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Need to allow original caller id to follow forwarded Call

Posted on 2015-02-13
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Last Modified: 2015-08-13
I have a Cisco CME implememation. I have a SIP trunk and 2 Pots lines. I have my main number still on the POTS line. When a call comes in and I do not answer it, I have it set to forward to my cell phone. Unfortunately, when it forwards, the number that comes up for Caller ID is my number (The SIP Trunk Number), not the incoming Caller's. I need to find out how I can set that. I am using SIP.US as my Sip Trunk provider and a local ILEC for the POTS lines.

The relevant config is:

voice service voip
 ip address trusted list
  ipv4 xxx.xxx.xxx.xxx
  ipv4 yyy.yyy.yyy.yyy
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 sip
  bind control source-interface GigabitEthernet0/1.12
  bind media source-interface GigabitEthernet0/1.12
  e911
  header-passing
  transport switch udp tcp
  asserted-id ppi
  localhost dns:rtr.mydomain.com
  midcall-signaling passthru
  no call service stop

voice class codec 1
 codec preference 1 g711ulaw
 codec preference 2 g729r8
!
!
voice class sip-profiles 1
 request INVITE sip-header From modify "<sip:username@zzz.zzz.zzz.zzz>" "<sip:<username>@gw1.sip.us>"
 request REINVITE sip-header From modify "<sip:<username>@1zzz.zzz.zzz.zzz0>" "<sip:<username>@gw1.sip.us>"
 
dial-peer voice 2 voip
 description **Outgoing Calls to SIP.US SIP Trunk**
 translation-profile outgoing SIP.US-Outgoing
 destination-pattern 1[2-9].........
 session protocol sipv2
 session target dns:gw1.sip.us
 voice-class sip dtmf-relay force rtp-nte
 no voice-class sip early-offer forced
 voice-class sip profiles 1
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad
 authentication username <SIPUSERNAME> password 7 <SIPPASSWORD>
 no supplementary-service h450.2
 no supplementary-service h450.3
 supplementary-service h450.12

sip-ua
 credentials username <SIPUSERNAME> password 7 <SIPPASSWORD> realm gw1.sip.us
 authentication username <SIPUSERNAME> password 7  <SIPPASSWORD>
 retry invite 2
 retry register 10
 timers connect 100
 registrar 1 dns:gw1.sip.us expires 360 refresh-ratio 20 auth-realm gw1.sip.us
 connection-reuse

telephony-service
 call-forward pattern .T
 call-forward system redirecting-expanded

ephone-dn  1
 number 8685 no-reg both
 label OFFICE
 description Daddy
 name Daddy
 loopback-dn 100
 caller-id passthrough
 call-forward busy 2000
 call-forward noan 15551234567 timeout 15
 hold-alert 30 originator

ephone-dn  100
 number 18153578685
 loopback-dn 1
 caller-id passthrough

ephone  1
 description description Daddy
 mac-address MAC1.MAC1.MAC1
 ephone-template 2
 username "Office"
 type 7965
 mwi-line 1
 button  1o1,198 2:13 3:98
!

Any thoughts? I would really like to get this working.

Thanks
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Comment
Question by:Wyant Niswonger
  • 3
5 Comments
 
LVL 47

Expert Comment

by:David
ID: 40610155
Well, can't help you with the cisco but are you aware that the caller id forwarding information is set up by the phone company?   Did you check with them that they forward the ID on call forwarding to begin with?    Certainly that is a deal-killer if they do not.

So step back and first make sure that the data is there in the first place.  You could very well discover it is not turned on, and that once the phone company turns that feature on for you, then it just automatically shows up.
0
 
LVL 44

Expert Comment

by:Davis McCarn
ID: 40610158
The only way it will work is to setup the forwarding directly on the POTS provider.  With Windstream, here for example, I use *72 to forward the landline to my cell and the caller id works fine.
0
 

Author Comment

by:Wyant Niswonger
ID: 40624205
I am using Momentum and they do forward the caller ID information. I see it on the display when a call comes in. What I am trying to have happen is for that caller ID information to be retained when CME forwards the call out my SIP trunk. I had it setup with Momentum before I started attempting to use CME and it worked. I am just attempting to see if there is any way possible to do it.
0
 

Accepted Solution

by:
Wyant Niswonger earned 0 total points
ID: 40920830
I was able to work this out on my own. I did not need to engage my provider. It was in the configuration.
0
 

Author Closing Comment

by:Wyant Niswonger
ID: 40927713
I was able to solve this on my own. The expert who answered was not correct.
0

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