Dial-Peer help

Unanswered Question
Oct 1st, 2008
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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.

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Marwan ALshawi Thu, 10/02/2008 - 04:26
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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

sdaniels44 Thu, 10/02/2008 - 06:23
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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











Marwan ALshawi Thu, 10/02/2008 - 06:34
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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


sdaniels44 Thu, 10/02/2008 - 06:49
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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.

Marwan ALshawi Thu, 10/02/2008 - 06:54
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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

sdaniels44 Thu, 10/02/2008 - 06:58
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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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