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 >>> >>
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