<u><b>Details :</b></u><br><br> * SIP Call<br> Direction: Outgoing<br> Call-ID: <a href="mailto:593722384f5174775f83837c30fd51b0@59.165.44.21">593722384f5174775f83837c30fd51b0@59.165.44.21</a><br>
Our Codec Capability: 256<br> Non-Codec Capability: 1<br> Their Codec Capability: 256<br> Joint Codec Capability: 256<br> Format g729<br> Theoretical Address: <a href="http://209.51.198.114:5060">209.51.198.114:5060</a><br>
Received Address: <a href="http://209.51.198.114:5060">209.51.198.114:5060</a><br> NAT Support: RFC3581<br> Audio IP: 59.XXX.XXX.XX (local)<br> Our Tag: as0c9f4a40<br> Their Tag: 0909210916425544477827101<br>
SIP User agent:<br> Username: 18186223080<br> Peername: sip209<br> Original uri: sip:<a href="http://209.51.198.114:5060">209.51.198.114:5060</a><br> Need Destroy: 0<br>
Last Message: Tx: ACK<br> Promiscuous Redir: No<br> Route: sip:209.51.198.114:5060;transport=udp<br> DTMF Mode: rfc2833<br> SIP Options: (none)<br><br><br><br>
<div class="gmail_quote">On Thu, Sep 10, 2009 at 2:06 AM, David @ULC <span dir="ltr"><<a href="mailto:ucoms2001@gmail.com">ucoms2001@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>Local/718186223080@d 718186223080@default Up Dial(SIP/18186223080@sip209||t<br><br><br>I see this in my Asterisk when I do <br><br>show channels<div><div></div><div class="h5"><br><br><br><br><div class="gmail_quote">
On Thu, Sep 10, 2009 at 1:49 AM, David @ULC <span dir="ltr"><<a href="mailto:ucoms2001@gmail.com" target="_blank">ucoms2001@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br>I don't know where is the problem. May be with VOIPSwitch OR may be with Asterisk..<br>
<br>Call getting stuck : My agent hang up the call but in Active calls , I see call connected and getting charged<br><br>I use VOIP and NOT PSTN<br>
<br>Didnt check the Asterisk CLI. Can I get any history of what asterisk REALLY had ?<div><div></div><div><br><br><br><br><div class="gmail_quote">On Wed, Sep 9, 2009 at 11:41 PM, David @ULC <span dir="ltr"><<a href="mailto:ucoms2001@gmail.com" target="_blank">ucoms2001@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">I am using asterisk.<br><br>I also have an access to VOIPSwitch ver 2 where I can see live calls.<br>
<br>Many times I have seen that my calls are getting strucked and then it gets disconneected after 59 mins ( as settings are done accordingly in VOIPSwitch)<br>
<br>What could be the reason ?<br>
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