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FAX connected to an FXS port. No incoming call!

s.souroutis
Level 1
Level 1

I have a 1760 with 2 FXS ports.

I connect FAXs on both FXS ports and I try to make a call but it's not succesful.

I can call a remote FAX (VOIP call) with no problem, but when there is an incoming call, the FAX answers the call but cannot handshake and finaly the connection terminates.

When using the VOIP call, I use the following configuration:

dial-peer voice 10 voip

desctination-pattern 123

fax rate disable

fax protocol pass-through g711alaw

but int the pots there are no commands

What do I do?

2 Replies 2

thisisshanky
Level 11
Level 11

What does the POTS dial-peer command look like ?

dial-peer voice 11 pots

destination-pattern xxxxx (the fax number)

port 1/0 (port number)

Sankar Nair
UC Solutions Architect
Pacific Northwest | CDW
CCIE Collaboration #17135 Emeritus

Hi,

Sorry but I didn't tell you the whole story.

The call is TDM - VoIP (SIP) - TDM

Forget the previus question.

It was just a test call to solve my following problem.

This is the router's copnfiguration:

dial-peer voice 7 pots

description POTS for FXS port

application session

destination-pattern 30899

fax rate disable

port 0/1

forward-digits all

!

dial-peer voice 20 voip

description VOIP for FAX call

application session

destination-pattern 0721680000

voice-class codec 1

session protocol sipv2

session target sip-server

fax rate disable

fax protocol pass-through g711alaw

no vad

!

Scenario 1

When cisco fax calls the other PSTN fax, the call starts with codec G729br8. When fax tone is detected, there is a re-invite from B-side and then a codec negotiation follows to g711alaw and the fax is working.

This call is succesful.

Scenario 2

When the other PSTN fax calls the cisco fax, the call starts with g729br8/g729r8 but there is no re-invite from cisco (B-side now). The codec never changes to g711alaw.

The call is not successful.

What I am looking for is the command

"fax protocol pass-through g711alaw"

in the POTS configuration to enable SIP re-invite for codec re-negotiation.

Any help here?

Thank you

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