Run a debug on the Patton and see if paetec is sending name and how it is mapped. If it was working before its not the pbx or phones, its the caller I'd specific settings in the gateway. On Feb 29, 2012 12:29 PM, "Becker, Jesse" <[email protected]> wrote:
> 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/> > Look up answers to frequently asked questions by visiting the Knowledge > Base <https://kb.sunyulster.edu/> > > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
_______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
