[asterisk-users] one way voice with IVR

[Anton Kvashenkin]

Run tcpdump with portrange 10000-20000 (or what range of ports you use for
rtp) and dial test ivr to see what happening.

2011/10/17 gincantalupo <gincantalupo at fgasoftware.com>

> **
> 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> 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 | *Main:* (706) 995-0200
> *Direct:* (706) 995-0201 | *Mobile:* (706) 255-9203
>
> * *
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
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>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
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