Mailing List Archive

SIP INFO
Hello,

does anybody know how to setup a AS5350 to sent out SIP INFO messages
for DTMF tones generate on the PSTN?

In the dial-peer configuration i can only see options for DTMF via
RTP or H.323. But according to the feature list it should be
possible.

Christian.
Re: SIP INFO [ In reply to ]
On Sun, Feb 01, 2004 at 11:00:48AM -0800, Hielke Christian Braun wrote:
> Hello,
>
> does anybody know how to setup a AS5350 to sent out SIP INFO messages
> for DTMF tones generate on the PSTN?
>
> In the dial-peer configuration i can only see options for DTMF via
> RTP or H.323. But according to the feature list it should be
> possible.

here's an example:

dial-peer voice 1 voip
destination-pattern 1
session protocol sipv2
session target dns:hostname-sip-proxy
dtmf-relay rtp-nte
codec g711ulaw
no vad
!

This causes the system to use the 'Named Telephone Event'. This
is what I use in my environment of 7960's, Asterisk (asteriskpbx.org),
and Cisco routers that have varying voice modules installed in them.

herer's the other choices:

Router(config-dial-peer)#dtmf-relay ?
cisco-rtp Cisco Proprietary RTP
h245-alphanumeric DTMF Relay via H245 Alphanumeric IE
h245-signal DTMF Relay via H245 Signal IE
rtp-nte RTP Named Telephone Event RFC 2833
sip-notify DTMF Relay via SIP NOTIFY messages

I do encounter a few places when dialed that do not properly
pass/handle the DTMF properly, i'm not sure what the cause
of this is. Anyone with clues, i'm interested in hearing your problems
and/or solutions.

- Jared

--
Jared Mauch | pgp key available via finger from jared@puck.nether.net
clue++; | http://puck.nether.net/~jared/ My statements are only mine.
Re: SIP INFO [ In reply to ]
On Sun, Feb 01, 2004 at 03:13:35PM -0500, Jared Mauch wrote:
> On Sun, Feb 01, 2004 at 11:00:48AM -0800, Hielke Christian Braun wrote:
> > Hello,
> >
> > does anybody know how to setup a AS5350 to sent out SIP INFO messages
> > for DTMF tones generate on the PSTN?
> >
> > In the dial-peer configuration i can only see options for DTMF via
> > RTP or H.323. But according to the feature list it should be
> > possible.
>
> here's an example:
>
> dial-peer voice 1 voip
> destination-pattern 1
> session protocol sipv2
> session target dns:hostname-sip-proxy
> dtmf-relay rtp-nte
> codec g711ulaw
> no vad
> !
>
> This causes the system to use the 'Named Telephone Event'. This
> is what I use in my environment of 7960's, Asterisk (asteriskpbx.org),
> and Cisco routers that have varying voice modules installed in them.
>
> herer's the other choices:
>
> Router(config-dial-peer)#dtmf-relay ?
> cisco-rtp Cisco Proprietary RTP
> h245-alphanumeric DTMF Relay via H245 Alphanumeric IE
> h245-signal DTMF Relay via H245 Signal IE
> rtp-nte RTP Named Telephone Event RFC 2833
> sip-notify DTMF Relay via SIP NOTIFY messages
>
> I do encounter a few places when dialed that do not properly
> pass/handle the DTMF properly, i'm not sure what the cause
> of this is. Anyone with clues, i'm interested in hearing your problems
> and/or solutions.
>

Thanks for the answer. That is strange. I have a quite recent IOS on
the gateway (Version 12.3(5a)). But i only see both h245 and the
rtp-nte options. sip-notify and cisco-rtp are not shown. Do i need
any other configuration to enable sip-notify?

I am trying to get something similar to work. I have a Asterisk and
want to use DTMF when dialing in through the Cisco gateway to the
Asterisk. With DTMF via rtp-nte i only get an error "Unknown RTP
codec 19 received" on the Asterisk.

Could you maybe sent your Cisco configuration and the IOS version on
or of list?


Thanks,
Christian.