[asterisk-users] Weird one way Audio situation

[Raúl Gómez C.]

Well, I have new information if anyone can/want to help me...

(Please read all the previous messages in this email)

If I call a number that can't hear me at all (calling from inside my network
using a Grandstream GXP-2000 phone through Asterisk) and then I put this
call on hold for a second and then I take again the call, then the callee
start hearing me, :s

Any ideas???

Thanks in advance...


-- 
Nacho
Linux Counter #156439


On Tue, Jun 17, 2008 at 7:50 PM, Raúl Gómez C. <nachogomez at gmail.com> wrote:

> I've been playing around in order to find something new and I've found
> this:
>
> I have created an IVR for test purposes, then I've placed a call from my
> sip phone using one of my telco lines to another of my telco lines attached
> to the PBX, in this situation I'm using two FXO channels, one for the
> outgoing call and another for the incoming call.
>
> Then I have created an extension in this IVR in order to make an echo test
> and I've used MixMonitor() to record the audio of the test. When I dial this
> extension I never can hear my echoed voice, but when I listen to the
> recording the audio have a lot of artifacts and the busy and dial tone are
> almost inaudible, the same effect that happens when you play to almost
> identical audio files, so I can presume that it is the same audio wave but
> out of phase (meaning the echo is working, I think).
>
> I don't know if this can be happening because of the Hardware Echo Canceler
> on my Remora A400D.
>
> If I call the extension of the echo test directly from my SIP phone without
> using any telco line (SIP <--> IP <--> Asterisk) then the test works just
> fine.
>
> Another test I've made is, during a call with the one way audio problem, I
> have used the ZapBarge() application to hear what's happening on the Zap
> Channel (from another SIP phone on my network). In this case I heard the
> callee complaining that he/she can't hear anything and I can't hear the
> caller (which is on the same network of my phone). In this case the caller
> can hear the callee.
>
> I have grabbed the sip debug messages of this call from the asterisk CLI
> and is attached (compressed) to this email.
>
>
> Well, thanks again for any comment/response...
>
>
> --
> Nacho
> Linux Counter #156439
>
>
>
> On Tue, Jun 17, 2008 at 5:14 PM, Raúl Gómez C. <nachogomez at gmail.com>
> wrote:
>
>> Hi Steve and the rest of the list,
>>
>> On Thu, Jun 12, 2008 at 7:11 AM, Steve Totaro <
>> stotaro at totarotechnologies.com> wrote:
>>
>>> Is your Asterisk box dual homed?  Firewalled?  Any output from the CLI
>>> with verbose turned on, that might help?  Turn on SIP debugging as
>>> well.
>>>
>>> Thanks,
>>> Steve T
>>>
>>>
>> My Asterisk Server has two NIC with a channel bonding setup (Balance TLB)
>> connected to the same switch, and it does not have any firewall rule.
>>
>>
>> I'm attaching a file with the output of "sip set debug" on the CLI of a
>> call in this situation.
>>
>> Although calls made with SIP phones have this strange behavior, when I
>> place a call with an analog phone connected to a FXS port of the same TDM
>> card (see below for full description) this does not happen.
>>
>>
>> Thanks, any help will be really appreciated...
>>
>>
>>
>> --
>> Nacho
>> Linux Counter #156439
>>
>>
>>
>> On Thu, Jun 12, 2008 at 7:11 AM, Steve Totaro <
>> stotaro at totarotechnologies.com> wrote:
>>
>>> On Tue, Jun 10, 2008 at 1:40 PM, Raúl Gómez C. <nachogomez at gmail.com>
>>> wrote:
>>> > Hi list,
>>> >
>>> > I'm having trouble with calls placed to the PSTN (through a TDM card),
>>> > sometimes (a lot indeed) when I dial a number the callee party can't
>>> hear me
>>> > at all.
>>> >
>>> > My setup is:
>>> >
>>> > Asterisk 1.4.20.1
>>> > Zaptel 1.4.11
>>> > libpri 1.4.4
>>> > Wanpipe 3.2.4
>>> >
>>> > I have a Sangoma Remora Card A400D (2 FXS / 10 FXO), Grandstream
>>> GXP-2000 IP
>>> > Phones, SuSE Linux Enterprise Server 10 (SP2) x86_64 with Kernel
>>> > 2.4.16.60-0.23-smp
>>> >
>>> > I'm using the ulaw audio codec.
>>> >
>>> > There is no NAT between the Asterisk Server and the Phones (the phone
>>> and
>>> > the server are in the same network segment).
>>> >
>>> > What can it be???
>>> >
>>> > Thanks in advance for any help/comment...
>>> >
>>> >
>>> > --
>>> > Raul
>>> > Linux Counter #156439
>>>
>>> Is your Asterisk box dual homed?  Firewalled?  Any output from the CLI
>>> with verbose turned on, that might help?  Turn on SIP debugging as
>>> well.
>>>
>>> Thanks,
>>> Steve T
>>>
>>
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