VG224 strange issue

[Anil Sharma]

Hello,

I am facing some strange issue with Vg224, the device is added as SIP trunk on CUCM. All incoming and outgoing calls are working fine.

The issue is that when someone calls the analog phone, as soon the person answer the call he first hear some tone like some digits are dialed and then can hear the other party.

When normally analog phone is picked we get normal dial tone.

Any suggestions are much appreciated.

!

voice-port 2/17

echo-cancel coverage 32

cptone CN

timeouts interdigit 3

station-id number 12345

!

!

dial-peer voice 100 pots

description *** Analog User ***

translation-profile incoming CLASS-NATIONAL-Mobile

destination-pattern 12345

port 2/17

forward-digits all

!

!

Regards,

[MOHIT SINGH · Accepted solution]

Hi Anil,

You have configured "forward-digits all" under dial peer for FXS port. I suspect the tones you are hearing are DTMF tones of 12345 which is getting forwarded to analog phone because of forward digit all configuration. Kindly try removing "forward-digits all" from dial peer and then check.

Regards,

Mohit Singh

[Anil Sharma]

Thank you Mohit, You are right. It did fix the issue.

However I had to add no digit-strip cmd under pots dial-peer to make it work.

Thanks again.

[Cr2803]

hi, had same issue here, but didn't have the 'Forward digits-all' command.

I had put the 'no digit-strip' command instead.

dial-peer voice 85904415 pots
destination-pattern 85904415
no digit-strip
port 2/0

resulting in same issue explained by Anil.

So, my fix was to remove 'no digit-strip' by typing 'digit-strip'

and worked like a charm.

[...]

voice-port 2/0
cptone FR
bearer-cap Speech
station-id name 85904415
station-id number 85904415
caller-id enable

[...]

dial-peer voice 1 voip
description Outgoing 10 number digits
destination-pattern ..........
session protocol sipv2
session target sip-server
session transport udp
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 3 voip
description All incoming calls
session target sip-server
incoming called-number .
voice-class codec 1
no vad
!
dial-peer voice 85904415 pots
destination-pattern 85904415
port 2/0
!
!
sip-ua
credentials username 85904415 password 7 08052475C25332087547863667752 realm sip.phone.ue
registrar 2 dns:sip.phone.ue expires 360
sip-server dns:sip.phone.ue
host-registrar

Brgds