Setting up buttons for SIP Trunks

petempetem
in Support
Hi,

I am having problems with getting any of the "auto" configure options working, so I have created a manual buttons.cfg and it works fine. Very nice implementation by the way!

I have some SIP trunks and do not know how to set them up, I have one from one vendor and I have another from another vendor. I would like to see calls come in and go out on the appropriate trunks, is that possible?

This does not work:
[SIP/voicenetwork-out]
type=trunk
label=Voice Network
channel=SIP/voicenetwork-out
server=1

[SIP/from-ipcoms]
type=trunk
label=IP Comms
channel=SIP/from-ipcoms
server=1

Any direction is appreciated.

--Pete

Comments

  • nicolasnicolas
    FOP2 logic Is really really simple, if the channel name matches.. it will work. The most common source of problems is that many people have incoming calls from voip providers coming as anonymous calls (and not matching a peer definition in sip.conf). When that happens, channel names in asterisk are formed with the source ip address. If you know the ip addresses the call comes from from provider A and its always the same and different from the one of provider A, then you can add as many channel lines as you want adding those names. But the ideal way would be to have incoming calls matching a peer name in sip.conf and really differentiate between anonymous sip calls and provider sip calls.

    [SIP/myprovider]
    type=trunk
    channel=SIP/1.2.3.4

    The above will match channels like:

    SIP/myprovider-1000ed3
    or
    SIP/1.2.3.4-0003aaf3

    but it will not match with:

    SIP/5.6.7.8-aa00bb

    So, do a "core show channels" in the asterisk cli when you have incoming and outgoing calls going through your provider, remember the channel names and configure fop2 accordingly.

    Best regards,
  • petempetem
    Thanks for the fast reply, however, I am still unsure of what to do here. I have tried a bunch of options and nothing seems to work. From the Asterisk CLI in type core show channels while a call is coming in from SIP provider A and I get:

    SIP/NNNNNNNNNN-LNLNL

    I don't know enough to know if the actual numbers are a security issue, so the above is N=a number, L=a letter

    Then I create a buttons.cfg entry like this:
    [SIP/VoiceNetwork]
    type=trunk
    label=Voice Network
    channel=SIP/NNNNNNNNNN-LNLNL

    I still get no information from the web client.
    I take it "Label" is just that, could be anything, it's what shows up on the button
    What "exactly" is "[SIP/VoiceNetwork]" Is this the context? Where would I get it from to be 100% sure it's right?

    I know the IP address of the vendor, it's static, let's say it is 1.2.3.4, How do I create the buttons.cfg entry then? I tried this:
    [SIP/VoiceNetwork]
    type=trunk
    label=Voice Network
    channel=SIP/1.2.3.4

    It didn't work

    Sorry I'm such a newbie, I am trying to learn. Thanks,

    --Pete
  • nicolasnicolas
    SIP/NNNNNNNNNN-LNLNL

    The number after the - sign is the session. That must be removed from the channel name as it is a random string for each call.

    The channel naming convention in Asterisk is TECH/NAME-SESSION, fop2 needs TECH/NAME as session will be different for each call.

    Change "channel=SIP/NNNNNNN-LNLN" to "channel=SIP/NNNNNNN"


    However, what I said first stands: If your NNNN is not just numbers but also dots, then thats an IP address. Calls from your provider are then coming as anonymous calls, but they are not. If your voip provider changes the ip, fop2 will not find a match and that call will not be shown. If you can configure the asterisk sip peer so inbound calls are matched, you will see inbound calls as SIP/voiceNetwork.. This problem is outside of fop, as it is an asterisk/voip provider configuration issue.

    Best regards,
  • petempetem
    Great,

    That took care of it. Working perfectly now...

    Thank you very much,

    --Pete
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