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prashanthpatha
Beginner

call forward to PSTN on cme

Hi,

unable to set up call forward to PSTN. I have tried activating the Call forward via the phone or manually via the config, but when I attempt a call to IP Communicator from PSTN or via extn I am not seeing re-INVITE which should be generated for the forwarded call. Am i missing something?

    PSTN / IP phone ------> Calling extn on CME (which is call forwarded to another PSTN number)

config below:

!
!
voice service voip
ip address trusted list
  ipv4 0.0.0.0 0.0.0.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
  registrar server expires max 250 min 200
  asserted-id pai
  localhost dns:XXXXX
  outbound-proxy dns:XXXXX
!

!
dial-peer voice 100 voip
description ** Incoming call from SIP trunk **
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 1 
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 101 voip
description ** Outgoinging call to SIP trunk **
translation-profile outgoing SIPOUT
destination-pattern 1T
session protocol sipv2
session target sip-server
voice-class codec 1 
voice-class sip dtmf-relay force rtp-nte
voice-class sip profiles 101
dtmf-relay rtp-nte
no vad
!
dial-peer voice 102 voip
description ** Outgoinging call to SIP trunk **
destination-pattern 0[2-9].T
session protocol sipv2
session target sip-server
voice-class codec 1 
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
!

!
!
telephony-service
max-ephones 4
max-dn 12
ip source-address 192.168.100.2 port 2000
calling-number initiator
timeouts interdigit 5
load 7960-7940 P00308010200
date-format dd-mm-yy
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
transfer-system full-consult dss
transfer-pattern .T
transfer-pattern 0.T
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-dn  1  dual-line
number 4961 secondary 99474961 no-reg both
label 4961
name 4961
call-forward all 021605547
!
!

10 REPLIES 10
Daniele Giordano
Rising star

Can you add the output of "debug ccsip message" and "debug voice ccapi inout" during a failed call?

Regards.

Please find debugs attached.

Calling number is "021605547" ---> called number "9474961" and call forwarded to "02102052745"

Can you post the translation-profile outgoing SIPOUT?

Thanks.

Hi

here it is

!
!
voice translation-rule 1
rule 1 /^10/ /0/
!
voice translation-rule 2
rule 1 /4961/ /99474961/
rule 2 /4962/ /99474962/
rule 3 /4963/ /99474963/
rule 4 /4964/ /99474964/
!
!
voice translation-profile SIPOUT
translate calling 2
translate called 1
!

Even if your CME uses SIP, try to add these commands under "voice service voip":

no supplementary-service h450.2
no supplementary-service h450.3

And try to change the calling number under "telephony-service":

calling-number local

Let me know the result.
Regards.

Hi,

Thank you for your help... given that a shot but it still fails. logs attached. I still don't see a re-invite from CME to call-forwarded number.

In my opinion the call forward process starts but it doesn't proceed.
These are informations from your logs:
*May  4 22:57:23.143: //59/B502B9E88077/CCAPI/ccCallSetupRequest:
   Calling Number=99474961(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
   Called Number=096232641(TON=Unknown, NPI=Unknown),
   Redirect Number=99474961, Display Info=4961
   Account Number=21605547, Final Destination Flag=TRUE,
   Guid=B503F238-75D8-11E0-807C-E605FE64EA64, Outgoing Dial-peer=101

All subjects of the call are defined correctly.

Next  logs:
*May  4 22:57:23.143: //60/B502B9E88077/CCAPI/ccIFCallSetupRequestPrivate:
   SPI Call Setup Request Is Success; Interface Type=3, FlowMode=1

And so the CME responds with a SIP TRYING but nothing else.

Now we know that the forward call uses the Dial-Peer 101.

Does a direct call (without forwarding) work through this dial-peer?

The session target of dial-peer 101 is the "sip-server". In wich way is configured? Is it an IP address or a name?

Can you ping it from the CME? The CME can resolve the name via DNS?
Can you post the sip-ua config?

Regards.

giordano_p wrote:

Even if your CME uses SIP, try to add these commands under "voice service voip":

no supplementary-service h450.2
no supplementary-service h450.3


Those have no bearing on SIP, BTW.  They are specifically for H.323, and used when H.323 to CUCM since it doesn't support H.450 and IOS does.

They do help in H.323 forwarding scenarios where CUCM is involved, though.

Does a direct call (without forwarding) work through this dial-peer? YES

The session target of dial-peer 101 is the "sip-server". In wich way is configured? Is it an IP address or a name? FQDN

Can you ping it from the CME? YES

The CME can resolve the name via DNS? Resolved on the CME Can you post the sip-ua config?

sip-ua

credentials number 99474960 username 99474960 password 7 XXXXXXXXX realm as-test.xys.net 

authentication username 99474960 password 7 XXXXXXX 

calling-info pstn-to-sip asserted-id number set 99474960 

no remote-party-id 

disable-early-media 180 

retry invite 2

retry register 3

timers connect 100 

registrar dns:as-test.xys.net expires 60  sip-server dns:as-test.xys.net 

host-registrar

I've attached my working configuration.

Under ephone-dn 14 is configure the "call-forward all" option.

Regards.