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

[Oswaldo Arratia]

Here is what should work for you.

In your Cisco
dial-peer voice xxxxx voip
 huntstop
 destination-pattern xxxxx  <- Extension number you want to dial
 progress_ind setup enable 3
 session protocol sipv2
 session target ipv4:y.y.y.y  <- Your * IP
 session transport udp
 dtmf-relay rtp-nte
 codec g711ulaw
 no vad 

In your *: 

EXTENSIONS.CONF:
exten => _xxxxx,1,Dial(SIP/${EXTEN},45,f)
[GW1]
include => extensions


SIP.CONF:
[general]
;progressinband=no

[GW1]
amaflags=billing
type=peer
host=x.x.x.x  <- Ip of your cisco gateway
defaultip=x.x.x.x  <- Ip of your cisco gateway
port=5060
context=GW1
disallow=all
allow=ulaw
allow=g729

That should solve your ringback problem

Of course, you have to adjust codecs, ports and context names to match your
setup.

Hope this helps.

Oswaldo A.


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Adam
Rothschild
Sent: Saturday, March 19, 2005 5:49 PM
To: asterisk-users at lists.digium.com
Cc: asr at latency.net
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

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