asterisk users - Oct 2009 - Asterisk with a Cisco AS5300 gateway

Hi

i test a new equipment on my backbone: a Cisco AS5300 with voice dsp 
ressource
connected at a E1 Voice Link.

I want that all call incoming on the cisco 5300 are sent to Asterisk and 
all Asterisk outgoing
call are sent to Cisco AS5300.

Actually, i configure the AS5300:

isdn switch-type primary-net5
!
voice service voip
 sip
!
voice class codec 400
 codec preference 1 g711alaw
 codec preference 2 g729r8
 codec preference 3 g723r63
 codec preference 4 g711ulaw
!
voice class codec 500
 codec preference 1 g729r8
 codec preference 2 g723r63
!
controller E1 0
 framing NO-CRC4
 pri-group timeslots 1-31
 description E1 Beta-Test


interface Serial0
 no ip address
 shutdown
 clock rate 2015232
 no fair-queue
!
interface Serial1
 no ip address
 shutdown
 clock rate 2015232
 no fair-queue
!
interface Serial2
 no ip address
 shutdown
 clock rate 2015232
 no fair-queue
!
interface Serial3
 no ip address
 shutdown
 clock rate 2015232
 no fair-queue
!
interface Serial0:15
 no ip address
 encapsulation ppp
 isdn switch-type primary-net5
 no cdp enable


voice-port 0:D
!
!
!
dial-peer voice 10 voip
 destination-pattern .T
 session protocol sipv2
 session target ipv4:IP_OF_ASTERISK:5060
 session transport udp
 dtmf-relay rtp-nte
 codec g711alaw
 no vad
!
dial-peer voice 42 pots
 destination-pattern .T
 direct-inward-dial
 port 0:D
!
sip-ua
 retry invite 3
 retry response 3
 retry bye 3
 retry cancel 3
 timers trying 1000
 sip-server ipv4:IP_OF_ASTERISK
!



Actually, a Tcpdump on my Asterisk server don't see any trafic between 
asterisk and cisco
and when i call a phone number that arrives on the E1, it's "busy"

anyone have a idea ?

bye
jerome

I don't have any experience with E1, but here are some comments from
the T1 perspective (on a 2800 series Cisco).  Here is also a link to
my collection of Cisco voice debugging commands:
http://thurmantech.com/node/5

On Thu, Oct 15, 2009 at 3:27 AM, Phibee Network Operation Center
<noc at phibee.net> wrote:

! Something like this to define what clock to use (internal usually
causes a lot of Slip/Error seconds)
! but don't quote me on the following line, I haven't used E1 or the
5300
!  network-clock-select 1 E1 0
> isdn switch-type primary-net5
> !
> voice service voip
> ?sip
> !
> voice class codec 400
> ?codec preference 1 g711alaw
> ?codec preference 2 g729r8
> ?codec preference 3 g723r63
> ?codec preference 4 g711ulaw
> !
! You don't seem to use either voice class, do you need
both?
> voice class codec 500
> ?codec preference 1 g729r8
> ?codec preference 2 g723r63
> !
> controller E1 0
> ?framing NO-CRC4
! linecode ?
! cablelength ?
> ?pri-group timeslots 1-31
> ?description E1 Beta-Test
> !
> interface Serial0:15
> ?no ip address
> ?encapsulation ppp
> ?isdn switch-type primary-net5
isdn incoming-voice voice
> ?no cdp enable
>
>
> voice-port 0:D
> !
> !
> !
> dial-peer voice 10 voip
destination-pattern .
redirect ip2ip
> ?session protocol sipv2
> ?session target ipv4:IP_OF_ASTERISK:5060
> ?session transport udp
> ?dtmf-relay rtp-nte
! ?codec g711alaw
! If you define the codec class, might as well use it
voice-class codec 400
dtmf-relay rtp-nte
> ?no vad
> !
> dial-peer voice 42 pots
! Don't make both patterns the same, maybe add a trunk prefix here
! ?destination-pattern .T
destination-pattern 8T
incoming called-number .T
> ?direct-inward-dial
> ?port 0:D
> !
> sip-ua
> ?retry invite 3
> ?retry response 3
> ?retry bye 3
> ?retry cancel 3
> ?timers trying 1000
! Don't need this, since you specified it on the dial-peer
!  sip-server ipv4:IP_OF_ASTERISK
> !
Good luck!

-Jonathan

On Thu, Oct 15, 2009 at 6:27 AM, Phibee Network Operation Center
<noc at phibee.net> wrote:
> dial-peer voice 10 voip
> ?destination-pattern .T
> ?session protocol sipv2
> ?session target ipv4:IP_OF_ASTERISK:5060
> ?session transport udp
> ?dtmf-relay rtp-nte
> ?codec g711alaw
> ?no vad
> !
> dial-peer voice 42 pots
> ?destination-pattern .T
> ?direct-inward-dial
> ?port 0:D
What does "destination-pattern .T" mean? I'm not familiar with
what
".T" would match. I would suggest using a more specific pattern that
you expect to be coming down the line.
> Actually, a Tcpdump on my Asterisk server don't see any trafic between
> asterisk and cisco
> and when i call a phone number that arrives on the E1, it's
"busy"
Doesn't see any traffic when? When the asterisk tries to call the
Cisco? That would suggest you have a sip.conf misconfiguration on
asterisk.

No traffic when the Cisco tries to call the asterisk? That could be
for a number of reasons. I would suggest your destination-pattern
could be bad, since I don't know what that syntax means. If an E1
works like a PRI T1, when you dial in, a DNIS is getting pushed to the
Cisco, and that's what you should match your destination-pattern on.
Regardless, SOMETHING is getting pushed down the wire when an E1 call
comes in, and you're getting a busy because Cisco has no matching
dial-peers.

Finally, it's rather embarrassing that you're asking this from a
'network operations center' email address. How about using a personal
email address with your real name?