[asterisk-users] strange sound on conference call

Hi. I am having a problem with a conference call on my server which a vps in the cloud. I am using chan_sip and meetme. What I get is a bit of a staticy or robotic sound, but it goes away if the user lowers the volume a bit which we can do with *4 in meetme. So, is the problem with the

Re: [asterisk-users] Asterisk sound file cache expiration

Hi, I found https://markmail.org/message/xh5sbqvsgwywrjje#query:+page:1+mid:vbzl4hup6jawrzup+state:results and it seems to have answered my question. By adding an expires value to the response Asterisk seems to cache it for the time specified. On Sun, Sep 5, 2021 at 6:23 AM Dovid Bender wrote:

[asterisk-users] Asterisk sound file cache expiration

Hi, Is there any way to set the default expiration for the media cache? After looking at the sqlite3 db it seems that asterisk by default sets the expiration to the time that the file was accessed so the file is never cached locally and everytime the file is played, it's downloaded again. --

[asterisk-users] play sound without stop it when dtmf is pressed

Hello, I would like to known if there is a way to use background without stopping playing the sound file until it's end and capture dtmf. i try : background(soundfile) waitexten(10) but background exit immediatly when any dtmf is press and continue to waitexten I want the full sound file play

Re: [asterisk-users] No sound with internal calls depending on which phones

Hi Denis That advice is correct for disabling RTP support in the phone and if you have achieved this then your quoted error about SRTP in the Asterisk console (when the call is failing) should no longer be appearing. This will help show that it was a red herring all along. The next step (IMO)

Re: [asterisk-users] No sound with internal calls depending on which phones

You might have to disable srtp negotiations inside the phone web ui options. Mitul On Nov 12, 2015 8:53 PM, "(lists) Denis BUCHER" wrote: > Dear all, > > I have a very strange problem : > >- external calls work perfectly, >- internal calls between some phones

Re: [asterisk-users] No sound with internal calls depending on which phones

Am 12.11.2015 um 16:22 schrieb (lists) Denis BUCHER: Dear all, I have a very strange problem : * external calls work perfectly, * internal calls between some phones too, * but internal call between two similar phones don't work !!! (Snom 710) When we have sound, there are no errors in

[asterisk-users] No sound with internal calls depending on which phones

Dear all, I have a very strange problem : * external calls work perfectly, * internal calls between some phones too, * but internal call between two similar phones don't work !!! (Snom 710) When we have sound, there are no errors in asterisk. When we do not have sound, there is the

Re: [asterisk-users] No sound with internal calls depending on which phones

ely,* cid:image001.jpg@01D0D5C4.27A0CBA0 *Sam Basan* cid:image003.png@01C918DA.6B3E4530 *From:*Mitul Limbani [mailto:mi...@enterux.in] *Sent:* Thursday, November 12, 2015 5:25 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> *Subject:* Re:

Re: [asterisk-users] No sound with internal calls depending on which phones

pg@01D0D5C4.27A0CBA0] > > *Sam Basan* > > [image: cid:image003.png@01C918DA.6B3E4530] > > > > *From:* Mitul Limbani [mailto:mi...@enterux.in <mi...@enterux.in>] > *Sent:* Thursday, November 12, 2015 5:25 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discus

Re: [asterisk-users] No sound with internal calls depending on which phones

Discussion <asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] No sound with internal calls depending on which phones You might have to disable srtp negotiations inside the phone web ui options. Mitul On Nov 12, 2015 8:53 PM, "(lists) Denis BUCHER" <dbuc

Re: [asterisk-users] Recording sound.

Anurag Rana wrote: Hi All, Kia ora, I am calling mobile numbers from Soft-phone and recording the call. In recording the level of sound from the receiver's side is perfect (loud enough) but my voice's sound level is very weak. I barely can hear it. During the call receiver is able to hear

[asterisk-users] Recording sound.

Hi All, I am calling mobile numbers from Soft-phone and recording the call. In recording the level of sound from the receiver's side is perfect (loud enough) but my voice's sound level is very weak. I barely can hear it. During the call receiver is able to hear me. But in recording my part of

Re: [asterisk-users] Pass Sound files as Argument to Macro Asterisk 1.8

It should be. I'd write something like the below: [macro-test] exten = s,1,NoOp exten = s,n,GotoIf($[${STAT(e,/var/lib/asterisk/sounds/${ARG1}.ulaw)} = 0]?NOPROMPT:PLAYBACK) exten = s,n(NOPROMPT),Background(nothing-recordedforpm-prompt-number) exten = s,n,SayPhonetic(${ARG1}) exten =

[asterisk-users] Pass Sound files as Argument to Macro Asterisk 1.8

Hi All, I was wondering if it is possible to pass sound files to a macro as an argument in Asterisk 1.8? Thanks! Regards, Daniel van den Berg SureTel South Africa 087-944-7873 -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] VoIP sound quality : highroad sound

2013-11-12 17:42, Jonas Kellens skrev: X.X.X.100 2f9a96ec3e1 00:00:42 000138 049741 (99.72%) 0. 000136 00 ( 0.00%) 0.0031 X.X.X.70 68289fc05ff 00:02:27 007318 060143 (89.15%) 0. 007301 00 ( 0.00%) 0.0001 A lot of packetloss for

Re: [asterisk-users] VoIP sound quality : highroad sound

On 11/13/2013 11:48 AM, Johan Wilfer wrote: 2013-11-12 17:42, Jonas Kellens skrev: X.X.X.100 2f9a96ec3e1 00:00:42 000138 049741 (99.72%) 0. 000136 00 ( 0.00%) 0.0031 X.X.X.70 68289fc05ff 00:02:27 007318 060143 (89.15%) 0. 007301 00 (

Re: [asterisk-users] VoIP sound quality : highroad sound

I frequently use Audacity to analyze the audio data. In many cases I can see from the spectra (and other graphical representations) with what kind of problem I am dealing. Meanwhile, for most of my problems I no longer depend on an audio editor. I don't know whether this is helpful in your

Re: [asterisk-users] VoIP sound quality : highroad sound

2013-11-13 11:55, Jonas Kellens skrev: On 11/13/2013 11:48 AM, Johan Wilfer wrote: 2013-11-12 17:42, Jonas Kellens skrev: X.X.X.100 2f9a96ec3e1 00:00:42 000138 049741 (99.72%) 0. 000136 00 ( 0.00%) 0.0031 X.X.X.70 68289fc05ff 00:02:27 007318 060143

[asterisk-users] VoIP sound quality : highroad sound

Hello, what could be causing the issue of poor sound quality ? Some calls, certainly not all of them, sound like if the caller is standing next to a very busy road with lots of cars passing. To be clear : the person calling is not standing next to a highway. But there seems to be a noise on

Re: [asterisk-users] VoIP sound quality : highroad sound

On 11/12/2013 04:29 PM, jg wrote: Did you have a look at the codecs that are involved? There are about 40 à 45 simultaneous calls (using G711a). Jonas. -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] VoIP sound quality : highroad sound

Did you have a look at the codecs that are involved? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

Re: [asterisk-users] VoIP sound quality : highroad sound

Current situation : sip1*CLI sip show channelstats Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter X.X.X.133 4d7b0a7f337 00:05:59 000243 00 ( 0.00%) 0. 000576 046854 (8134.38%) 0.0002 X.X.X.42

Re: [asterisk-users] VoIP sound quality : highroad sound

Yes, all SIP. Current situation : sip1*CLI sip show channelstats Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter X.X.X.133 4d7b0a7f337 00:05:59 000243 00 ( 0.00%) 0. 000576 046854 (8134.38%)

[asterisk-users] Extra Sound Packages

Maybe this is a stupid question. Are the files in Extra Sound Packages related to any product or are they just supplemental material? I searched the source files for some of the file names and didn't find any reference. jg --

[asterisk-users] fax - sound/tone - dealing with SPAM

I receive several calls from this scamer: Senior SafeAlert It is an automated call and they keep rotating their caller ID so it is harder to block them. Does asterisk have a fax sound tone? If I block their number and play fax tone/sound maybe they will remove me from their calling list. I've

Re: [asterisk-users] fax - sound/tone - dealing with SPAM

Why dont u run a reverse dialer on the admin contacts phone number. Leave him clueless as well. Mitul On Apr 5, 2013 1:25 AM, Joseph syscon...@gmail.com wrote: I receive several calls from this scamer: Senior SafeAlert It is an automated call and they keep rotating their caller ID so it is

Re: [asterisk-users] fax - sound/tone - dealing with SPAM

On 04/05/13 01:29, Mitul Limbani wrote: Why dont u run a reverse dialer on the admin contacts phone number. Leave him clueless as well. Mitul Reverse dialer on 7044972383 ? What is going to do? -- Joseph On Apr 5, 2013 1:25 AM, Joseph [1]syscon...@gmail.com wrote: I

Re: [asterisk-users] fax - sound/tone - dealing with SPAM

It is an automated call and they keep rotating their caller ID so it is harder to block them. Automate it. We have an extension that the operators forward calls to that add the number to the black list database. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to

Re: [asterisk-users] fax - sound/tone - dealing with SPAM

On 04/04/13 16:21, Doug Lytle wrote: It is an automated call and they keep rotating their caller ID so it is harder to block them. Automate it. We have an extension that the operators forward calls to that add the number to the black list database. Doug Can you please share more details,

Re: [asterisk-users] fax - sound/tone - dealing with SPAM

Can you please share more details, is it done via dial plan? I'll post a portion of my dial plan when I get home from work. It's currently for Asterisk 1.4 using the mysql command, I'm in the process of moving it to Asterisk 11 and func_odbc. Doug -- Ben Franklin quote: Those who would

Re: [asterisk-users] fax - sound/tone - dealing with SPAM

On Thu, 4 Apr 2013, Joseph wrote: I receive several calls from this scamer:... Does asterisk have a fax sound tone? If I block their number and play fax tone/sound maybe they will remove me from their calling list. The fax CNG may work. How about SIT?

Re: [asterisk-users] fax - sound/tone - dealing with SPAM

On 04/04/2013 09:54 PM, Joseph wrote: +1.7044972383 If that number is his actual number, maybe create a script that calls him 10 times an hour, every hour between 00:00 - 07:00am and plays screaming monkeys every time he picks up (or his voicemail kicks in). Regards, Patrick --

Re: [asterisk-users] fax - sound/tone - dealing with SPAM

On 04/05/13 00:58, Patrick Lists wrote: On 04/04/2013 09:54 PM, Joseph wrote: +1.7044972383 If that number is his actual number, maybe create a script that calls him 10 times an hour, every hour between 00:00 - 07:00am and plays screaming monkeys every time he picks up (or his voicemail kicks

Re: [asterisk-users] fax - sound/tone - dealing with SPAM

Joseph wrote: Can you please share more details, is it done via dial plan? We use extension 2000. You'll need to added the 'check_blacklist' sub-routine to the inbound parts of your dial plan. [tele_torture] exten = 2000,1,GotoIf($[${CALLERID(number)} = 0]?7:2) exten =

[asterisk-users] clicking sound with alaw codec

I'm trying to interface Asterisk with an Alcatel PABX and trying to find a code that works well. It says it doesn't support ulaw, though it doesn't reject it. It supports G.729, and that works fine, but we'd prefer not to use compression. When I use alaw, the path from Asterisk to the Alcatel

Re: [asterisk-users] clicking sound with alaw codec

Of Richard Kenner Sent: Thursday, January 24, 2013 2:45 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] clicking sound with alaw codec I'm trying to interface Asterisk with an Alcatel PABX and trying to find a code that works well. It says it doesn't support ulaw, though it doesn't reject

Re: [asterisk-users] clicking sound with alaw codec

Your sounds might be too loud. We use a lot of custom sounds here and when the volume approaches 0 db (asterisk standard is -3 db) we get fuzz and clicks. Sorry I wasn't clear. This is *always*. I hear it over the call when there's talking and when there's dead silence (e.g., an empty

Re: [asterisk-users] clicking sound with alaw codec

On 01/24/2013 09:44 PM, Richard Kenner wrote: [snip] When I use alaw, the path from Asterisk to the Alcatel is completely clean, but the other way has a set of clicks that kind of sound like old-fashioned audio noise. [snip] It's been ages since I experienced that but things to check that come

Re: [asterisk-users] clicking sound with alaw codec

When I use alaw, the path from Asterisk to the Alcatel is completely clean, but the other way has a set of clicks that kind of sound like old-fashioned audio noise. [snip] It's been ages since I experienced that but things to check that come to mind in no particular order are:

Re: [asterisk-users] clicking sound with alaw codec

- jitterbuffer settings (try on/off) I added jbenable=yes and get lots of: [Jan 24 17:53:41] WARNING[12317]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-6c7 received frame with invalid timing info: has_timing_info=1, len=0, ts=371371424, src=RTP [Jan 24 17:53:41] WARNING[12317]:

Re: [asterisk-users] clicking sound with alaw codec

On 01/24/2013 11:57 PM, Richard Kenner wrote: - jitterbuffer settings (try on/off) I added jbenable=yes and get lots of: [Jan 24 17:53:41] WARNING[12317]: abstract_jb.c:284 ast_jb_put: DAHDI/i1/2128518396-6c7 received frame with invalid timing info: has_timing_info=1, len=0,

Re: [asterisk-users] clicking sound with alaw codec

Check https://issues.asterisk.org/jira/browse/ASTERISK-12042 I did. But that was with an unofficial G.729. This is with the supplied alaw codec. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New

Re: [asterisk-users] play sound file

-users@lists.digium.com Subject: Re: [asterisk-users] play sound file 2012-01-26 10:11, Eyal skrev: Thanks But this is not what I am looking for, in this way I can start the sound file from some point in the file but the callers must hear the file until the end. I need something

Re: [asterisk-users] play sound file

You can use controlplayback On Jan 25, 2012 9:00 PM, Eyal e...@mcr-m.com wrote: Hi, How can I play a sound file from the middle and end it after a certain number of seconds? -- _ -- Bandwidth and Colocation

Re: [asterisk-users] play sound file

check this http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback Nasir Iqbal ICTBroadcast SMS, Fax and Voice broadcasting solution http://www.ictbroadcast.com/ On Wed, Jan 25, 2012 at 8:29 PM, Eyal e...@mcr-m.com wrote: Hi, How can I play a sound file from the middle and end

Re: [asterisk-users] play sound file

Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] play sound file check this http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback Nasir Iqbal ICTBroadcast SMS, Fax and Voice broadcasting solution http://www.ictbroadcast.com/ On Wed, Jan 25, 2012

Re: [asterisk-users] play sound file

-users] play sound file ** ** check this http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback Nasir Iqbal ICTBroadcast SMS, Fax and Voice broadcasting solution http://www.ictbroadcast.com/ On Wed, Jan 25, 2012 at 8:29 PM, Eyal e...@mcr-m.com

Re: [asterisk-users] play sound file

...@lists.digium.com] *On Behalf Of *Nasir Iqbal *Sent:* Thursday, January 26, 2012 10:53 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] play sound file check this http://www.voip-info.org/wiki/view/Asterisk+cmd+ControlPlayback Nasir

[asterisk-users] play sound file

Hi, How can I play a sound file from the middle and end it after a certain number of seconds? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

[asterisk-users] maximizing sound quality in 10.0

Hi list, I have a set of 300 or so WAV files I was combining and playing using playback/background in 1.4.X. Now that I have moved on to the 10.0 set, I understand that I can replace my 8 Khz mono files with virtually unlimited Khz mono files (still no stereo, but a quantum leap

Re: [asterisk-users] maximizing sound quality in 10.0

On Ter, 2011-12-27 at 13:34 -0600, Danny Nicholas wrote: Hi list, I have a set of 300 or so WAV files I was combining and playing using playback/background in 1.4.X. Now that I have moved on to the 10.0 set, I understand that I can replace my 8 Khz mono files with virtually

[asterisk-users] no sound with ICES ?

Hi, I'm trying to have Asterisk pick up a call and stream it to Liquidsoap (Icecast2 compatible). This is what I have in my extensions.conf : [default] exten = s,1,Answer exten = s,n,Wait(1) exten = s,n,Ices(/etc/asterisk/asterisk-ices.xml) exten = s,n,HangUp Here's what working so far:

[asterisk-users] IVR sound after dial sip

Hi, I have a ivr, and I need to make a beep sound playback after phone when to dial sip DIALSTATUS} = $ {ANSWER example 1234,1,Answer() 1234,n,Dial(SIP/1234) ;When 1234 sip phone answer te call, playback beep on this sip phone. how could I do this? thanks for any help --

[asterisk-users] Alarms Sound files

Dear All Can anyone let me know where i can free sound file whcih i can use for system monitoring alrams. Regards Amit-- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a

Re: [asterisk-users] Retrieving sound files from DB as opposed to filesystem

On Tue, May 03, 2011 at 01:09:14AM -0400, A E [Gmail] wrote: On Mon, May 2, 2011 at 9:45 PM, C F shma...@gmail.com wrote: Just from my experience with different DBs, stay away from BLOB data types as much as possible. Hi CF, any particular reason why? I've had a good experience with

Re: [asterisk-users] Retrieving sound files from DB as opposed to filesystem

On Tue, May 3, 2011 at 1:09 AM, A E [Gmail] all.efor...@gmail.com wrote: On Mon, May 2, 2011 at 9:45 PM, C F shma...@gmail.com wrote: Just from my experience with different DBs, stay away from BLOB data types as much as possible. Hi CF, any particular reason why? I've had a good experience

[asterisk-users] Retrieving sound files from DB as opposed to filesystem

Hello All, Probably a silly question, but we're wondering if people have had any experience and have data to demonstrate if the performance of the Asterisk system might suffer in terms of latency etc. if we're to have it retrieve sound files from a database using odbc as opposed to storing them

Re: [asterisk-users] Retrieving sound files from DB as opposed to filesystem

Just from my experience with different DBs, stay away from BLOB data types as much as possible. On Mon, May 2, 2011 at 3:15 AM, A E [Gmail] all.efor...@gmail.com wrote: Hello All, Probably a silly question, but we're wondering if people have had any experience and have data to demonstrate if

Re: [asterisk-users] Retrieving sound files from DB as opposed to filesystem

On Mon, May 2, 2011 at 9:45 PM, C F shma...@gmail.com wrote: Just from my experience with different DBs, stay away from BLOB data types as much as possible. Hi CF, any particular reason why? I've had a good experience with it, in fact that's recommended by DB developers when it's a case of

[asterisk-users] siren sound

Hi All, I have an strange behaviour, sometime (so far I am not sure how to reproduce the problem) when I call to a meetme room, the system asks me for the pin and after that what I can hear is a sound like an ambulance siren. After restarting the asterisk process everthing works again. The

Re: [asterisk-users] iax2 sound problem

Mailing List - Non-Commercial Discussion Subject: [asterisk-users] iax2 sound problem Hello, I installed 1.6.2.17 version of asterisk. Set the user database to realtime. I have no problems with sip users. They can register talk etc.. With iax clients, they can register also.. And when

[asterisk-users] iax2 sound problem

Hello, I installed 1.6.2.17 version of asterisk. Set the user database to realtime. I have no problems with sip users. They can register talk etc.. With iax clients, they can register also.. And when they call iax to sip, it works. When they make an echo test..no voice received on iax clients.

Re: [asterisk-users] scratchy sound on TE410P

On 11/11/10 11:06 PM, Carlos Chavez wrote: On Thu, 11 Nov 2010 20:08:09 -0600, Russ Meyerriecks wrote On 11/11/10 7:23 PM, Carlos Chavez wrote: I seem to be having the same problem with a new server. I am using a TE220 with a VPM450 module. Using Dahdi 2.4.0 and Asterisk 1.6.2.13 on a

Re: [asterisk-users] scratchy sound on TE410P

On Thu, 11 Nov 2010, Russ Meyerriecks wrote: Yes, this is a snapshot after about 24 hours since I cleared the counters. I see what you mean - how can I have 76 seconds of errors but no bumped error counters. I ran again just now: r...@vigw3:/etc/asterisk# dahdi_maint -s 1 Span 1: FEC :

Re: [asterisk-users] scratchy sound on TE410P

On 11/12/10 10:44 AM, Jeff LaCoursiere wrote: On Thu, 11 Nov 2010, Russ Meyerriecks wrote: Yes, this is a snapshot after about 24 hours since I cleared the counters. I see what you mean - how can I have 76 seconds of errors but no bumped error counters. I ran again just now:

Re: [asterisk-users] scratchy sound on TE410P

On Tue, 9 Nov 2010, Daniel Tryba wrote: I am curious about the tool dahdi_maint... what do the various acronyms stand for? Yea there seemed to be a bit of confusion here as well so I patched trunk with some more descriptive error counter labels :O) FEC : 0: Framing Errors CEC : 0: CRC

Re: [asterisk-users] scratchy sound on TE410P

On Thu, 11 Nov 2010, Russ Meyerriecks wrote: On Tue, 9 Nov 2010, Daniel Tryba wrote: I am curious about the tool dahdi_maint... what do the various acronyms stand for? Yea there seemed to be a bit of confusion here as well so I patched trunk with some more descriptive error counter labels

Re: [asterisk-users] scratchy sound on TE410P

On 11/11/10 5:44 PM, Jeff LaCoursiere wrote: On Thu, 11 Nov 2010, Russ Meyerriecks wrote: On Tue, 9 Nov 2010, Daniel Tryba wrote: I am curious about the tool dahdi_maint... what do the various acronyms stand for? Yea there seemed to be a bit of confusion here as well so I patched trunk

Re: [asterisk-users] scratchy sound on TE410P

On Thu, 2010-11-11 at 18:28 -0600, Russ Meyerriecks wrote: On 11/11/10 5:44 PM, Jeff LaCoursiere wrote: On Thu, 11 Nov 2010, Russ Meyerriecks wrote: On Tue, 9 Nov 2010, Daniel Tryba wrote: I am curious about the tool dahdi_maint... what do the various acronyms stand for? Yea

Re: [asterisk-users] scratchy sound on TE410P

On 11/11/10 7:23 PM, Carlos Chavez wrote: I seem to be having the same problem with a new server. I am using a TE220 with a VPM450 module. Using Dahdi 2.4.0 and Asterisk 1.6.2.13 on a Dell server. All calls to the outside have bad voice quality (echo and distortion). Internal calls

Re: [asterisk-users] scratchy sound on TE410P

On Thu, 11 Nov 2010 20:08:09 -0600, Russ Meyerriecks wrote On 11/11/10 7:23 PM, Carlos Chavez wrote: I seem to be having the same problem with a new server. I am using a TE220 with a VPM450 module. Using Dahdi 2.4.0 and Asterisk 1.6.2.13 on a Dell server. All calls to the outside

Re: [asterisk-users] scratchy sound on TE410P

On Mon, Nov 08, 2010 at 02:44:26PM -0500, Jeff LaCoursiere wrote: It could be the echo canceller, I had this kind of problem with OSLEC. I also thought the PRI provider was sending clipped audio. I switched to the VPM450 daughterboard and since audio has been crystal clear. What is your setup

Re: [asterisk-users] scratchy sound on TE410P

On Sun, Nov 07, 2010 at 10:26:59AM -0500, Jeff LaCoursiere wrote: asterisk 1.4.35 dahdi 2.3.0.1+2.3.0 one span on a 4port T1 card Got complaints this morning that outbound and inbound calls were scratchy and I made a few test calls. It kind of sounds like the gain is too high somewhere,

Re: [asterisk-users] scratchy sound on TE410P

On Mon, 8 Nov 2010, Daniel Tryba wrote: On Sun, Nov 07, 2010 at 10:26:59AM -0500, Jeff LaCoursiere wrote: asterisk 1.4.35 dahdi 2.3.0.1+2.3.0 one span on a 4port T1 card Got complaints this morning that outbound and inbound calls were scratchy and I made a few test calls. It kind of

Re: [asterisk-users] scratchy sound on TE410P

On Mon, Nov 8, 2010 at 1:44 PM, Jeff LaCoursiere j...@sunfone.com wrote: I inherited this board, and don't think it has the echo canceller daughterboard. Is there a way to query for it without taking the machine down? It is loading MG2 otherwise. 'dmesg | grep VPM' should tell you if you

[asterisk-users] scratchy sound on TE410P

asterisk 1.4.35 dahdi 2.3.0.1+2.3.0 one span on a 4port T1 card Got complaints this morning that outbound and inbound calls were scratchy and I made a few test calls. It kind of sounds like the gain is too high somewhere, and the audio is overdriven. Is this a problem at the carrier? I'm

Re: [asterisk-users] scratchy sound on TE410P

On 11/7/10 9:26 AM, Jeff LaCoursiere wrote: asterisk 1.4.35 dahdi 2.3.0.1+2.3.0 one span on a 4port T1 card Got complaints this morning that outbound and inbound calls were scratchy and I made a few test calls. It kind of sounds like the gain is too high somewhere, and the audio is

Re: [asterisk-users] no sound between extensions

Also check the codecs as if you are using g729 or g723, there is a chance that they are not available in codecs directory ( /usr/lib/asterisk/modules). -THQ- !!!ONE Date: Tue, 1 Jun 2010 19:24:41 -0400 From: zisha...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users

Re: [asterisk-users] no sound between extensions

I have remote access to the server so I checked the canreinvite .. they are all set to no. I can't try the call from here, I will get back to you. Gary Baribault On 06/01/2010 07:24 PM, Zeeshan Zakaria wrote: Do you agree something is blocking the audio in one direction? Can you do a 'rtp

Re: [asterisk-users] no sound between extensions

directory ( /usr/lib/asterisk/modules). *-THQ- !!!ONE* Date: Tue, 1 Jun 2010 19:24:41 -0400 From: zisha...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] no sound between extensions Do

Re: [asterisk-users] no sound between extensions

-0400 From: zisha...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] no sound between extensions Do you agree something is blocking the audio in one direction? Can you do a 'rtp debug' and then initiate a SIP call and see if there is two way audio traffic. Also make

[asterisk-users] no sound between extensions

Hello all, I have Asterisk 1.4.26 installed on an OpenSuSE 11.2 server with a Digium 8 port FXO card. The local network is 100Mbps Ethernet and my phones are Linksys SPA-921 or Linksys Analog adaptors. The phones are setup with DHCP, and are on the same flat non-routed network. There is no

Re: [asterisk-users] no sound between extensions

Incoming and outgoing calls are on SIP or on ZAP? Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-06-01 3:28 PM, Gary Baribault g...@baribault.net wrote: Hello all, I have Asterisk 1.4.26 installed on an OpenSuSE 11.2 server with a Digium 8 port FXO card. The local

Re: [asterisk-users] no sound between extensions

Baribault Sent: Tuesday, June 01, 2010 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] no sound between extensions Hello all, I have Asterisk 1.4.26 installed on an OpenSuSE 11.2 server with a Digium 8 port FXO card. The local network is 100Mbps

Re: [asterisk-users] no sound between extensions

Output of 'iptables -L -n' would also be helpful. I am sure its a NAT issue if incoming and ougoing calls are on ZAP channels. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-06-01 3:53 PM, Danny Nicholas da...@debsinc.com wrote: My assumption is that inbound/outbound

Re: [asterisk-users] no sound between extensions

Incomming calls are on TDM lines connected to the Digium card. Calls between extentions are on the LAN for SIP registered users/ip phones. Gary Baribault On 06/01/2010 03:32 PM, Zeeshan Zakaria wrote: Incoming and outgoing calls are on SIP or on ZAP? Zeeshan A Zakaria -- Sent from my

Re: [asterisk-users] no sound between extensions

Subject: [asterisk-users] no sound between extensions Hello all, I have Asterisk 1.4.26 installed on an OpenSuSE 11.2 server with a Digium 8 port FXO card. The local network is 100Mbps Ethernet and my phones are Linksys SPA-921 or Linksys Analog adaptors. The phones are setup with DHCP

Re: [asterisk-users] no sound between extensions

As I stated, the incoming calls are on TDM DS0s connected to the Digium card, and the extensions are on the same local network as the Asterisk server. There is currently no NAT anywhere. Gary Baribault On 06/01/2010 05:22 PM, Zeeshan Zakaria wrote: Output of 'iptables -L -n' would also be

Re: [asterisk-users] no sound between extensions

Do you agree something is blocking the audio in one direction? Can you do a 'rtp debug' and then initiate a SIP call and see if there is two way audio traffic. Also make sure these extensions have 'canreinvite=no'. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-06-01

[asterisk-users] scratchy sound

Hi, I'm experiencing a few (but meaningful) cases of audio distortion (or bad quality). I can't say yet how often this happens. Please listen to the following sound file: http://213.96.91.201/temp/distorted_audio_1.wav This was recorded by Asterisk while the local SIP caller was dialing out a

Re: [asterisk-users] scratchy sound

Hi Vieri, The sound I hear does not seem caused by packet loss, jitter or latency, this problems usually produces a robotic or synthetic voice. It seems produced by some kind of bad contact (most probable). It is strange that you are seeing it using hard phones, I could bet on the headphones.

Re: [asterisk-users] scratchy sound

Am 09.04.2010 13:10, schrieb Vieri: Please listen to the following sound file: I've experienced similar (well, vaguely similar) distortion on a horstbox pro when echo cancellation is switched on for the zap channels (ISDN). Turning it off resulted in no distortion at all, but then i

Re: [asterisk-users] scratchy sound

Hi, sounds for me like when i use an headset and the microfone handle scratches on my beard while i talk ;) maybe you have a network cable whitout screening. I had bad problems on different phones which sounds like that you have cause of electric or magnetic inteferences but when i changed

Re: [asterisk-users] Cache sound files for faster processing

@sedwards.com wrote: From: Steve Edwards asterisk@sedwards.com Subject: Re: [asterisk-users] Cache sound files for faster processing To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tuesday, April 6, 2010, 7:15 AM Are there any way

Re: [asterisk-users] Cache sound files for faster processing

Are there any way of configuring of Asterisk so it'll cache sound files in memory, and when Asterisk receive a call, instead of loading sound files from the disk On Mon, 5 Apr 2010, Luki wrote: Not directly, but it's not really needed. A long as the machine has enough RAM, the files will

Re: [asterisk-users] Cache sound files for faster processing

On Tue, Apr 6, 2010 at 12:36 AM, huu giang huugiang...@yahoo.com wrote: Dear List, Are there any way of configuring of Asterisk so it'll cache sound files in memory, and when Asterisk receive a call, instead of loading sound files from the disk, it will load from the memory and so Asterisk

[asterisk-users] Cache sound files for faster processing

Dear List, Are there any way of configuring of Asterisk so it'll cache sound files in memory, and when Asterisk receive a call, instead of loading sound files from the disk, it will load from the memory and so Asterisk can process much more call at a time than with faster speed it is not

Re: [asterisk-users] Cache sound files for faster processing

Are there any way of configuring of Asterisk so it'll cache sound files in memory, and when Asterisk receive a call, instead of loading sound files from the disk Not directly, but it's not really needed. A long as the machine has enough RAM, the files will be served from RAM by the operating

[asterisk-users] Robotic sound sometimes

We have inherited an installation with Ast 1.4 and Aastra phones. The client complains that sometimes the call audio turns tinny and robotic...I heard it and it sounds wierd. Has anyone else experienced this? Cause? Solutions? Thanks, MD --

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