Are they really sending NI2? I'd confirm with them that they are NI2. I had a similar issue, with a different brand of Gateway, and it turned out they weren't sending NI2. Everything worked, but the Name. Once we lined up to the same ISDN protocol, it worked great.
From: [email protected] [mailto:[email protected]] On Behalf Of Becker, Jesse Sent: Wednesday, February 29, 2012 9:29 AM To: Discussion list for users of sipXecs software Subject: [sipx-users] No Caller ID Name on PSTN Calls 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 -- <http://www1.sunyulster.edu/images/logo.jpg> 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 <mailto:[email protected]> [email protected] | <http://www.sunyulster.edu/> 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/>
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