<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:times new roman,new york,times,serif;font-size:12pt"><div><font size="3"><span style="font-family: arial,helvetica,sans-serif;">I couldn't find any pointers for this online. Is it possible through Async AGI?</span></font> <span style="font-family: arial,helvetica,sans-serif;">Let me know if anybody is aware of doing this.</span><br></div><div style="font-family: times new roman,new york,times,serif; font-size: 12pt;"><br><div style="font-family: arial,helvetica,sans-serif; font-size: 13px;"><font size="2" face="Tahoma"><hr size="1"><b><span style="font-weight: bold;">From:</span></b> Martin <asterisklist@callthem.info><br><b><span style="font-weight: bold;">To:</span></b> Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><br><b><span style="font-weight: bold;">Sent:</span></b> Tuesday, 14 April, 2009
4:21:17 AM<br><b><span style="font-weight: bold;">Subject:</span></b> Re: [asterisk-users] Send Re-invite from Dialplan application?<br></font><br>sorry,<br><br>You can set SIP_CODEC before the call is answered ... most likely as<br>one of the first priorities.<br>It causes the 200 OK to INVITE contain the codec you specify as the first one.<br><br>I'm not aware of reinviting while in a call other than to switchover<br>to T.38 ... it can be coded in but I'm not<br>sure if it's already there<br><br>Martin<br><br>On Mon, Apr 13, 2009 at 6:05 AM, Sai P. Varanasi<br><<a ymailto="mailto:saiprabhakarv@yahoo.com" href="mailto:saiprabhakarv@yahoo.com">saiprabhakarv@yahoo.com</a>> wrote:<br>> Hi,<br>> I have a requirement where an IVR application on asterisk has to play a<br>> audio file in g729 and when a digit is pressed, the call should switch to<br>> another codec (say ulaw). So, What can I do in the extensions.conf to<br>>
trigger a re-negotiation of codec?<br>><br>> I used<br>> exten => 55xx,n,Set(SIP_CODEC=ulaw)<br>><br>> but, I suppose this affects the next call and not the current one.<br>><br>> Please help ASAP<br>><br>> Thanks,<br>> Sai<br>> ________________________________<br>> Add more friends to your messenger and enjoy! Invite them now.<br>> _______________________________________________<br>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>><br>> asterisk-users mailing list<br>> To UNSUBSCRIBE or update options visit:<br>> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>><br><br>_______________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com"
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