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I don't think it's a snom specific issue as I have 5 branch offices all
with identical configurations and only one of the 5 is experiencing
this problem. I have checked and re-checked the config files until
they're practically burned into my brain and everything appears looks
OK. Any place there is a config option for the server address it
matches the server's physical address. The phones and all computers on
the network are statically assigned (these are very small 1-person or 2
at the most offices) so I'm absolutely sure that there is no chance of
an IP conflict. The phones and the Asterisk server are plugged in to
the same ethernet switch.<br>
<br>
I have tried tried using both sip debug and Wireshark to try to catch
anything relating to this, but the calls seem to happen very rarely
and, as usually happens when I'm trying to troubleshoot some sporadic
problem, the two people in that office can't ever reliably reproduce
this when I happen to be in the office or remotely logged in to the
server.<br>
<br>
To make things even more difficult, the Snom phones have a switch built
in to them and the PC's are in the pass-through port. This means that
any traffic generated by the phones is in the PC's blind spot. Anyway,
I've got about 1000 other projects going right now. Once I get a
couple of the more pressing ones knocked out I'll try to spend a couple
of days at the problematic office and see if I can't come up with some
way to resolve the problem.<br>
<br>
Thanks for the suggestions,<br>
Brent<br>
<br>
OCG Technical Support wrote:
<blockquote id="mid_011801c93634_af6c9d20_0e45d760__ca"
cite="mid:011801c93634$af6c9d20$0e45d760$@ca" type="cite">
<pre wrap="">Well, if this is snom specific I can't offer more insight. It really sounds
like misconfigured iptables and/or sip helper (conntrack/nat/etc).
Are you sure your IP address is right in your sip.conf? If you don;t have
NAT set to yes for these phones, they will trust the sip header for IP
address and may misroute.
-----Original Message-----
From: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Brent Davidson
Sent: October 24, 2008 7:36 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Sporadic One Way Audio
Importance: High
The asterisk server is connected to the PSTN via a Rhino R4FXO-EC card.
The lost RTP would have be between the Asterisk server and the phones.
There are only 2 phones in the building, 2 lines coming in to the
asterisk server and the server is on the same ethernet switch as the
phones. The phones are SIP phones. This is a simple PBX system that
picks up calls from the analog lines and routes them to the appropriate
phone, although it will eventually be linked to a larger system once all
the minor bugs are resolved.
OCG Technical Support wrote:
</pre>
<blockquote id="StationeryCiteGenerated_1" type="cite">
<pre wrap="">How is your asterisk server connected to the PSTN? SIP/IAX out...ISDN/T1
out? Etc...
Are you looking for lost RTP between * and internal phones or * and
</pre>
</blockquote>
<pre wrap=""><!---->external
</pre>
<blockquote id="StationeryCiteGenerated_2" type="cite">
<pre wrap="">provider?
-----Original Message-----
From: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>
[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Brent
</pre>
</blockquote>
<pre wrap=""><!---->Davidson
</pre>
<blockquote id="StationeryCiteGenerated_3" type="cite">
<pre wrap="">Sent: October 24, 2008 5:55 PM
To: Asterisk Users List
Subject: [asterisk-users] Sporadic One Way Audio
I'm having an unusual problem at one of my branch offices. Every now
and then they will make a call and the person they call is unable to
hear them, but they are able to hear the person. The Asterisk server
has only one ethernet interface and is on the same physical network as
the 2 snom 300 phones and is connected to the PSTN lines with a Rhino
R4FXO-EC card. Usually hanging up and calling back solves the problem,
but it is still aggravating to the customer that has been called.
Normally I'd suspect that something was only passing packets in one
direction, but there is no firewall between the asterisk server and the
phones and no iptables or anything like that running on the Asterisk
server and sifting through sip debug logs to try to find one call out of
maybe 50 has so far proven fruitless.
Are there any common issues that might cause this?
Thanks,
Brent Davidson
</pre>
</blockquote>
</blockquote>
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