Did you solve your problem? I have the same setup and it works for me. -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Rothschild Sent: Saturday, March 19, 2005 5:49 PM To: [email protected] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk and Cisco AS53xx/54xx Access ServerPlatform
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 _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
