No flash clients connected
Any ideas?
Cheers,
Carlos
Hi Nicolas,
Just installed Asterisk 1.4 in order to test this. The equivalent output is:
127.0.0.1 <- Event: Newchannel
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: SIP/87.127.240.98-08595bf8
127.0.0.1 <- State: Down
127.0.0.1 <- CallerIDNum: 07*******69
127.0.0.1 <- CallerIDName: 07*******69
127.0.0.1 <- Uniqueid: 1266495006.279
127.0.0.1 <- Server: 0
127.0.0.1 <- Event: Newstate
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: SIP/87.127.240.98-08595bf8
127.0.0.1 <- State: Ring
127.0.0.1 <- CallerID: 07*******69
127.0.0.1 <- CallerIDName: 07*******69
127.0.0.1 <- Uniqueid: 1266495006.279
127.0.0.1 <- Server: 0
The main difference between this and Asterisk 1.6 appears to be the "CallerIDName:" being presented as the caller ID in v1.4, but presented as the CID Name Prefix + caller ID in v1.6.
Could this be why it's working in 1.4 and not in 1.6?
Carlos.
Hi Nicolas,
Just a thought, but could it be that the problem lies here:
127.0.0.1 <- Event: NewCallerid
127.0.0.1 <- Privilege: call,all
127.0.0.1 <- Channel: SIP/87.xxx.xxx.242-09ea55c0
127.0.0.1 <- CallerIDNum: 017******00
127.0.0.1 <- CallerIDName: Main Number - 017******00
127.0.0.1 <- Uniqueid: 1266074985.21
127.0.0.1 <- CID-CallingPres: 0 (Presentation Allowed, Not Screened)
127.0.0.1 <- Server: 0
This is the output from Asterisk 1.6, and I think that the 1.4 version would output:
127.0.0.1 <- CallerID: 017******00
Rather than CallerIDNum:
Could this be the issue?
Carlos.