I'm only expecting NAT issues if not the latency issues. SIP traces of any such calls will make more sense.<br><br><div class="gmail_quote">On Mon, Jan 16, 2012 at 6:05 PM, Arlen Nascimento <span dir="ltr"><<a href="mailto:arlen.nascimento@gmail.com">arlen.nascimento@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">the client is aware of the adverse environment and this is the only solution for him<div class="HOEnZb"><div class="h5">
<br><br><div class="gmail_quote">On Mon, Jan 16, 2012 at 9:00 AM, Flavio Miranda <span dir="ltr"><<a href="mailto:flaviormiranda@hotmail.com" target="_blank">flaviormiranda@hotmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div><div dir="ltr">
Unless you are doing test with SIP under adverse environmet, that is not the point, but, if you intend to have Communication, you should worry about this detail. <br> Basic infra-estructure is the first thing to think in any new project.<br>
<br>Good luck!<div><br>Att,<br>
<br>
Flavio Roberto Miranda<br>
<a href="mailto:MSN%3Aflaviormiranda@hotmail.com" target="_blank">MSN:flaviormiranda@hotmail.com</a><br>Skype: flaviormiranda<br><br></div><div><div></div><hr>Date: Mon, 16 Jan 2012 07:58:34 -0400<br>From: <a href="mailto:arlen.nascimento@gmail.com" target="_blank">arlen.nascimento@gmail.com</a><br>
To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>Subject: Re: [asterisk-users] Peer doesn't answer<div><div><br><br>It is a satellite connection, so ping is about 500ms. I know it is not ok to keep a normal conversation, that is not the point.<br>
<br><br><div>On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda <span dir="ltr"><<a href="mailto:flaviormiranda@hotmail.com" target="_blank">flaviormiranda@hotmail.com</a>></span> wrote:<br>
<blockquote style="border-left:1px #ccc solid;padding-left:1ex">
<div><div dir="ltr">
Hi Arlen,<br><br> A reasonable time to Voip calls is about 250 ms. What about the Ping test end-to-end ? <br><br>Att,<br>
<br>
Flavio Roberto Miranda<br>
<a href="mailto:MSN%3Aflaviormiranda@hotmail.com" target="_blank">MSN:flaviormiranda@hotmail.com</a><br>Skype: flaviormiranda<br><br><div><div></div><hr>Date: Sun, 15 Jan 2012 21:53:46 -0400<br>From: <a href="mailto:arlen.nascimento@gmail.com" target="_blank">arlen.nascimento@gmail.com</a><br>
To: <a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a><br>Subject: [asterisk-users] Peer doesn't answer<div><div><br><br>Hi all,<br><br>i'm implementing an asterisk server that will have several peers connected by satellite links.<br>
When qualify=yes or some value (from 3000 to 50000), 'sip show peers' shows the peer as unreachable. In this case i can place calls from the phone in the satellite link, but can't call to it.<br>
When i turn off qualify, the status changes to unmonitored. In this case, I can make calls in both directions but the call is never established. The phone keeps ringing until 'ring time' expires even when I answer the call on the phone/softphone.<br>
<br>Any thoughts?<br><br>Regards,<br clear="all"><br>-- <br>Arlen Nascimento<br><br>
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