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Dial-Peer help

sdaniels44
Level 1
Level 1

Can someone help me with an example dial-peer configuration? I would like to see how to get a call coming in from a SIP VOIP network to get routed out a T1 trunk group. Thanks for the help in advance. Please let me know if I need to provide more info.

6 Replies 6

Marwan ALshawi
VIP Alumni
VIP Alumni

ok

lets say from the calls coming from the sip dial 1234... to go thoguh ur T1

now we need to creat an incoming VOIP dial-peer and an OUTgoing POTS dial-peer

the voip take the call from sip VOIP link and the pots route it out through ur T1

dial-peer voice 9 voip

incoming called-number 1234...

codec g711

dial-peer voice 91 pots

destination-pattren 1234...

port [ur t1 port]

forward-digits all

prefix 1234

or what ever diggits change u wanna do

good luck

if helpful Rate

Thank you for the reply,

I have taken your advice and I am still running into the same issue I had previously.

let me try to describe what is going on.

a sip call comes into my 3825 with the dnis 82589.

If I debug dial-peers I can see that it matches the incoming dial peer you had me enter. I can also see that it matches the outgoing dial-peer as well.

The issue is that sometimes the call makes it just fine, but about 60% of the time I get "I'm sorry there are no available lines to complete your call"

Let me paste the relevant entrys from my config:

card type t1 0 0

card type t1 0 1

card type t1 0 2

card type t1 0 3

network-clock-participate wic 0

network-clock-participate wic 1

network-clock-participate wic 2

network-clock-participate wic 3

network-clock-select 1 T1 0/0/0

network-clock-select 2 T1 0/0/1

isdn switch-type primary-dms100

voice-card 0

no dspfarm

trunk group 1

translation-profile incoming profile1

translation-profile outgoing profile1

voice service voip

fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco

modem passthrough nse codec g711ulaw

voice translation-rule 1

rule 1 // // type any national plan any isdn

voice translation-profile profile1

translate calling 1

translate called 1

controller T1 0/0/0

framing esf

linecode b8zs

pri-group timeslots 1-24 nfas_d primary nfas_int 0 nfas_group 1

description Sales Primary NFAS

!

controller T1 0/0/1

framing esf

linecode b8zs

pri-group timeslots 1-24 nfas_d backup nfas_int 1 nfas_group 1

description Sales Backup NFAS

!

controller T1 0/1/0

framing esf

linecode b8zs

pri-group timeslots 1-24 nfas_d none nfas_int 2 nfas_group 1

description Sales NFAS

!

controller T1 0/1/1

framing esf

linecode b8zs

pri-group timeslots 1-24 nfas_d none nfas_int 3 nfas_group 1

description Sales NFAS

!

controller T1 0/2/0

framing esf

linecode b8zs

ds0-group 0 timeslots 1-24 type e&m-wink-start dtmf dnis

cas-custom 0

trunk-group 1

description Concerto Tie #1

!

controller T1 0/2/1

framing esf

linecode b8zs

ds0-group 0 timeslots 1-24 type e&m-wink-start dtmf dnis

cas-custom 0

trunk-group 1

description Concerto Tie #1

!

controller T1 0/3/0

framing esf

linecode b8zs

ds0-group 0 timeslots 1-24 type e&m-wink-start dtmf dnis

cas-custom 0

trunk-group 1

description Concerto Tie #1

!

controller T1 0/3/1

framing esf

linecode b8zs

ds0-group 0 timeslots 1-24 type e&m-wink-start dtmf dnis

cas-custom 0

trunk-group 1

description Concerto Tie #1

interface Serial0/0/0:23

no ip address

encapsulation hdlc

no logging event link-status

isdn switch-type primary-dms100

isdn incoming-voice voice

no cdp enable

voice-port 0/0/0:23

!

voice-port 0/1/0:23

!

voice-port 0/2/0:0

shutdown

!

voice-port 0/3/0:0

!

voice-port 0/0/1:23

!

voice-port 0/1/1:23

!

voice-port 0/2/1:0

!

voice-port 0/3/1:0

dial-peer cor custom

name Local

name LD

name International

dial-peer voice 2 pots

description Outbound dial peer via TIE-LINE

translation-profile outgoing profile1

preference 1

destination-pattern 8T

direct-inward-dial

!

dial-peer voice 101 voip

description IC Server #1

preference 3

service session

destination-pattern ....

session protocol sipv2

session target ipv4:10.2.1.13

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

dial-peer voice 102 voip

description IC Server #2

preference 2

service session

destination-pattern ....

session protocol sipv2

session target ipv4:10.2.1.12

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

dial-peer voice 3 pots

description Sales NFAS

translation-profile outgoing profile1

preference 1

destination-pattern 6T

direct-inward-dial

port 0/0/0:23

!

dial-peer voice 1 pots

description Default Incoming Dial Peer

incoming called-number .

direct-inward-dial

!

dial-peer voice 9 voip

incoming called-number 82589

codec g711ulaw

!

dial-peer voice 91 pots

trunkgroup 1 1

destination-pattern 82589

forward-digits all

prefix 2589

attached is output from a dialpeer debug

if the called number come to the router as 82589 matched dial -peer 9 as incoming

the outgoing dial-peer has not port assigned to be send through

dial-peer voice 91 pots

trunkgroup 1 1

destination-pattern 82589

forward-digits all

prefix 2589

u need to add a port

by the way the called number based on the above dialpeer will be send to the pstn as 2589 !

good luck

try it and let me know

You are quick to respond! thanks!

Sorry, I should let you know that this trunk group is not going to the PSTN but rather a dialer system (Concerto) it expects the DNIS of 2589.

I didn't specify a port, but i did tell it to send it down trunkgroup 1, will this not work properly?

Sorry if I sound like a noob, but I am new to this.

its ok

do u have the cloking between ur router and the system seted up corectly

when u do show isdn status do u have L1 L2 link state as connected or established

Someone else mentioned that the clocking may be screwed up...

I am not sure how to check this, this trunk is not ISDN PRI.

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