<div dir="ltr">Hello,<br><br>Thanks for your replies.<br><br>We checked our sip.conf and we have canreinvite=no already. I agree it could be a firmware issue. I will get another vendors phone hooked up to the pbx before going crazy with support.<br>
<br>Thanks,<br>Neal<br><br><br><br><div class="gmail_quote">On Sun, Oct 12, 2008 at 6:14 AM, Vieri <span dir="ltr"><<a href="mailto:rentorbuy@yahoo.com">rentorbuy@yahoo.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;">
<div class="Ih2E3d"><br>
--- On Sat, 10/11/08, Eric "ManxPower" Wieling <<a href="mailto:eric@fnords.org">eric@fnords.org</a>> wrote:<br>
<br>
> Try setting canreinvite=no in each of the device sections on<br>
> a couple of<br>
> phones, reload chan_sip.so and see if that fixes things.<br>
> It has fixed<br>
> the issue when I've tried it.<br>
><br>
> <a href="mailto:nrbwpi@gmail.com">nrbwpi@gmail.com</a> wrote:<br>
> > Hello,<br>
> ><br>
> > We are using asterisk 1.6, sangoma pri card, and Cisco<br>
> 7960 phones. When we<br>
> > make or receive calls there is a delay before voice is<br>
> heard. Anyone have<br>
> > any ideas on where to start to debug or has anyone<br>
> seen this before. We<br>
> > have played with settings on pri, asterisk, and phones<br>
> with no change.<br>
<br>
</div>I'm having the same problem but with ATA-connected analog phones. The ATAs are Grandstream GXW4008 with firmware v. <a href="http://1.0.1.15" target="_blank">1.0.1.15</a>. The "canreinvite" option in sip.conf doesn't change anything for me. Downgrading the GXW4008 solves this issue so this is obviously a firmware bug in my case. I had a vague report once of a user in another installation having this 1-second delay on call connection. That user had a Cisco phone but I don't remember which one. I suggest you check this with Cisco Support if you can.<br>
<font color="#888888"><br>
Vieri<br>
</font><div><div></div><div class="Wj3C7c"><br>
<br>
<br>
<br>
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