10-22-2013 02:17 AM - edited 03-16-2019 08:00 PM
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.
10-22-2013 05:26 AM
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
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10-22-2013 05:30 AM
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
10-22-2013 05:48 AM
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.
10-22-2013 05:51 AM
I try to ask you what is access code))) Just dialed numbers or you mean COR lists?
10-22-2013 05:53 AM
It can be a single number.
HTH
--
Jorge Armijo
Please remember to rate helpful responses and identify helpful or correct answers.
10-22-2013 05:55 AM
Could you provide example?
10-22-2013 06:48 AM
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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