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SIP trunk only for one fax

Hello.

I have the next scheme:

IP Phones --- CME ---- SIP Trunk ---- Service provider

All my phones use this SIP trunk. Now I need to add one analogue fax and new SIP trunk. Only this new fax must use this new SIP trunk.  How could I do it?

Please help.

7 Replies 7

Jorge Armijo
Level 4
Level 4

If the analog fax is connected to the same CME box, you can use a different access code to route the calls using one SIP Trunk or the other.

--
Jorge Armijo

Please remember to rate helpful responses and identify helpful or correct answers.

-- Jorge Armijo Please remember to rate helpful responses and identify helpful or correct answers.

But how can I do it? For now I only can add the second registrer. My config:

sip-ua

credentials number xxx_HoLdInG username xxx_HoLdInG password 7 xxx realm asterisk

authentication username xxx password 7 xxx

registrar 1 ipv4:xxx expires 3600

registrar 2 ipv4:xxx expires 3600 - my new SIP server

sip-server ipv4:xxx

My analog fax:

dial-peer voice 11 pots

destination-pattern 2101

authentication username xxx password xxx - my credentials for registrar 2

port 0/0/1

no sip-register

So?

All you have to do is a dial-peer with a different access code, pointing to the second SIP Trunk, that code will be dialied from the FAX only, then you route the call through the SIP Trunk after removing the code with a translation.

--
Jorge Armijo

Please remember to rate helpful responses and identify helpful or correct answers.

-- Jorge Armijo Please remember to rate helpful responses and identify helpful or correct answers.

I try to ask you what is access code))) Just dialed numbers or you mean COR lists?

It can be a single number.

HTH

--
Jorge Armijo

Please remember to rate helpful responses and identify helpful or correct answers.

-- Jorge Armijo Please remember to rate helpful responses and identify helpful or correct answers.

Could you provide example?

This is what Jorge is saying...configure access code to use with fax calls and configure xlation rule to strip the access code before presenting to telco

1.  voice translation-rule 1

rule 1 /^8/ //

voice translation-profile fax

translate called 1

2. Configure a different dial-peer with an access code...

eg dial-peer voice 1000 voip

description fax dial-peer

destination-pattern 8T

translation-profile outgoing fax

codec g711u

session target ipv4:x.x.x.x

dtmf-relay rtp-nte

session protocol sipv2

no vad

3. Ask your users to dial 8 before they dial the fax numbers.. (your access code can be anything..its up to you)

The other solution is similar this one but without using an access code for dialling..

configure two xlation rules. The first to prefix any digit to the number dialled by your fax machine...lets use 8 again and the sencond to strip the 8 before sending to your ITSP

voice translation-rule 1

rule 1 /\(.+\)/ /8\1/

translation-profile fax1

translate called 1

voice translation-rule 2

rule 1 /^8/ //

voice translation-profile fax2

translate called 2

Then apply it the first rule  to the fax pots dial-peer

dial-peer voice 11 pots

destination-pattern 2101

translation-profile outgoing fax1

and apply the second rule to the dial-peer going to your ITSP

dial-peer voice xx voip

session portocol sipv2

translation-profile outgoing fax2

Explanation..

The first example your users will dial an access code and that access code will match the dial-peer going to your ITSP sip trunk on your gateway, then your xlation rule will strip the access code and send the call out

The second example, your users will dial normally, but your first xlation rule wil prefix a digit so that you can match the dial-peer going to your ITSP sip trunk and the second xaltion rule will strip the prefixed digit

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