Hi Pierre,<br><br>You can also add the following if you think its helpful.<br><br>relaxdtmf = yes ;relaxes the DTMF detection parameters<br>qualify = yes ; will send a SIP Optoin command regularly to check if the device is still online, if the device did not answer within 2 seconds it will be considered offline (default time is 2 seconds but can be configured)
<br><br>Best Regards,<br>Joanna<br><br><br><div><span class="gmail_quote">On 2/21/07, <b class="gmail_sendername">Benjamin Jacob</b> <<a href="mailto:benjamin.jacob@mgl.com">benjamin.jacob@mgl.com</a>> wrote:</span>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">rfc2833 is the prefered way, as inband will work perfectly only with the<br>ulaw codec.<br><br>
Pierre Marceau wrote:<br><br>>Okay, in the SPA-941 admin I changed:<br>><br>>;DTMF Tx Method: Auto<br>>DTMF Tx Method: Inband<br>><br>>and now it works.<br>><br>>Thanks!<br>>Pierre<br>><br>>
<br>><br>>>>><a href="mailto:benjamin.jacob@mgl.com">benjamin.jacob@mgl.com</a> 2/21/2007 12:09 AM >>><br>>>>><br>>>>><br>>Pierre,<br>>Thats exactly what Joanna said in her reply.
<br>>Check the client DTMF settings on your phones.<br>>set it to rfc2833 or out-of-band, whatever the config says.<br>><br>>Grandstream by default have inband DTMF set, and usualy ulaw is<br>>supported as well, and thats the reason ur grandstream works but others
<br>>dont.<br>><br>>cheerz<br>>- Ben.<br>><br>>Pierre Marceau wrote:<br>><br>><br>><br>>>Hi Joanna,<br>>><br>>>Thanks for your reply.<br>>><br>>>In my mind I think it must be some setting in the client (phone) becasue the Grandstream GXP 2000 does work and it is using the same
sip.conf<br>>><br>>>Extensions:<br>>>6000 is xlite softfone<br>>>6003 is Linksys SPA941<br>>>6004 is Grandstream GXP 2000<br>>>6005 is Linksys PAP2NA<br>>><br>>>Please have a look at my sip conf and suggest any changes I could try...
<br>>><br>>>[general]<br>>>context=internal<br>>>bindport=5060<br>>>bindaddr=<a href="http://0.0.0.0">0.0.0.0</a><br>>>srvlookup=yes<br>>>type=friend<br>>>secret=XXXXXXX<br>
>>nat=no<br>>>host=dynamic<br>>>dtmfmode=rfc2833<br>>>disallow=all<br>>>allow=ulaw<br>>>subscribecontext=internal<br>>>canreinvite=no<br>>>register=<a href="mailto:8885551234:XXXXXXXXXXXX@proxy.atlasvoice.com">
8885551234:XXXXXXXXXXXX@proxy.atlasvoice.com</a><br>>><br>>>[atlasvoice]<br>>>type=friend<br>>>host=<a href="http://proxy.atlasvoice.com">proxy.atlasvoice.com</a><br>>>username=8885551234<br>
>>secret=XXXXXXX<br>>>fromuser=8885551234<br>>>fromdomain=<a href="http://proxy.atlasvoice.com">proxy.atlasvoice.com</a><br>>>canreinvite=no<br>>>insecure=very<br>>>nat=yes<br>>>context=incoming
<br>>><br>>>[6000]<br>>>mailbox=6000@internal<br>>>[6001]<br>>>[6003]<br>>>[6004]<br>>>[6005]<br>>>[6006]<br>>>[6007]<br>>>[6008]<br>>><br>>><br>>>Thanks,
<br>>>Pierre<br>>><br>>><br>>><br>>><br>>><br>>><br>>>>>><a href="mailto:joannaliza@gmail.com">joannaliza@gmail.com</a> 2/20/2007 10:47 PM >>><br>>>>>>
<br>>>>>><br>>>>>><br>>>>>><br>>>Hi Pierre,<br>>><br>>>Just a thought..check your dtmfmode in your SIP client configuration, if<br>>>your using inband but your codec is not ulaw or alaw the DTMF tones will be
<br>>>misrepresented and thus will not be recognised due to the audio compression,<br>>>on the other hand if your phones are rfc2833 and asterisk is set to inband<br>>>you wont hear anything.<br>>>
<br>>>Hope that helps.<br>>><br>>>Best Regards,<br>>>Joanna<br>>><br>>>On 2/21/07, Pierre Marceau <<a href="mailto:pierre@forestcitynetwerxs.com">pierre@forestcitynetwerxs.com</a>> wrote:
<br>>><br>>><br>>><br>>><br>>>>Hello,<br>>>><br>>>>I can call out to the PSTN and talk to people but when I have to enter a<br>>>>dtmf tone in an ivr or voicemail system those systems do not recognise that
<br>>>>I have sent a tone. This is the case when I make the call with the Xlite<br>>>>softfone or a regular telephone plugged into a PAP2NA or a Linksys SPA941.<br>>>><br>>>>However... a Grandstream GXP2000 works just great ???
<br>>>><br>>>>All are extensions on my Asterisk 1.4 box. I am using a voip trunk through<br>>>>Atlasvoice. All extensions are setup identical in sip.conf.<br>>>><br>>>>One last thing, if a system wants me to respond 1 for sales 2 for service
<br>>>>I can hit the 1 button quickly 4 or 5 times and the remote system will get<br>>>>it. That does not work for a three digit extension as you may well imagine.<br>>>><br>>>>Any help would be appreciated.
<br>>>><br>>>>Pierre<br>>>><br>>>>_______________________________________________<br>>>>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --
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