[Asterisk-Users] Asterisk and Cisco AS53xx/54xx Access Server Platform

[Adam Rothschild]

Hello,

I've got an ISDN PRI circuit terminating in a Cisco AS5350, which in
turn is talking to an Asterisk server via SIP for call origination and
termination.  Seems simple enough, and it works for the most part,
but:

1) Caller ID name data comes in on the PRI, but doesn't appear to get
   handed off to the Asterisk server via SIP, at least not in any
   format that Asterisk understands.  Caller ID _number_ works fine.

   (I'm guessing this has something to do with the 'remote-party-id'
   header, but I've tried with it both enabled and disabled in the
   'sip-ua' IOS configuration stanza.)

2) Ring tone is not generated or audible on PSTN -> AS5350 -> Asterisk
   -> Dial([...],,r) calls placed.  Music on hold ([...],,m) works
   fine.

Clues appreciated, on or off list.  Relevant 'show' output and
configuration snippets below...

Thanks, and my apologies for the cross-posting,
-a

--==--
Router#sh ver
Cisco Internetwork Operating System Software 
IOS (tm) 5350 Software (C5350-IK9S-M), Version 12.3(13), RELEASE SOFTWARE (fc2)

Router#show spe ver
IOS-Bundled Default Firmware-Filename      Version        Firmware-Type
=====================================      ============   =============
system:/ucode/spe_firmware-1               0.10.2.2        SPE firmware

On-Flash Firmware-Filename                 Version        Firmware-Type
=====================================      ============   =============

  SPE-#   Type   Port-Range           Version   UPG Firmware-Filename
   1/00  CSMV6    0000-0005          0.10.2.2   N/A ios-bundled default
[...]

Router#show conf
[...]
spe country t1-default
isdn switch-type primary-ni
!
voice hunt user-busy
voice call send-alert
voice call convert-discpi-to-prog
voice rtp send-recv
voice service voip 
 fax protocol pass-through g711alaw
 h323
 sip
  bind all source-interface FastEthernet0/0
controller T1 3/0
 framing esf
 linecode b8zs
 pri-group timeslots 1-24
interface Serial3/0:23
 description T1 to CLEC
 no ip address
 load-interval 30
 isdn switch-type primary-ni
 isdn incoming-voice modem
 no cdp enable
voice-port 3/0:D
 bearer-cap Speech
dial-peer voice 1 pots
 incoming called-number 21255512[00-50]
 direct-inward-dial
!
dial-peer voice 100 voip
 destination-pattern 21255512[00-50]
 progress_ind setup enable 3
 session protocol sipv2
 session target ipv4:10.10.10.10
 codec g711ulaw
 no vad
!
dial-peer voice 1000 pots
 destination-pattern ..........
 port 3/0:D
sip-ua 
 retry invite 4
 retry response 3
 retry bye 2
 retry cancel 2
 sip-server ipv4:10.10.10.10
line 1/00 1/59
 no flush-at-activation
 no modem InOut
 transport input all
line 2/00 2/59
 no flush-at-activation
 no modem InOut
 transport input all