if is a codec problem start putting all the systems in the same codec and disallow all other.<br>put for example all in alaw and disalaw all other includeing ulaw.<br><br>you can make calls from asterisk to the sip extencion registered in opensip?<br>
check that you can start the call from a soft phone or you can use orginate and send the call to open sip one side and music on hold in the other side.<br><br>David<br><br><div class="gmail_quote">2009/2/24 michel freiha <span dir="ltr"><<a href="mailto:michofr@gmail.com">michofr@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div dir="ltr">Dera All,<br><br>I have the following scenario,<br>A customer dial a DID number...The call is routed to a PSTN GW that send the call to asterisk...<br>
On asterisk I created an AGI Script that send the call to an extension registered on OpenSIPS server...<br>
The extension is ringing successfully, but as soon as I accept the call on OpenSIPS side the call is hangd up...<br>I checked rhe SIP debug and it seems that I have a Codec issue as you can see on <a href="http://pastebin.com/m767a2172" target="_blank">http://pastebin.com/m767a2172</a><br>
<br>Need some help please<br><br></div>
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