[cisco-voip] Question regarding new implementation...

[Bryan Deaver]

Stephen,

You might try the dial-peer subcommand 'progress_ind setup enable 3' on the
voip dial-peer to see if that helps.  See url below which talks about h323
but some of this is applicable for sip as well.

If this does not help, can you please send the version you are running along
with a debug trace showing a _single_ call from the 5400 to the ip phone.
This would include the following debug:
    - 'debug vtsp tone' (or debug voip vtsp tone if you are running a later
release)
        - if you are running a later release of software (something like
12.3(4)T and above), then get 'debug voip dsm tone' as well.
    - 'debug isdn q931'.  Assuming you are running isdn on the teelphony
side.
    - 'debug voip ccapi inout'

Also, get the output of 'show call active voice brief' when the call is in
alerting stage.

Bryan



http://www.cisco.com/en/US/tech/tk652/tk701/technologies_tech_note09186a0080
094c33.shtml

> From: Stephen Malenshek <stephen at valuelinx.net>
> Organization: Valuelinx Corporation, Inc.
> Date: Sun, 1 May 2005 14:54:12 -0500
> To: Cisco Systems VoIP List <cisco-voip at puck.nether.net>
> Subject: [cisco-voip] Question regarding new implementation...
> 
> I have recently implemented a SIP VoIP implementation using Asterisk.  I can
> go through and place a call to a particular number from the PSTN, the phone
> rings, but I am not getting the ring response back to the calling party.  I
> am not sure as to where this problem is coming from, but I know it stopped
> working once I added the configurations....
> 
> dial-peer voice 82010151 pots
>  incoming called-number 2010151
>  direct-inward-dial
>  forward-digits all
> !
> dial-peer voice 2010151 voip
>  destination-pattern 2010151
>  session protocol sipv2
>  session target ipv4:XXX.XXX.XXX.XXX
>  session transport udp
>  incoming called-number 2010151
>  dtmf-relay sip-notify rtp-nte
>  codec g711ulaw
> !
> dial-peer voice 82010152 pots
>  incoming called-number 2010152
>  direct-inward-dial
>  forward-digits all
> !
> dial-peer voice 2010152 voip
>  destination-pattern 2010152
>  session protocol sipv2
>  session target ipv4: XXX.XXX.XXX.XXX
>  session transport udp
>  incoming called-number 2010152
>  dtmf-relay sip-notify rtp-nte
>  codec g711ulaw
> !
> !
> sip-ua
>  max-forwards 15
>  retry invite 10
>  timers trying 1000
>  timers expires 300000
>  sip-server ipv4: XXX.XXX.XXX.XXX
>  no transport tcp
> !
> 
> I also have a Cisco Call Manager Express sending and receiving calls to and
> from this same equipment without the problem existing.  I am sure that this
> problem is something with the way that I have the SIP commands configured on
> this AS5400, but I just do not know enough to fix it.
> 
> Thanks for your thoughts.
> 
> Stephen
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