Redirect Calls

abukhazneh

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#1
Hello All,

I have a question about redirectin calls from one box to another , i have 2 boxes connected togother and i want the calls that come to box one go to extensions on box 2 i.e( i want to add DID from the ISDN line connected to box one and direct them to extensions on box 2 ) any ideas how to do that ,, pliz help me as soon as possible ,

thnx in advance,\
Best regards
 

abukhazneh

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#2
another thing i want to ask about , if i register aphone remotly how much it costs me from the bandwidth i have , will take alot or what ?????
 

jgutierrez

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#3
To redirect calls from one server into another one, you will need the following:

1. Create an sip/iax2/dahdi trunk to connect both servers
2. Create outbound routes to intercommunicate both servers
3. Create a misc destination (PBX->PBX Config->Misc Destinations) on the dial box, enter the extension on the other server
4. Go to the inbound routes, there you should get a "Misc Destination" field
 

dicko

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#4

Chilling_Silence

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#5
Haha, good to see the love being shared around ;)

For what its worth, if you ever forget the URL, just google "Asterisk bandwidth Calc" and its there :)
 

rafael

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#6
Nice,

I liked it:)
 

ericng

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#7
Hi jgutierrez,

I am having the same needs of redirect calls from one box to another, appreciate if you could explain how to setup using an example as I still cannot figure out how to proceed?

BTW, via the ways you have described, can the receiving party gets the caller id and dialed number ID?


Thanks


Eric
 

jgutierrez

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#8
Hello ericng,
I recommend you to take a look at "elastix without tears" to see how to configure a trunk between two servers, once you have that the other steps that I have included would be very easy to follow, else tell me which step isn't clear
 

ericng

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#9
Hi jgutierrez,

I did followed through the section in "elastix without tears" and managed to redirect calls from one server to another but the receiving always get the caller id. I wanted both the caller id and dialed number (I am using E1 DID lines) to appear on the receiving extension. I have been struggling for this for the past many weeks but no solution so far. Hope some one can shed some light

Eric
 

dicko

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#10
Perhaps if you constructed custom extensions that closely (or indeed , exactly) matched the inbound did's, and further matched these custom extensions one-to-one with the inbound routes you have on the E1 if necessary, and used the dial string to send on the DID (in what ever fashion you choose) to the trunk you have built between your two servers then the DID data would, in effect, be preserved between the boxes. (true it's ugly and you could write a custom context to do it much better using the ${OUTNUM} construct , but this could all be done from the FreePBX gui)

possible example:

inter-pbx trunk is called iax2iax

did = 1234 destination would be 1234

custom extension 1234 has a dial string of IAX2/iax2iax/1234

and at the far end 1234 would be de-constructed as an inbound route to suit

(there is some concept of DID "globbing" or more like RegEx parsing built in if your number plan suits that might save you some work (with ${OUTNUM}) )

I use IAX trunking as I find it to be clean and efficient, but feel free to use any trunking that works for you.
 

ericng

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#11
Hi Dicko,

Thank you for your hints. I am still new to asterisk PBX and therefore, appreciate if you could provide some sample dialplan that can pass the caller ID and dialed number via IAX trunk between the servers
 

dicko

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#12
Unfortunately that is not easy to do directly in Elastix/FreePBX, My method showed a way that "regenerates" a DID for the next leg in the "tandem" call you are trying to synthesise. Whilst it is possible to build a custom context to do exactly what you need it would require more understanding than you have right now, I encourage you to seek that understanding, but until you reach that point I would stick with GUI and try my method.

the Custom extension dial string "IAX2/iax2iax/1234" will just seize the iax2iax trunk and dial 1234 it is up to you what you use as the dial number and up to you what you do at the far end when it receives a call to DID 1234, If you care to use anything other than 1234, for example the incoming DID in it's entirety, please do so.

If you build a Custom extension for every DID you want to redirect (in it's entirety) , each of these DID's can be redirected directly to the trunk you wish to use without the use of an intermediary inbound route, the DID will be regenerated by the dialstring you program in that custom extension

DID 8144355467
Custom extension 8144355467 has a dialstring of IAX2/iax2iax/8144355467
dials 8144355467 on the interoffice trunk iax2iax

DID 8144355468
Custom extension 8144355468 has a dialstring of SIP/sip2sip/8144355467
dials 8144355467 on another interoffice trunk sip2sip (here you replace the original DID with one of your choosing)

DID 8144355469
has an inbound route set to IVR1 on the local box and goes there


DID 8144355470
Custom extension 8144355470 has a dialstring of ZAP/G0/0012125551212
dials 0012125551212 on your E1 trunk and gets you "information" in New York (substitute your cell phone number if that is more useful)

etc.

(In all cases the CID will be preserved and honored if the trunk supports it (maybe not the E1))

You could do a similar thing with a custom trunk but it would be a little harder to understand and has less granularity than this method.
 

ericng

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#13
Hi dicko,

Thanks for a very good explanation. Will attempt using the methods you have described.


Eric
 

mila

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#14
anyone has find configuration to get solution that question??
to direct call from one server elastix to another server elastix

...ehm i has problem for it (-_-)
 

mila

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#15
eric..i cannot to reply your messages on my email...that is error email address
 

ericng

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#16
Hi mila,

Appreciate if you could send to me to another email address specified in a email that I will be re-compose to you shortly.

Thanks


Eric
 

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