04-15-2015 01:49 AM - edited 03-17-2019 02:39 AM
Hi,
When calling from 7821 SIP phone PSTN number (9[23]XXXXXX) turns into 9 in phone display.
When calling from 6961 SCCP phone registered to the same CME 9[23]XXXXXX turns into [23]XXXXXX, i.e. 9 is stripped after translation.
How do I prevent PSTN numbers from being stripped when calling from 7821?
show dial-peer voice
VoiceEncapPeer20006
peer type = voice, system default peer = FALSE, information type = voice,
description = `',
tag = 20006, destination-pattern = `140$',
forward-digits 0
session-target = `', voice-port = `50/0/40',
direct-inward-dial = disabled,
digit_strip = enabled,
register E.164 number with H323 GK and/or SIP Registrar = FALSE
fax rate = system, payload size = 20 bytes
supported-language = ''
CME version 10.5, PSTN connection is SIP.
04-15-2015 03:08 AM
Can you post your dial-peer and voice translation config?
BR,
Dragan
04-15-2015 08:33 PM
voice translation-rule 100
rule 10 /1../ /XXXXXXXXX/
!
voice translation-rule 102
rule 1 /^91/ /1/
rule 2 /^92/ /2/
rule 3 /^93/ /3/
rule 10 /^98/ /8/
rule 20 /^907/ /8/
rule 40 /^90/ /810/
voice translation-profile SIP_OUT
translate calling 100
translate called 102
dial-peer voice 110 voip
translation-profile outgoing SIP_OUT
destination-pattern 9[23]......
session protocol sipv2
session target ipv4:10.0.0.12
session transport udp
voice-class sip localhost dns:sip.XXXX.XX preferred
voice-class sip dtmf-relay force rtp-nte
voice-class sip early-offer forced
dtmf-relay rtp-nte sip-notify
codec g711alaw
fax protocol pass-through g711alaw
no vad
04-20-2015 02:54 AM
It turns out KPML is stripping digits. With KPML enabled even extension numbers are truncated,
124 truncated to 1:
Received:
INVITE sip:1@10.12.2.202;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.121.173:5060;branch=z9hG4bK376308e2
From: "A" <sip:140@10.12.2.202>;tag=00e16dbb7514000973d075bd-6811206e
To: <sip:1@10.12.2.202>
Call-ID: 00e16dbb-75140005-52232cfb-560f35c9@192.168.121.173
Max-Forwards: 70
Date: Mon, 20 Apr 2015 09:44:29 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7821/10.2.1
Contact: <sip:85E8-1A01@192.168.121.173:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow:
With KPML disabled:
Received:
INVITE sip:124@10.12.2.202;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.121.164:5060;branch=z9hG4bK51679917
From: "B" <sip:180@10.12.2.202>;tag=00e16dbb749c00075eb3513c-706a7410
To: <sip:124@10.12.2.202>
Call-ID: 00e16dbb-749c0004-5b8f3798-25c69f9c@192.168.121.164
Max-Forwards: 70
Date: Mon, 20 Apr 2015 09:47:19 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7821/10.2.1
Contact: <sip:402DAC-1256@192.168.121.164:5060;transport=udp>
Expires: 180
Accept: application/sdp
Is there any workaround?
04-20-2015 03:32 AM
I have the same issue with 7821 and CUCM 9.1
05-13-2015 03:43 AM
Dialplans are one way of solving this
voice register template 10
dialplan 10
voice register dialplan 10
type 7940-7960-others
pattern 1 91..
pattern 2 9.......
voice register pool 14
type 7821
number 1 dn 140
template 10
incoming called-number
no digit collect kpml
dtmf-relay rtp-nte sip-kpml sip-notify
username cisco password cisco
codec g711alaw
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