08-09-2014 10:46 PM - edited 03-16-2019 11:41 PM
I have CME. connected to pstn by FXO ports and SIP trunk. so I have incoming numbers in fxo ports and incoming numbers in sip trunk too.
every inconming called number has corresponding local number.
how to make outgoing call so that everybody will call with its own corresponding incoming number?
as documented CME first uses for outgoing call lowest preference POTS dial-peer. and only when all pots bisy it uses lowest preference voip dial-peer
08-10-2014 02:02 AM
Can you please give some more information, like what number is coming in.
what number user wnats to dial, and for what interface it should go out.
and example will be much helpful.
regards
Chintan
08-10-2014 08:39 PM
local numbers 4901 to 4939
voice-port 0/0/0
connection plar 4937
description 393604 --- incoming called number from pstn 393604 transfered to local 4937
voice-port 0/0/0
connection plar 4937
description 393604
voice-port 0/0/1
connection plar 4932
description 393603
voice-port 0/0/2
connection plar 4935
description 393605
!
voice-port 0/0/3
connection plar 4901
description 393601
!
outgoing dial-peers for call to pstn 9......$
dial-peer voice 60 pots
translation-profile outgoing cut-9
preference 5
destination-pattern 9......$
no digit-strip
port 0/0/3
forward-digits all
!
dial-peer voice 61 pots
translation-profile outgoing cut-9
preference 4
destination-pattern 9......$
no digit-strip
port 0/0/2
forward-digits all
!
dial-peer voice 7182 voip
description SIP to kt pavl
translation-profile outgoing cut-9p
preference 1
destination-pattern 9......
session protocol sipv2
session target ipv4:10.235.226.241
voice-class codec 2
voice-class sip bind control source-interface Loopback2
voice-class sip bind media source-interface Loopback2
dtmf-relay rtp-nte
!
for outgoing call CME first uses dial-peer voice 61 pots. when it is bisy CME uses dial-peer voice 60 pots.
when it is bisy too CME use dial-peer voice 7182 voip
08-10-2014 08:48 PM
Hi,
I still dont understand what exactly your question is?
my assumption:
you have fxo ports and you have number on which PSTN will ring FXO port.
now every fxo port maps one to one with some internal directory number.
so when you dial out, instead of using internal directory number you want use the external fxo numbers?
regards
Chintan
08-10-2014 09:33 PM
yes.
I want those connected to FXO to dial out throu FXO
those connected to inbound SIP nimbers to dial out throu SIP
is it possoble?
and I want to be able to switch preference between FXO and SIP outgoing calls
08-10-2014 09:41 PM
yes, looks like it is possible.
now question 1.
who are connected to FXO port, who are those plar numbers? (CME ephones?)
and who are connected to inbound sip numbers? any exapmle?
regards
Chintan
08-10-2014 10:12 PM
yes. CME ephones
this example FXO incoming
ephone-dn 32 dual-line
number 4932
ephone 5
button 1:5 2:32
this example SIP incoming 7182704056 translated to 704056
voice translation-rule 92
rule 1 /^7182704/ /704/
ephone-dn 56 dual-line
number 704056
!
ephone 13
button 1:13 2:56
08-10-2014 11:08 PM
Man!! now it looks crystal clear.
so lets start by fxo attched ephone. i would just provide you the logic on how you can achieve this.
lets say than you have 4 ephone, and 4 ephone-dn
ephone-dn 1 is mapped to fxo port 1, ephone-dn to is mapped to fxo2, ephone-dn 3 mapped to fxo3 and ephone-dn 4 to sip trunk.
any outgoig calls from ephone dn should you the mapped fxo port or sip trunk, that is your requirement.
i would use prefix and traslation pattern to achive this.
prefix 1 is for fxo1
prefix 2 is for fxo2
prefix 3 is for fxo3
prefix 4 is for siptrunk
so create a trasation rule to prefix number ephones are calling with prefix
voice translation-rule 1
rule 1 /\(.*\)/ /1\1/
voice translation-rule 2
rule 1 /\(.*\)/ /2\1/
voice translation-rule 3
rule 1 /\(.*\)/ /3\1/
voice translation-rule 4
rule 1 /\(.*\)/ /4\1/
create profile to appply this rule on called number.
voice translation-profile fxo1
translate called 1
voice translation-profile fxo2
translate called 2
voice translation-profile fxo3
translate called 3
voice translation-profile sip4
translate called 4
now on appropriate ephone dn apply thi profile on inbound leg.
ephone-dn 1 (maps to fxo1)
translation-profile incoming fxo1
ephone-dn 2 (maps to fxo2)
translation-profile incoming fxo2
ephone-dn 3 (maps to fxo3)
translation-profile incoming fxo3
ephone-dn 4 (maps to sip trunk)
translation-profile incoming sip4
so after all this,
1. if ephone-dn 1 dials number XXX it will be traslated to 1XXX
2. if ephone-dn 2 dials number XXX it will be traslated to 2XXX
3. if ephone-dn 3 dials number XXX it will be traslated to 3XXX
4. if ephone-dn 4 dials number XXX it will be traslated to 4XXX
now some thing to play with dial-peers,
create dial-peer which routes anything that have dialed 1XXX goes throgh fxo1 (you can see maping on ephone 1 to fxo1)
dial-peer voice 1 pots
destination-pattern 1T
port fxo1
dial-peer voice 2 pots
destination-pattern 2T
port fxo2
dial-peer voice 3 pots
destination-pattern 3T
port fxo3
default digit strip will strip prefix 1,2,3 by default and you get your original diled number is being sent to pstn.
for voip this will not work, so you have to create 1 more trastion profile remove prefix before sending the number out
lets first create dial-peer for pattern 4xxx to be routed through sip
dial-peer voice 4 voip
destination-pattern 4xxx
session protocal sipv2
session targer <sip server>
but the prefix 4 will not be stripped so , here you have to create one more tranlation profile to strip prefix.
voice translation-rule 5
rule 1 /^1/ //
voice translation-profile sip_out
translate called 4
now apply it to sip dial-peer,
dial-peer voice 4 voip
destination-pattern 4xxx
session protocal sipv2
session targer <sip server>
translation-profile outgoing sip_out
fewwww, that was lot information.
i hope you may got the idea how you can do that.
Regards
Chintan
~please rate and mark answers as correct if helpful
now apply this t
08-11-2014 12:58 AM
tnx
very complicated solution
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