Re: [asterisk-users] Weird one way Audio situation
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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird one way Audio situation
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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weird one way Audio situation
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-143.txt.gz Description: GNU Zip compressed data ___ -- 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
Re: [asterisk-users] Weird one way Audio situation
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 ___ -- 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
Re: [asterisk-users] Weird one way Audio situation
Hi Steve, thanks for your response... I will try it this saturday and I'll let you know... Best regards On Wed, Jun 11, 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 -- Nacho Linux Counter #156439 ___ -- 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
Re: [asterisk-users] Weird one way Audio situation
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
[asterisk-users] Weird one way Audio situation
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 ___ -- 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
