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SIP Trunk Problem 2

Hi Guys

I have a question for ISDN imcoming call then transfer to another voice gateway

our customer has two voice gateway as srst gateway and with three isdn bind lines , below are imcoming call flow

isdn-----vg2(callcenter phone srstGW)-----cm

isdn-----vg1(office user phone srstGW)-----cm

vg1------------sipv2------------vg2

To callcenter incoming call flow

outside user -----vg2-----cm----callcenter agent

outside user -----vg1-----vg2---cm----callcenter agent

To office incoming call flow

outside user------vg1---cm---office phone user

outside user------vg2----vg1----cm------office phone user

I have below testing

when an outside user dial toll-free number from vg1  to callcenter  then the call need to forward to vg2 firstly then transfer to cm directly .

I have enabled the debug isdn q931 on vg1 and output the below inforamtion . after 10 seconds the outside user heard du du du sound and disconnect the call.

Cause i = 0x80B9 - Bearer capability not presently authorized

when i have testing to toll-free number again from vg1 to callcenter after 10 seconds i can heard the welcome voice and can be transfer to phone.

but have a strange problem , the pstn call connected by vg2 but the incoming call from vg1  so anyone can explain this appearance ?

but delete the sip trunk and set a dial-peer to cm directly in vg1 and tested is fine and so fast. what's the cause for this issue ?

  • IP Telephony
1 REPLY

SIP Trunk Problem 2

Hi.

Can you please post your config?

Thanks

Carlo

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