[asterisk-users] Weird one way Audio situation

[Raúl Gómez C.]

Well, I think I've solved the problem, just to let you know, I've just added
the Answer() app before the Call(Zap/N) app. Thanks a lot to Yannick Lam
Hang of Sangoma Technologies for suggesting that!!!

On Wed, Jun 25, 2008 at 9:04 PM, Raúl Gómez C. <nachogomez at gmail.com> wrote:

> 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
>>>>
>>>


-- 
Nacho
Linux Counter #156439
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