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New Member

Analog Lines & SIP Trunk

Hello all,

I am having 6 analog lines terminating on my CME 9.1 at branch.

I have SIP Trunk between the same branch CME & HQ having CUCM 4.1.

I have configured translation rule to divert call coming from Analog Line to an extension = 9033 (configured for IVR at CUCM in HQ)

so the call route is Call coming in on CME from Analog Line ------> on CME translation rule, & forwarding to extension 9033 through SIP Trunk

following is the configuration :

voice translation-rule 2

rule 1 /77335508/ /9033/

!

voice translation-profile 2

translate called 2

!

voice-port 0/2/1

translation-profile incoming 2

supervisory disconnect dualtone mid-call

supervisory dualtone-detect-params 1

timeouts call-disconnect 5

timeouts wait-release 5

connection plar opx 9033

caller-id enable

!

dial-peer voice 9033 voip

corlist outgoing call-national

destination-pattern 9033

session protocol sipv2

session target ipv4:192.168.0.10

voice-class codec 1 

voice-class h323 1

dtmf-relay rtp-nte

no vad

!

When I can call internally to extension 9033, it plays IVR prompt then it transfer to Agent (which is expected behaviour) but if i call from my mobile to 77335508 some times it hit 9033 & i hear the prompt then transfer to the agent but most of the time it forwards to some voice msging system & i dont know from where this prompt is playing.

I shutdown dial-peer voice 9033 + CUE dial-peer & tried call on 77335508 still the same prompt plays. If i disconnect the analog line from CME & connect through analog phone it does not forward to any prompt & the analog phone rings.

One this i just remembered that i updated my CME from 8.6 to 9.1 & it worked on 8.6 version couple of times.

Is there any compatibility or design issue with receiving call in analog lines & then forwarding to SIP Trunk, is this right design ?

Regards,

2 REPLIES
New Member

Analog Lines & SIP Trunk

can any body suggest how to trace the incoming & outgoing calls in analog connectivity ? or what are the debug commands ?

New Member

Analog Lines & SIP Trunk

Hello,

I just noticed that my fxo port does not off-hook if i drop the call from my mobile while hearing the IVR prompt (configured at HQ CUCM 4.1) & as soon as the port remains off-hook the telco provider plays msg prompt

Any suggestion as per updated config :

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 handle-replaces

fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none

sip

  bind control source-interface GigabitEthernet0/1.192

  bind media source-interface GigabitEthernet0/1.192

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g711alaw

codec preference 3 g729r8

codec preference 4 g729br8

!

voice class h323 1

  h225 timeout tcp establish 3

  h225 timeout setup 3

!

voice class dualtone-detect-params 1

freq-max-deviation 20

freq-max-delay 20

cadence-variation 20

!

voice class custom-cptone AR-custom

dualtone disconnect

  frequency 416 416

  cadence 400 320 240 520

voice-port 0/2/1

translation-profile incoming 2

supervisory disconnect dualtone mid-call

supervisory answer dualtone sensitivity high

supervisory custom-cptone AR-custom

supervisory dualtone-detect-params 1

no battery-reversal

no vad

no comfort-noise

timeouts call-disconnect 5

timeouts wait-release 5

connection plar opx 9033

description **** Voice Port for Etisal IVR ****

caller-id enable

!

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