Run tcpdump with portrange 10000-20000 (or what range of ports you use for rtp) and dial test ivr to see what happening.<br><br><div class="gmail_quote">2011/10/17 gincantalupo <span dir="ltr"><<a href="mailto:gincantalupo@fgasoftware.com">gincantalupo@fgasoftware.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><u></u>
<div text="#000000" bgcolor="#ffffff">
Hi John,<br>
<br>
there is no firewall:<br>
<br>
snom <--> pbx <--> patton <--> pstn<br>
<br>
It happens ONLY with IVRs. Normal calls are fine. How can it be?<br>
<br>
I call my pbx from the customer pbx: when I directly call my phone
it works, when I call a test ivr it does not work...can a timing
problem be the cause???<br>
<br>
Giorgio<div><div></div><div class="h5"><br>
<br>
<br>
On 10/14/2011 03:48 PM, John Knight wrote:
<blockquote type="cite">Hi Giorgio,<br>
<br>
This behavior usually indicates some sort of firewall issue where
either inbound or outbound rtp traffic (the voice) is being
blocked or not routed correctly, though the SIP traffic makes it
through (as the call is being set up correctly). This could also
be multiple SIP extensions attempting to register over the same
port from a single location.<br>
<br>
What kind of firewall/router is being used at the location where
these Snoms are registering from? Are all the phones attempting
to register over port 5060 or are you setting them up to register
over unique ports to Asterisk? If you are setting them up to
register over specific ports, are they registering over those
ports according to 'asterisk show peers'? Also, is your asterisk
box local or hosted somewhere?<br>
<br>
Comparing IAX2 to SIP registrations is somewhat different: IAX2
tends to handle cutting through firewalls better though SIP is far
better supported by everyone.<br>
<br>
<br>
<br>
<br>
<div class="gmail_quote">On Fri, Oct 14, 2011 at 6:21 AM,
gincantalupo <span dir="ltr"><<a href="mailto:gincantalupo@fgasoftware.com" target="_blank">gincantalupo@fgasoftware.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;border-left:1px solid rgb(204, 204, 204);padding-left:1ex">Hi all,<br>
<br>
I'm stuck on a tricky problem.<br>
I set up an Asterisk 1.4.26.2 on a box with a bunch of Snom
Phones. When I call an IVR I get the damned one way voice
phenomena. It is not randomic, it happens all the time.<br>
I tried to upgrade the snom firmware to 7.3.30 but nothing
changed.<br>
If I call a phone I get a normal conversation and no problem
occurs if I (blind) transfer the call.<br>
If I use a IAX phone everything is fine.<br>
I think it is a SIP problem but I checked the sip files and
they seem ok.<br>
Tones seems to pass since the caller (me) can make a choice
from within the IVR menu.<br>
<br>
Sincerely, I haven't any idea left to try...<br>
<br>
Any hints?<br>
<br>
Thanks<br>
<br>
Giorgio<br>
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