cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
865
Views
0
Helpful
5
Replies

Digit strip in 7821 SIP Phone

nmdc.kzstan
Level 1
Level 1

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.

5 Replies 5

Dragan Ilic
Level 4
Level 4

Can you post your dial-peer and voice translation config?

BR,

Dragan

HTH,
Dragan

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

 

 

 

nmdc.kzstan
Level 1
Level 1

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?

I have the same issue with 7821 and CUCM 9.1

nmdc.kzstan
Level 1
Level 1

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