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 >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > >
SIP-Debug-141.txt.gz
Description: GNU Zip compressed data
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