asterisk users - Nov 2003 - g723 to g723 SIP call - warning message

Hi,

I am calling from a grandstream phone with g723 codec through * to iconnect.
Incoming context as well as outgoing context set to g723.1 codec in *.

Call get connected and I can talk. However I get the following warning,
which scrolls on my screen until I hang-up.

[root@asterisk sath]# cat g723.1
- Executing SetCallerID("SIP/-08122ae0", "1001") in new
stack
    -- Executing AbsoluteTimeout("SIP/-08122ae0", "6000") in
new stack
    -- Set Absolute Timeout to 6000
    -- Executing Dial("SIP/-08122ae0",
"Sip/15105418168@iconnect|90|r") in
new stack
    -- Called 15105418168@iconnect
    -- SIP/iconnect-c682 is making progress passing it to SIP/-08122ae0
WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
detect p
rocess 1 frames
WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
detect p
rocess 1 frames
WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
detect p
rocess 1 frames
WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
detect p
rocess 1 frames
WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
detect p
rocess 1 frames
WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
detect p
rocess 1 frames
WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
detect p
rocess 1 frames

Here is my sip.conf

[general]
port=5060
context=default
allow=g723.1
maxexpirey=180
defaultexpirey=160
;Connect to iconnect
register=1510xxxxxx:xxxx@natrelay.deltathree.com/1510xxxxxx


[iconnect]
type=friend
secret=xxxx
username=xxxxxxx
host=natrelay.deltathree.com
dtmfmode=inband
canreinvite=no
context=vobb-in
allow=g723.1

Can someone be able to debug this ?

If I make the codec to g729, call not even get through. * complains that it
can't bridge the codec.

Now in GS phone I can see following setting;

Voice Frames per TX: 2    (up to 10/20/32/64 frames for G711/G726/G723/other
codecs respectively)

Could there be a mismatch here ?


Cheers

Sathya

Don't try to do inland DTMF on anything but G.711.

Jeremy McNamara



Sathya Weerasooriya wrote:
>Hi,
>
>I am calling from a grandstream phone with g723 codec through * to iconnect.
>Incoming context as well as outgoing context set to g723.1 codec in *.
>
>Call get connected and I can talk. However I get the following warning,
>which scrolls on my screen until I hang-up.
>
>[root@asterisk sath]# cat g723.1
>- Executing SetCallerID("SIP/-08122ae0", "1001") in new
stack
>    -- Executing AbsoluteTimeout("SIP/-08122ae0",
"6000") in new stack
>    -- Set Absolute Timeout to 6000
>    -- Executing Dial("SIP/-08122ae0",
"Sip/15105418168@iconnect|90|r") in
>new stack
>    -- Called 15105418168@iconnect
>    -- SIP/iconnect-c682 is making progress passing it to SIP/-08122ae0
>WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
>detect p
>rocess 1 frames
>WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
>detect p
>rocess 1 frames
>WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
>detect p
>rocess 1 frames
>WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
>detect p
>rocess 1 frames
>WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
>detect p
>rocess 1 frames
>WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
>detect p
>rocess 1 frames
>WARNING[1217602880]: File dsp.c, Line 1198 (ast_dsp_process): Unable to
>detect p
>rocess 1 frames
>
>Here is my sip.conf
>
>[general]
>port=5060
>context=default
>allow=g723.1
>maxexpirey=180
>defaultexpirey=160
>;Connect to iconnect
>register=1510xxxxxx:xxxx@natrelay.deltathree.com/1510xxxxxx
>
>
>[iconnect]
>type=friend
>secret=xxxx
>username=xxxxxxx
>host=natrelay.deltathree.com
>dtmfmode=inband
>canreinvite=no
>context=vobb-in
>allow=g723.1
>
>Can someone be able to debug this ?
>
>If I make the codec to g729, call not even get through. * complains that it
>can't bridge the codec.
>
>Now in GS phone I can see following setting;
>
>Voice Frames per TX: 2    (up to 10/20/32/64 frames for G711/G726/G723/other
>codecs respectively)
>
>Could there be a mismatch here ?
>
>
>Cheers
>
>Sathya
>
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>  
>

Hi,

Thanks Jeramy and Eric.

Sorry for my ignorance. I still did not get the point.

Do you mean that I have to set each of my context in sip.conf  with
dtmfmode=inband ?

I have the GS phone set as DTMF mode = Via SIP Info. Would that need to be
change to something else ?

(Send DTMF:    in-audio     via RTP (RFC2833)     via SIP INFO)

Cheers

Sathya


Date: Wed, 19 Nov 2003 06:15:35 -0600
From: Eric Wieling <eric@fnords.org>
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] g723 to g723 SIP call - warning message
Reply-To: asterisk-users@lists.digium.com

Jeremy McNamara wrote:
> Don't try to do inland DTMF on anything but G.711.
>
> Jeremy McNamara
>
Someone really needs to patch Asterisk to print some ugly warning or
notice to the Asterisk console when the codec that is being used for a
call is not ulaw/alaw and trhe dtmfmode=inband (manyually or
automagically set)