Calls to Unity Express fail using SIP trunk

Answered Question
Apr 29th, 2010
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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

!

Correct Answer by Brandon Buffin about 6 years 12 months ago

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

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Correct Answer
Brandon Buffin Thu, 04/29/2010 - 07:47
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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

craig.corbett Thu, 04/29/2010 - 08:20
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Spot on


all I had to do was:


dial-peer voice 3000 voip
codec g711ulaw


Cheers.

spazziani Fri, 11/18/2011 - 05:32
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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

Brandon Buffin Fri, 11/18/2011 - 05:42
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  • Purple, 4500 points or more

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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