All,
We just cutover our PRIs from a Cisco 2811 to our new Patton gateways.
Everything is working fine, however, we do not have caller-id name on
incoming calls.
We have three PRIs from PAETEC using NI-2 to the Pattons to Sipx. Phones
are Polycom with 3.2.6 firmware.
Below is the patton config. Masked IPs and password:
no terminal telnet
administrator administrator password < omitted >
clock local default-offset +00:00
dns-client server xxx.xxx.xxx.10
dns-relay
webserver port 80 language en
sntp-client
sntp-client server primary xxx.xxx.xxx.10 port 123 version 4
system hostname SRCGW.sunyulster.edu
system
ic voice 0
pcm law-select uLaw
system
clock-source 1 e1t1 0 0
profile r2 default
profile napt NAPT_WAN
profile ppp default
profile call-progress-tone US_Dialtone
profile call-progress-tone US_Alertingtone
play 1 2000 440 -10 480 -19
pause 2 4000
profile call-progress-tone US_Busytone
play 1 500 480 -24 620 -24
pause 2 500
profile tone-set default
profile tone-set US
map call-progress-tone dial-tone US_Dialtone
map call-progress-tone ringback-tone US_Alertingtone
map call-progress-tone busy-tone US_Busytone
map call-progress-tone release-tone US_Busytone
map call-progress-tone congestion-tone US_Busytone
profile voip default
codec 1 g711ulaw64k rx-length 20 tx-length 20
codec 2 g711alaw64k rx-length 20 tx-length 20
dtmf-relay rtp
flash-hook-relay rtp
rtp traffic-class local-default
fax transmission 1 relay t38-udp
fax detection fax-frames
profile pstn default
profile sip default
no autonomous-transitioning
profile aaa default
method 1 local
method 2 none
context ip router
interface WAN
ipaddress dhcp
use profile napt NAPT_WAN
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu
interface LAN
ipaddress xxx.xxx.xxx.3 255.255.255.0
tcp adjust-mss rx mtu
tcp adjust-mss tx mtu
context ip router
route 0.0.0.0 0.0.0.0 xxx.xxx.xxx.1 0
context cs switch
digit-collection timeout 4
routing-table called-e164 SIP_TO_PSTN
route default dest-service OUTBOUND
routing-table called-e164 PSTN_TO_SIP
route .%T dest-interface IF_SIPX
mapping-table called-e164 to called-e164 STIP-ALL
map default to ""
interface isdn IF_PRI_1
route call dest-table PSTN_TO_SIP
use profile tone-set US
caller-name send-information-following
user-side-ringback-tone
interface isdn IF_PRI_2
route call dest-table PSTN_TO_SIP
use profile tone-set US
caller-name send-information-following
user-side-ringback-tone
interface isdn IF_PRI_3
route call dest-table PSTN_TO_SIP
use profile tone-set US
caller-name send-information-following
user-side-ringback-tone
interface isdn IF_PRI_4
route call dest-table PSTN_TO_SIP
use profile tone-set US
caller-name send-information-following
user-side-ringback-tone
interface sip IF_SIPX
bind context sip-gateway GW-SIP
route call dest-table SIP_TO_PSTN
remote sipx.sunyulster.edu
overlap-dialing new-transaction emit
service hunt-group OUTBOUND
drop-cause normal-unspecified
drop-cause no-circuit-channel-available
drop-cause network-out-of-order
drop-cause temporary-failure
drop-cause switching-equipment-congestion
drop-cause access-info-discarded
drop-cause circuit-channel-not-available
drop-cause resources-unavailable
route call 1 dest-interface IF_PRI_1
route call 2 dest-interface IF_PRI_2
route call 3 dest-interface IF_PRI_3
route call 4 dest-interface IF_PRI_4
context cs switch
no shutdown
location-service SIPX_SERVER
domain 1 sipx.sunyulster.edu
context sip-gateway GW-SIP
interface IF_SIPX
bind interface LAN context router port 5060
context sip-gateway GW-SIP
bind location-service SIPX_SERVER
no shutdown
port ethernet 0 0
medium auto
encapsulation ip
bind interface WAN router
shutdown
port ethernet 0 1
medium auto
encapsulation ip
bind interface LAN router
no shutdown
port e1t1 0 0
port-type t1
clock slave
linecode b8zs
framing esf
encapsulation q921
q921
uni-side user
encapsulation q931
q931
protocol ni2
uni-side user
bchan-number-order ascending-cyclic
encapsulation cc-isdn
bind interface IF_PRI_1 switch
port e1t1 0 0
no shutdown
port e1t1 0 1
port-type t1
clock slave
linecode b8zs
framing esf
encapsulation q921
q921
uni-side user
encapsulation q931
q931
protocol ni2
uni-side user
bchan-number-order ascending-cyclic
encapsulation cc-isdn
bind interface IF_PRI_2 switch
port e1t1 0 1
no shutdown
port e1t1 0 2
port-type t1
clock slave
linecode b8zs
framing esf
encapsulation q921
q921
uni-side user
encapsulation q931
q931
protocol ni2
uni-side user
bchan-number-order ascending-cyclic
encapsulation cc-isdn
bind interface IF_PRI_3 switch
port e1t1 0 2
no shutdown
port e1t1 0 3
port-type t1
clock slave
linecode b8zs
framing esf
encapsulation q921
q921
uni-side user
encapsulation q931
q931
protocol ni2
uni-side user
bchan-number-order ascending-cyclic
encapsulation cc-isdn
bind interface IF_PRI_4 switch
port e1t1 0 3
no shutdown
We had caller ID when the PRIs were on the Cisco side. No changes were made
at PAETEC. Any ideas on what to change to allow name of caller from PSTN?
Thanks,
Jes
--
Jesse Becker
Technical Manager
Office of Information Technology
Network+ | Linux+ Certified Professional
DATATEL+SGHE @ SUNY Ulster
491 Cottekill Road, Stone Ridge, NY 12484
Tel 845-687-5064 | Fax 845-687-5105
[email protected] | www.sunyulster.edu
<http://www.sunyulster.edu/>
Open or check the status of a ticket by visiting Helpdesk
Online<https://helpdesk.sunyulster.edu/>
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