I created a second sip trunk that my provider has as a backup. When I disabled the first trunk and then made calls, it would not display anything in the trunk. It would show the call on the extension but not in the trunk. Any ideas?
Thanks
-Dimitry
I created a second sip trunk that my provider has as a backup. When I disabled the first trunk and then made calls, it would not display anything in the trunk. It would show the call on the extension but not in the trunk. Any ideas?
Thanks
-Dimitry
You do not provide enough information. Ideas? The trunk device name in your button config does not match the device name in Asterisk. Do a "core show channels" in the asterisk CLI when you have ongoing calls on that trunk, and compare to the button config in fop2 (between brackets). Just an example, you might have a trunk named:
[SIP/myprovider]
type=trunk
label=myprovider
And when you receive a call, in asterisk you see:
SIP/x.x.x.x-yyyyyyy
Instead of
SIP/myprovider-yyyyyyy
In that case, you sip peer definition is incorrect for matching inbound calls, or your dial string is also incorrect as it is not dialing the peer by name but by ip.
Best regards,
I guess I am confused about the trunk display in fop2. I have 4 people on the phone because I can see 4 active extensions in fop2 showing phone calls with outside lines. In the trunks it shows 3 lines in use and only one call with details.
Does trunks only show outgoing calls? Or should it show both incoming and outgoing?
I thought trunk is supposed to shows call details. It used to show the details of the phone number in the trunk in version 2.10 but after I upgraded to version 2.11, it shows the following:
3 lines in use
SIP/bw_outbound_pri-08c44648
Any ideas?
Thanks
-Dimitry
Hello,
Trunk buttons will display any type of call, inbound or outbound, and details (callerid and timer) for any bridged call.
If you do not see inbound calls, it might be because the inbound trunk device in asterisk does not match the button definition in fop (as I tried to explain in the previous post).
Do a "core show channels" and look for device names when you have inbound/outbound calls, they should be the same if you want to monitor them in just one trunk button.
Best regards,
I am not an Asterisk expert so excuse my ignorance.
When I look in my asterisk logs it shows:
[2010-10-04 11:55:05] VERBOSE[22217] app_dial.c: -- Called bw_outbound_pri/+1443XXXXXXX (X are real numbers)
[2010-10-04 11:55:05] VERBOSE[22217] app_dial.c: -- SIP/bw_outbound_pri-08c449c8 is ringing
[2010-10-04 11:55:06] VERBOSE[22216] pbx.c: > Channel SIP/bw_outbound_pri-08bd8928 was answered.
but in fop2, it shows SIP/bw_outbound_pri-08bd8928 for all calls (meaning bw_outbound_pri-some numbers but not telephone numbers), what am I doing wrong?
In the extensions, it shows the phone numbers correctly.
Thanks
-Dimitry
Hello,
I did not understand what you see in fop2 and what you expect to see. The trunk name is:
SIP/bw_outbound_pri
The random numbers are not accounted in fop2 to match the channel.. so if you have a trunk button like
[SIP/bw_outbound_pri]
type=trunk
label=outbound pri
It will show the call you posted in that button...
Nicolás,
Could you explain your reasoning for only showing trunk details for Bridged Channels as opposed to all the time?
Thanks
-Dimitry
It was implemented that way for data consistency, as callerid is proper only when the call is bridged or there is a Dial manager event sent. It is possible to display all lines going trough, but many will complain that the callerid is not shown, or not correctly, etc.
Perhaps the next release will display details on all trunks, but I was busy on implementing some other and important features in the last couple weeks.
Thanks for your post.. This helped me solve my incoming from SPA-3000 on FOP.
My incoming trunk User Context is 289-8834-IN but in the CLI command showed
Channel Location State Application(Data)
SIP/115-08c07a58 1105@from-internal:1 Ringing AppDial((Outgoing Line))
SIP/114-08d4acc8 1105@from-internal:1 Ringing AppDial((Outgoing Line))
SIP/8834-b7544270 s@macro-dial:7 Ring Dial(SIP/114&SIP/115,25,trM(au
So i changed in the Panel conf file the from [SIP/289-8834-IN] to the Username of the PTSN line. In mine it is 8834.
[SIP/8834]
It now shows incoming calls on FOP
I have 3 sip trunks
1 inbound
1 outbound
1 bidirectional
inbound works great
outbound - shows the sip/trunk-hex# of call and either s or blank
directional shows inbound/outbound mostly correct, but occasionally will show the correct hex code, but the wrong outbound information
my buttons.cfg has
[att_trunk1]
type=trunk
label=att-in
channel=SIP/att_trunk1
[att_trunk2]
type=trunk
label=att-out
channel=SIP/att_trunk2
[telnet]
type=trunk
label=telnet
channel=SIP/telnet