cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
267
Views
0
Helpful
1
Replies

SIP Trunk Problem 2

yun zhang
Level 1
Level 1

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 ?

1 Reply 1

Hi.

Can you please post your config?

Thanks

Carlo

Please rate all helpful posts

"The more you help the more you learn"

Please rate all helpful posts "The more you help the more you learn"
Getting Started

Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community: