04-29-2010 06:45 AM - edited 03-15-2019 10:32 PM
Hi All,
Calling my ext 200 internally after 15 seconds I go to voicemail, great no issues.
Calling in from my SIP trunk and my ext 200 rings, after 15 seconds the call will end and will not be put through to voicemail.
Relevant config below.
Any comments greatly appreciated.
Thanks,
Craig.
voice service voip
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 none
no fax-relay sg3-to-g3
sip
registrar server expires max 3600 min 3600
localhost dns:sip.xxxxxxx.co.uk
no update-callerid
!
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
dial-peer voice 2000 voip
description ** cue voicemail pilot number **
destination-pattern 400
b2bua
voice-class sip outbound-proxy ipv4:10.1.10.1
session protocol sipv2
session target ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 3000 voip
description Main-VOIP-123Telecom-In
translation-profile incoming voip_incoming
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number 44144650xxxx
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
no dial-peer outbound status-check pots
sip-ua
authentication username 44144650xxxx password 7 xxxxxxxxxxxxxx
no remote-party-id
retry invite 2
retry register 10
timers connect 100
registrar dns:sip.xxxxxxxxx.co.uk expires 3600
sip-server dns:sip.xxxxxxxxx.co.uk
connection-reuse
host-registrar
!
ephone-dn 1 dual-line
number 200 secondary 44144650xxxx no-reg primary
pickup-group 1
label Reception
name Reception
call-forward busy 400
call-forward noan 400 timeout 15
!
!
ephone 1
video
mac-address 0026.CBC0.2091
username "Reception" password 1234
type 7975
button 1:1 6m2 7m3 8m4
!
Solved! Go to Solution.
04-29-2010 07:47 AM
Looks like a codec problem. Your incoming dial peer is using g.729. Outbound to CUE is g.711. You will either need to change inbound to g.711 (both on the dial peer and with your carrier) or use a transcoder.
Hope this helps.
Brandon
04-29-2010 07:47 AM
Looks like a codec problem. Your incoming dial peer is using g.729. Outbound to CUE is g.711. You will either need to change inbound to g.711 (both on the dial peer and with your carrier) or use a transcoder.
Hope this helps.
Brandon
04-29-2010 08:20 AM
Spot on
all I had to do was:
dial-peer voice 3000 voip
codec g711ulaw
Cheers.
11-18-2011 05:32 AM
Hi Brandon,
I'm having the same problem where the incoming call from SIP TRUNK to Unity Express doesn't complety. I saw that the codec of Provider is G729 and the unity use the G711ulaw, but my doubt is how can I change the codec from Provider to Codec Unity Express G711ulaw ?
Could you hel me please ?
Regards,
Vinicius
11-18-2011 05:42 AM
Does your provider support g.711? If so, change the inbound dial peer to g.711. If not, you will need to invoke a transcoder to translate between g.729 and g.711.
Brandon
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