Transfer calls impossible.

[Tobey]

Hi all, I am a beginner user of Fusionpbx, I would like to describe here my problem, which for you may turn out to be trivial (which I very much hope), below I will present my configuration, which I made originally, which does not work - I ask here for any advice/observations.
Specifically, I have two siptrunks named siptrunk1 and siptrunk2.
The call falls from siptrunk1 to the number 777100102, - this is a virtual number that is automatically forwarded to the number 102 after entering FusionPbx, this is the number registered on the physical phone on the desk, after 10 seconds of not answering the call is pushed to the next siptrunk “siptrunk2” on the queue 5023.
Below are the configuration slides:

1.



2.

And the configured bridge to siptrunk2:



Maybe you guys will know what the error lies in, or how it should look properly?
Thank you beautifully in advance.

 

[pbxgeek]

One issue I saw right away is that you have a ring group with extension 102 and a destination 102 within that ring group. If 102 is an extension in your system, then you have a duplicate. Ring groups and extensions can not have the same extension number. It causes conflicts. I had to program my own code so the system checks for duplicates before saving new items.

 

[Tobey]

In this case, how to properly do the forwarding from the virtual number 777100102 to the extension 102 and after not answering for 10 seconds throws the call to the corresponding queue in another siptrunk?
I would appreciate your answer.

 

[pbxgeek]

You should use ring groups, but ensure that you create unique extensions. For example, you could route extension 777100102 to ring group 500. In this case, ring group 500 would have extension 102 set as the destination, ringing for 10 seconds, and the timeout destination should be your trunk.

 

[Tobey]

Beautiful thanks for the answer, please tell me in this case, which configuration for this purpose should be appropriate (I hope that any will be). I did the same thing, but in two different ways. The screenshots below:

1.


2a).



In the second step should be transfer or bridge?

2b).

 

[pbxgeek]

Why are you modifying these settings manually? Use the GUI to create a number and route it to the ring group. It will build the dialplan for you automatically. Screenshot 2b looks good for the ring group setup.

 

[Tobey]

Tell me this kind of thing more please. Actually as in the previous example the call works correctly, that is, when 777100102 redirects to the ring group 7810, it rings 102 and then goes out to siptrunk.



What if I do as in the screenshot below? That is, instead of the extension number of the group name 7810 I will give the extension number 102 and then forward to siptrunk, will this also work?

 

[pbxgeek]

I've never tried this particular combination myself, but it's worth a try. As long as you can enter the trunk extension in the ring group, I don't see any issues. Let us know if it worked. It would be good for future readers.

 

[Tobey]

@pbxgeek thanks for your reply.
I'm still looking for an option to do my dialplan as before, call forwarding from 777100102 > record playback > 10 seconds ringing to 102 > (SIPtrunk) 5023 - without using ring groups to encapsulate this in one dialplan entry, the 10 seconds is triggered directly on the number in the call timeout option. I wanted to do it the following way, but it works forwarding only to 102, after which it disconnects the call. The question is whether you have the knowledge and willingness to point out to me what is wrong here.

 

[pbxgeek]

From what I can tell, your issue is the transfer directive. Once you use it, you leave this particular dialplan, and that's why your bridge doesn't execute.

Try switching it to the bridge statement. Your final XML file should look similar to this.

    <extension name="DID_Incoming_Call">
        <!-- Accept the call for a specific DID -->
        <condition field="destination_number" expression="^1234567890$">
            <!-- Answer the call -->
            <action application="answer"/>

            <!-- Stream a greeting file -->
            <action application="playback" data="/path/to/greeting.wav"/>

            <!-- Attempt to call extension 102 -->
            <action application="bridge" data="user/102@${domain_name}"/>

            <!-- If extension 102 does not answer, go to another bridge destination -->
            <action application="bridge" data="sofia/gateway/another_gateway/5023"/>
        </condition>
    </extension>