[asterisk-users] one way voice with IVR

[gincantalupo]

Hi John,

there is no firewall:

snom <--> pbx <--> patton <--> pstn

It happens ONLY with IVRs. Normal calls are fine. How can it be?

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???

Giorgio


On 10/14/2011 03:48 PM, John Knight wrote:
> 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 <mailto: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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> -- 
> *
> *
>
> *John Knight*
> Classic City Telco LLC
> *Email:* john at classiccitytelco.com <mailto:john at classiccitytelco.com> 
> | *Main:* (706) 995-0200
> *Direct:* (706) 995-0201 | *Mobile:* (706) 255-9203
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>
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