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[Asterisk-Dev] Some quiestions

 
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PostPosted: Sun Mar 16, 2003 8:50 pm    Post subject: [Asterisk-Dev] Some quiestions

Hi.

Can anyboyde explain why I get this error :

NOTICE[18446]: File rtp.c, Line 291 (ast_rtp_read): Unknown RTP codec 13
received

... And what DTMF doesn'r work in incoming calls from my SIP provider ?

/Regards Mike



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PostPosted: Sun Mar 16, 2003 9:15 pm    Post subject: [Asterisk-Dev] Some quiestions

On Sun, 2003-03-16 at 14:50, Mikael Andersson wrote:
Quote:
Hi.

Can anyboyde explain why I get this error :

NOTICE[18446]: File rtp.c, Line 291 (ast_rtp_read): Unknown RTP codec 13
received

I'm pretty sure this was covered, and is in the archives for either the
users list, or the parent to the user and dev list.

Quote:
... And what DTMF doesn'r work in incoming calls from my SIP provider ?

Which provider? Do you know if they send in band DTMF, out of band DTMF,
or what?

Quote:
http://lists.digium.com/mailman/listinfo/asterisk-dev
--

Steven Critchfield <critch@basesys.com>


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PostPosted: Sun Mar 16, 2003 11:40 pm    Post subject: [Asterisk-Dev] Some quiestions

Steven Critchfield wrote:

Quote:


>... And what DTMF doesn'r work in incoming calls from my SIP provider ?
>
>

Which provider? Do you know if they send in band DTMF, out of band DTMF,
or what?




As far as I can tell no "out of band DTMF" on incoming calls is working
at all,
unless the incoming calls comes to a registered user. This got broke
late last week
sometime. I have not check todays CVS. YMMV


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PostPosted: Mon Mar 17, 2003 6:35 pm    Post subject: [Asterisk-Dev] Some quiestions

Quote:
This got broke late last week
sometime. I have not check todays CVS.

FYI, for those of you had SIP break last week after downloading the latest
CVS version, please download the newest CVS now, and try again.

Unfortunately my changes (to support dynamic RTP payload types)
inadvertently broke outgoing SIP calls, but Mark Spencer and I seem to have
fixed the problems since then.

Ross.


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PostPosted: Tue Mar 18, 2003 7:53 am    Post subject: [Asterisk-Dev] Some quiestions

Todays CVS, fixed everything except the dtmf problem.
I'll explain the dtmf problem in more detail. I do not
register phones with Asterisk, So this problem may only
effect a few folks like me.

As I have no phone profiles in sip.conf for phones,
I can not use "dtmfmode=rfc2833" in each profile
which fixed a simpler problem for others. If I use
"dtmfmode=rfc2833" in the [general] I can enter
digits in the voice mail app, but then can not use
#transfer. If I use "dtmfmode=inband" I can use
#transfer but then the voice mail app will not
acknowledge digits typed in via my Cisco 7960
phones.

LOL, so I keep going back to last weeks code because
I can both use the voice mail, and #transfer.

No big deal till some feature gets added in CVS that I
just got to have!!! <smile>

I'm trying to make the time to get to know the code
so I can fix thing like this myself. To have spare time!


Ross Finlayson wrote:

Quote:

> This got broke late last week
> sometime. I have not check todays CVS.


FYI, for those of you had SIP break last week after downloading the
latest CVS version, please download the newest CVS now, and try again.

Unfortunately my changes (to support dynamic RTP payload types)
inadvertently broke outgoing SIP calls, but Mark Spencer and I seem to
have fixed the problems since then.

Ross.

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