Forwarding a call and call display

cmiller82

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#1
I have a local extension, and a external extension that calls my cell phone. I have "follow-me" setup so it rings my cell if I don't pickup the local extension. How do I forward on the call display information so that I know who is really calling instead of seeing me home number?

Thanks,
Chris
 

Bob

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#2
Basically this is a restriction of your Voice provider, whether it be VoIP or Carrier lines, not the Elastix system.

As your Elastix system is the one making the call to your Mobile, it is using a VoIP or Carrier line to make the call to your mobile. Almost all carriers will not let you present a number is it coming from, unless it is one of the numbers associated with VoIP or Carrier line.

So in summary,

If you are using PSTN, it does not support it anyhow
, it will always report the line that is being used to call your Mobile.

If you are using E1/T1 or VoIP provider, it will only allow you to present a number (CID) that is part of the range of DID's that you have assigned to that service.

The other thing you must remember is that Elastix/Asterisk actually answers the call, then based on your followme, calls your mobile like a separate call from the Asterisk system. Once you answer the call, it then bridges the call. I have not checked if the code allows the preservation of the CID information and passes it to the call going out to your mobile, but even if it did, you would find that 99% of Telco's would block or ignore this information (the basis for call spoofing).

This is different to a Telco diversion service which is all handled at their end and is part of their routing where it preserves the CID information.

For this purpose, what would be ideal is a modification of the Call Confirm ability when ringing external mobile numbers. Normally you just select 1 to accept the call and lets Asterisk know to bridge the call (mainly implemented so that you can make Mobiles part of a Ring Group or similar). But instead of just that message, that it says (by voice), the original phone number as well. So while you make a decision on whether to take the call, the caller continues to hear ring or musiconhold.

So basically you would receive a call on your mobile and the voice would be when you take the phone call:

"Incoming Call from 041555555. Press 1 to confirm call acceptance"

This, I believe is feasible, and not overly complicated. Might look into it this weekend...

Regards

Bob
 

cmiller82

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#3
Thank you very much. The first two sentences would have been enough, but I really appreciate the detailed explanation (and possible solution).
 

Bob

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#4
No Problems,

Sorry for the detail, I was not sure what level you needed, so it is always good to add little.

Secondly, others may be looking for answers, and come across your question, so it may assist others as well, and they may be looking for more detail why it does not work.

But will look at the Confirm with Phone number feature, as it would be quite useful...

Regards

Bob
 

itjumper

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#5
I also have a question. I have set up followme to call my cell. I can see the callerid (name & phone #) of the original caller. Now I would like to know this is actually a call forwarded from my pbx system. Any idea how to add a prefix (the number that the caller called) to the original caller name ?

e.g.

caller Joe Doe, phone number 1111111111 calls 2222222222 (pbx)
asterisk then calls my cell 3333333333

right now my cell will show

Joe Doe
1111111111

I would like to see

22222222222-Joe Doe
1111111111
 

newnet82

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#6
Hi,

I have the same need.
Have you found a solution ?

Bruno
 

Xexexe

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#7
Hi,
At a client site they have a SIP trunk and with follow-me the
callerID of the original caller is shown on the mobile that the
call is forwarded to!
I have done nothing to achieve that so it seems to work out
of the box (1.6-12)
The SIP provider has this feature open though (for the moment)
and i expect this to stop at some point as its kind
of "illegal" if you don't own the number.

Perhaps you could send out one of your unused DIDs to indicate that
the call is one that is being forwarded ?
 

Phaseholder

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#8
Bob said:
Basically this is a restriction of your Voice provider, whether it be VoIP or Carrier lines, not the Elastix system.
Hi,
I have same problem as cmiller82 with my SIP trunk, my VSP swears that there were no limits, but they have a requirement - Elastix should send back Invite message with Diversion and PAI headers with information about call forwarding, they gave me an examples of incoming & redirecting Invite messages.

Invite from VSP:

INVITE sip:+77275556677-e433aag41ivj2@10.14.0.37:5060;ep=10.20.105.26:5062;fw=10.20.105.26:5062;transport=udp SIP/2.0
Via:SIP/2.0/UDP 10.14.0.2;branch=z9hG4bK-BroadWorks.as1.bwks.snom.kz-10.14.0.37V5060-0-156472942-320893338-1273770749146-
From:<sip:+77274445566@10.14.0.2;user=phone>;tag=320893338-1273770749146-
To:"Phaseholder"<sip:+77275556677@sip.snom.kz;ep=10.20.105.26:5062;fw=10.20.105.26:5062>
Call-ID:BW231229146130510-1792672556@10.14.0.2
CSeq:156472942 INVITE
Contact:<sip:10.14.0.2:5060>
Supported:100rel
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY
Call-Info:<sip:10.14.0.2>;appearance-index=1
Accept:multipart/mixed,application/media_control+xml,application/sdp
Max-Forwards:70
Content-Type:application/sdp
Content-Length:197

Invite from Elastix should be like this one:

INVITE sip:+77007778899@10.14.0.39:5060;user=phone;transport=udp SIP/2.0
Via:SIP/2.0/UDP 10.14.0.2;branch=z9hG4bK-BroadWorks.as1.bwks.snom.kz-10.14.0.39V5060-0-156398081-1303411593-1273770599424-
From:<sip:+77274445566@10.14.0.2;user=phone>;tag=1303411593-1273770599424-
To:<sip:+77007778899@10.14.0.39:5060;user=phone>
Call-ID:BW230959424130510643068587@10.14.0.2
CSeq:156398081 INVITE
Contact:<sip:10.14.0.2:5060>
P-Asserted-Identity:<sip:+77274445566@10.14.0.2;user=phone>
Privacy:none
Diversion:"Phaseholder"<sip:+77275556677@10.14.0.2;user=phone>;reason=unconditional;privacy=off;counter=1
Supported:100rel
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY
Accept:multipart/mixed,application/media_control+xml,application/sdp
Max-Forwards:70
Content-Type:application/sdp
Content-Length:198

How can i set up my Elastix to Re-send invite with Diversion?
 

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