04-20-2009 09:25 PM - edited 03-15-2019 05:33 PM
I have a small router with Call manager Express on it. I have the system recieving and sending calls from a SIP provider. I also have calls coming in over regular internet to my IP address. Some calls are rejected though. If there is a leading + infront, the Call manager sends a 404 response. If the call comes in without the leading + it works no problem.
I guess my question is how do you get all inbound calls to remove a + if a call comes in with a plus but leave all other calls alone if there is no leading +
Solved! Go to Solution.
04-21-2009 10:53 PM
can we do:
debug voice ccapin ino
and debug the voice translation rules as well.
04-21-2009 02:24 AM
You can configure a Voice translation rule to remove the (+) plus, before passing it to cucm
in CUCM \+ represents \+
CSCsj17772 is the DDTS which strips the leading '+' in calling/called numbers.
This change was re-introduced so as to be in compliance with the Q.931 standard and also this was the old behavior of
IOS.
According to Q.931, there are two ways to represent an international
number in Q.931 calling party number field (Table 4-11/Q.931, TON):
1. Set TON (Type Of Number) as 'unknown', and include the full number
including the international prefix (+) in Octet 4.
2. Set TON as 'international', and include the full number without
international prefix in Octet 4.
The latter method is the typical one.
To overcome the below issue and if CSCsj17772 fix should not come into
picture, then configure a translation-profile at the inbound dial-peer
such that + will not be stripped.
At global level, configure the following
voice translation-rule 1
rule 1 /^\+/ //
In this case configure a dummy translation rule such that whatever is received will be retained as it is */
!
!
voice translation-profile voip-generic
translate called 1
!
At inbound dial-peer level, configure the following:
Since the matching inbound dial-peer in the customer scenario is "2",
please add the following to this dial-peer.
"translation-profile incoming voip-generic"
dial-peer voice 2 voip
translation-profile incoming voip-generic
service session
session protocol sipv2
session transport tcp
incoming called-number .
codec g711ulaw
no vad
By configuring as shown above, the fix of CSCsj17772 will not be
applied. So now the '+' will not be stripped and TON not set to
"International". By this you don't send the + to cucm
SiteA#test voice translation-rule 1 +14089022653
Matched with rule 1
Original number: +14089022653 Translated number: 14089022653
Original number type: none Translated number type: none
Original number plan: none Translated number plan: none
04-21-2009 02:33 PM
Thanks for the tip. This actually works to remove the + when I do the test command you show above. The one thing that is still not happening is that the DID is not forwarding to the 201 extension but still gives a 404 response to the caller.
So in your example, if a person calls 14089022653 from outside the system it is translated to 201 it rings my IP communicator. When sending the call with the plus, it gave 404. with the rule I put in we are still expeiriencing a 404 on the inbound call.
What would I be missing in this configuration.
This is what is in the Call Manager Express router.
voice translation-rule 1
rule 1 /^\+/ //
!
voice translation-rule 4
rule 1 /14165551212/ /201/
!
!
voice translation-profile FDID_Called_4
translate called 4
voice translation-profile voip-generic
translate called 1
!
dial-peer voice 3000 voip
description FDID
translation-profile incoming FDID_Called_4
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number 14165551212
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 3006 voip
description strip the plus
translation-profile incoming voip-generic
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number .
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
ephone-dn 11 dual-line
number 201 secondary 14165551212 no-reg primary
label 201
description Phone User
name Phone User
call-forward busy 401
call-forward noan 401 timeout 15
ephone 6
device-security-mode none
video
mac-address 001F.6C80.9033
username "phoneuser" password 1234
type 7971
button 1:11
04-21-2009 10:53 PM
can we do:
debug voice ccapin ino
and debug the voice translation rules as well.
04-22-2009 02:31 PM
I turned on the debug, and saw it was trying dial-peer 1000.
I looked for that and it was not with the rest of the voip dial-peers. It was matching on .%
I added the translation rule to that dial-peer and it is striping the +.
Thanks
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