10-01-2008 08:56 PM - edited 03-15-2019 01:41 PM
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.
10-02-2008 04:26 AM
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
10-02-2008 06:23 AM
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
10-02-2008 06:34 AM
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
10-02-2008 06:49 AM
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.
10-02-2008 06:54 AM
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
10-02-2008 06:58 AM
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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