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. <[EMAIL PROTECTED]> 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. <[EMAIL PROTECTED]> > 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. <[EMAIL PROTECTED]> >> wrote: >> >>> Hi Steve and the rest of the list, >>> >>> On Thu, Jun 12, 2008 at 7:11 AM, Steve Totaro < >>> [EMAIL PROTECTED]> 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 < >>> [EMAIL PROTECTED]> wrote: >>> >>>> On Tue, Jun 10, 2008 at 1:40 PM, Raúl Gómez C. <[EMAIL PROTECTED]> >>>> 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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