[asterisk-users] one way voice with IVR

[Danny Nicholas]

Netstat -anp has been useful in finding this error for me in the past.   A
"normal" Asterisk call will have 2 or 4 udp connections to carry traffic
to/from phone to pbx.  On a one-way call, this will be an "odd" count.  Then
you can check your rtp.conf and firewall and see how the channel got
blocked.

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of John Knight
Sent: Friday, October 14, 2011 8:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] one way voice with IVR

 

Hi Giorgio,

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.

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?

Comparing IAX2 to SIP registrations is somewhat different:  IAX2 tends to
handle cutting through firewalls better though SIP is far better supported
by everyone.





On Fri, Oct 14, 2011 at 6:21 AM, gincantalupo <gincantalupo at fgasoftware.com>
wrote:

Hi all,

I'm stuck on a tricky problem.
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.
I tried to upgrade the snom firmware to 7.3.30 but nothing changed.
If I call a phone I get a normal conversation and no problem occurs if I
(blind) transfer the call.
If I use a IAX phone everything is fine.
I think it is a SIP problem but I checked the sip files and they seem ok.
Tones seems to pass since the caller (me) can make a choice from within the
IVR menu.

Sincerely, I haven't any idea left to try...

Any hints?

Thanks

Giorgio


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