Re: [Asterisk-Users] Line Noise HELP!
Hi, For those that were having the same line noise problem that we were, an update: * Our TDM400P *was* sharing an IRQ, despite the output from 'cat /proc/interrupts' showing that it wasn't. Running 'lspci -v' showed that it was and we had to perform some card juggling to get it (and our ISDN card) to sit on different IRQ's. * It appears that this change has resulted in a better *average* accuracy via zttest (although as zttest does not provide an 'average' figure, this is anecdotal only). Certainly the 'best' and 'worst' output is no different (best = 99.975586%, worst = 99.963379%). * The users reported a significant improvement on the last full day of use - they experienced a couple of 'spikes' of noise, but no incidences of the prolonged noise that they were experiencing earlier. This was only over one day, though, so not sure if this means the problems licked or if Friday was just a good day. * I've asked Digium for further advice, as they have told me that we should try and get no less than 99.98% accuracy via zttest. * They've recommended disabling hyper threading and we are yet to give this a go. Will be pretty annoyed if we have to leave hyper threading turned off in order to get this to work, as performance is sure to suffer. Keen to hear if anyone else has managed to get anywhere with their noise problem. Not sure if we're on the right track or not, but will report back in a couple of days to see if the intermittent crackling has returned. Rich Adamson wrote: Excerpt of email from Digium support: Digium does not support CAPI or BRI, either freely or commercially. We can only provide support for you Digium TDM400P card. Your zttest output is extemely low. We are looking for nothing less than 99.98%. If you are receiving 99.96% this will cause major problems. Have a best output of 99.975% is really low. For the record, I ran zttest against a new TDM04b Rev H board and it consistently reported 99.975586%. There is no noise, crackling, etc. So, not sure what 99.975% is really low is based on. If you are running an IDE hard drive please verify that you are using DMA mode with a UDMA setting of no lower than 2 or higher than 3. UDMA mode 2 is ATA33. UDMA mode 3 is ATA44. This can be done using hdparm. We suggest using hdparm -d 1 -X udma2 -c 3 /dev/[IDE Device]. You can check the status using hdparm /dev/[IDE Device] and hdparm -i /dev/[IDE Device]. If you make modifications to your IDE hard drive settings they will only be kept until you reboot. Also, did the above, which had zero impact on the zttest results. If I try to use spandsp-pre11 for fax reception, it results in far less then usable output (*.tiff), and supposedly that is due to missed frames, missed interrupts, or something like that with the TDM04b card. Checked and double-checked share interrupts, and that isn't a problem. This is on a cvs-head RHv9 box with 2.2ghz processor, and nothing else running on the system. So, best guess is that 99.975586% is impacting fax but not voice. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Line Noise HELP!
Excerpt of email from Digium support: Digium does not support CAPI or BRI, either freely or commercially. We can only provide support for you Digium TDM400P card. Your zttest output is extemely low. We are looking for nothing less than 99.98%. If you are receiving 99.96% this will cause major problems. Have a best output of 99.975% is really low. For the record, I ran zttest against a new TDM04b Rev H board and it consistently reported 99.975586%. There is no noise, crackling, etc. So, not sure what 99.975% is really low is based on. If you are running an IDE hard drive please verify that you are using DMA mode with a UDMA setting of no lower than 2 or higher than 3. UDMA mode 2 is ATA33. UDMA mode 3 is ATA44. This can be done using hdparm. We suggest using hdparm -d 1 -X udma2 -c 3 /dev/[IDE Device]. You can check the status using hdparm /dev/[IDE Device] and hdparm -i /dev/[IDE Device]. If you make modifications to your IDE hard drive settings they will only be kept until you reboot. Also, did the above, which had zero impact on the zttest results. If I try to use spandsp-pre11 for fax reception, it results in far less then usable output (*.tiff), and supposedly that is due to missed frames, missed interrupts, or something like that with the TDM04b card. Checked and double-checked share interrupts, and that isn't a problem. This is on a cvs-head RHv9 box with 2.2ghz processor, and nothing else running on the system. So, best guess is that 99.975586% is impacting fax but not voice. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Line Noise HELP!
rusty*CLI show version
Asterisk CVS-HEAD-03/26/05-17:05:44 built by [EMAIL PROTECTED] on a
i686 running Linux
rusty*CLI
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rich
Adamson
Sent: Monday, April 11, 2005 6:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Line Noise HELP!
And, what asterisk version are you running?
Ooops, sorry folks.. A correction..
I don't have digium X100 cards, I have Digit Networks X100P clone cards..
Don't know if it
matters, but wanted to get the facts straight :-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andre Normandin
Sent: Monday, April 11, 2005 5:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Line Noise HELP!
Hi,
I'm having very similiar problems.. However, I'm running a development
version, and it
happens on both SIP phones, and on Analog
phones connected via Sipura SPA-2000's (I have 2 different SPA2000's,
and 4 analog lines..
Seems to happen on all of them as well)..
The problem seems to be EXACTLY as described.
THe call seems fine at first, then within minutes the call degrades to
the point that
neither end can hear each other.. First, the volume
seems to lower, and then static, breaking up, etc..
I have both DIGIUM X100 cards for my pots lines (3 of them), and
BROADVOICE for outgoing
calls. It seems to happen no matter if I'm
on an analog line (I.E. someone called me), or if it was me that
initiated the call
(BROADVOICE outbound).
I do have a 'remote' SIPURA SPA2000 located at a friends house in a
different state -- he
is an extension on my pbx so he can call me, and
he can call his friends locally (He just moved away) via my POTS or
BROADVOICE line.. He
experiences the same problems as I described
above, unless he calls me directly at my 'internal' extension, or I
call him at his
'internal' extension.. I.E. If it doesn't touch POTS or
BROADVOICE, the problem doesn't seem to occur..??
The other interesting thing that has happened of recent development is
that some people
are complaining that they are hearing the
'electronic beep' sound as if the call is being recorded, but I am not
recording the call.
This has occured with my friend as well as incoming
and outgoing POTS/BROADVOICE calls.
If anyone has an idea, I'd love to hear it.. The problem is driving me
(and others who
talk to me) crazy!!!
- Andre
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Damian Funnell
Sent: Monday, April 11, 2005 3:08 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Line Noise HELP!
Hi mate,
Interesting - you're using a different version of Asterisk than I
am, but your problem
sounds identical. We thought it was the SIP
phones that we were using as well, but then it started occurring
on the analogue
phones as well.
Post again when you've tried a new phone, will you? Let us know
if the problem goes
away.
Cheers,
Damian.
Paul wrote:
@page Section1 {size: 8.5in 11.0in; margin: 1.0in 77.95pt
1.0in 77.95pt; }
P.MsoNormal { FONT-SIZE:
12pt; MARGIN: 0in 0in 0pt; COLOR: #66; FONT-FAMILY: Times
New Roman }
LI.MsoNormal {
FONT-SIZE: 12pt; MARGIN: 0in 0in 0pt; COLOR: #66;
FONT-FAMILY: Times New
Roman }
DIV.MsoNormal { FONT-SIZE: 12pt; MARGIN: 0in 0in 0pt; COLOR:
#66; FONT-FAMILY:
Times
New Roman } A:link { COLOR: blue; TEXT-DECORATION:
underline } SPAN.MsoHyperlink
{ COLOR:
blue; TEXT-DECORATION: underline } A:visited { COLOR: blue;
TEXT-DECORATION:
underline }
SPAN.MsoHyperlinkFollowed { COLOR: blue; TEXT-DECORATION:
underline }
P.MsoPlainText {
FONT-SIZE: 10pt; MARGIN: 0in 0in 0pt; FONT-FAMILY: Courier
New } LI.MsoPlainText
{
FONT-SIZE: 10pt; MARGIN: 0in 0in 0pt; FONT-FAMILY: Courier
New }
DIV.MsoPlainText {
FONT-SIZE: 10pt; MARGIN: 0in 0in 0pt; FONT-FAMILY: Courier
New } PRE {
FONT-SIZE: 10pt;
MARGIN: 0in 0in 0pt; COLOR: #66; FONT-FAMILY: Courier
New }
SPAN.EmailStyle18 {
COLOR: navy; FONT-FAMILY: Arial; mso-style-type: personal }
DIV.Section1 { page:
Section1 }
Damian,
pbx*CLI show version
Asterisk CVS-HEAD-03/23/05-00:44:07 built by [EMAIL PROTECTED] on
a
i586 running Linux
There is my version info. Someone on the list has suggested
that its my SIPura
phone. It could very well be the phone, but it
just seems unlikely that the conversation would be perfectly
clear for some time
before the static starts. I tried
Re: [Asterisk-Users] Line Noise HELP!
Thanks for that Rich. Etheral trace is going to be almost impossible for various reasons, but will try the other two options. Can't find much online re. debugging - any chance of killing the box by turning this on? SIP show channels and the various CAPI show commands do not show anything out of the ordinary when the problem occurs. In order for anyone to help identify the noise problem, you really are going to have to find a way to capture some data, otherwise we're all spinning our wheels and guessing. To implement debugging, look at /etc/asterisk/logger.conf and add the keyword 'debug' like: messages = notice,warning,error,debug Adding that keyword requires that * be stopped and restarted to take effect. That tells asterisk to log all debug statements (that are embedded in asterisk source code) to write to /var/log/asterisk/debug file. That debug file will grow to a very large size rather quickly, so you need to pay attention to available disk space, etc. When the noise problem occurs, note the specific system time, and take a look at /var/log/asterisk/debug to see what was happening around that time. Once you've captured at least some data, you may want to remove the debug statement. If you haven't tried some of the other cli debug tools, you might want to do help sip debug, help rtp debug, etc. If you can't run ethereal on the system with the problem, there are other tools like tcpdump, etc, that can be used to capture packets. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line Noise HELP!
Hi Rich, Hear your point about the trace and all, will try and figure something out. Will also look at logging debug messages. We did the unthinkable and purchased a support incident through Digium and they have zeroed in on the zttest output, as per the info below (I've pasted in an excerpt from their email in case anyone else finds it useful). Not sure if this is the cause of our problem or not, as I don't understand whether the TDM card is used as a timing source for calls over CAPI, but will look at getting zttest output up regardless. Have to say that I am pretty impressed with Digium support so far - the engineer even rang me in New Zealand to follow up on their email and to inform me that I still have 45 minutes of my hour left to use. Cheers, Damian. Excerpt of email from Digium support: Digium does not support CAPI or BRI, either freely or commercially. We can only provide support for you Digium TDM400P card. Your zttest output is extemely low. We are looking for nothing less than 99.98%. If you are receiving 99.96% this will cause major problems. Have a best output of 99.975% is really low. Please ensure that you are running either Asterisk Release 1.07, Zaptel Release 1.07, and Libpri Release 1.07 (Libpri only required if using a PRI) from www.asterisk.org or Asterisk CVS Stable/HEAD, Zaptel CVS Stable/HEAD, and Libpri CVS Stable/HEAD (Libpri only required if using a PRI) as of the current date from Digium's CVS server. You may obtain instructions on downloading CVS Stable/HEAD from Digium's CVS server by visiting the download area on www.asterisk.org. Please verify that your Digium hardware is not sharing an IRQ on your system. You can accomplish this by running cat /proc/interrupts. Do not solely rely on cat /proc/interrupts to determine whether your Digium hardware is sharing an IRQ on your system. Make sure your Digium hardware is on its own IRQ by itself and that it is taking interrupts. You can determine whether it is taking interrupts from the 2nd column of output from cat /proc/interrupts. This should be something other than zero. You will also need to verify that your Digium hardware is not sharing an IRQ by examining the output after runninglspci -v and lspci -vb. Using lspci is the best way to determine whether or not your Digium hardware is sharing an IRQ on your system. Please verify that all Digium hardware is on its very own IRQ by itself. You may need to disable unnecessary hardware on your machine such as sound controllers, USB controllers, extra ethernet controllers, firewire, parallel ports, and/or serial ports. You should try moving and swapping our card to different PCI slots in order to get it on it's own IRQ. Some BIOS's will allow you to specify an IRQ for each PCI slot and/or onboard devices. If you are running an IDE hard drive please verify that you are using DMA mode with a UDMA setting of no lower than 2 or higher than 3. UDMA mode 2 is ATA33. UDMA mode 3 is ATA44. This can be done using hdparm. We suggest using hdparm -d 1 -X udma2 -c 3 /dev/[IDE Device]. You can check the status using hdparm /dev/[IDE Device] and hdparm -i /dev/[IDE Device]. If you make modifications to your IDE hard drive settings they will only be kept until you reboot. You will need to add the hdparm command you executed to one of your distribution's startup scripts. This way the IDE hard drive settings will be updated on each reboot. You can check whether or not your Digium hardware on your system is experiencing IRQ misses by using the zttest application which should be located in yourzaptel source directory. Do not solely rely on zttest to determine whether you are having IRQ misses with your Digium hardware on your system. Optimally,we are looking for output of 100% from zttest. Our cards will function properly as long as they do not report back less than 99.98%. Some people have reported no apparent problems with output as low as 99.975%, while others will have many apparent problems with an output as low 99.975%. You are almost guaranteed to have many apparent problems with an output lower than 99.975%. We strongly suggest doing everything possible in order to obtain atleast 99.98% output from zttest. I would watch the output over a 5 minutes period to check for spikes on intervals. You may also look for IRQ misses using the zttool application. Do not solely rely on zttool to determine whether you are having IRQ misses with your Digium hardware on your system. This application should be built while compiling zaptel. zttool requires the libnewt development package to be installed on your system in order to compile properly. IRQ misses with your Digium hardware can be due to I/O problems on your system. You may test if you are having I/O problems on your system by running hdparm -t /dev/[Hard Drive Device]. This will causes massive amounts of I/O on your system. The symptoms of an I/O problem on your system could be cracklingand/or static
RE: [Asterisk-Users] Line Noise HELP!
Damian, Cheers to you for buying that support hour and posting their response. Hopefully we all can come to a solution for this pesky problem and in a very timely manner. I will try all of the suggestions and post any solutions I find. I made a call today on the SIP phone and it was clear for abou 45 seconds and then went to static/line noise for about 3 seconds and then returned to normal. The remaining 30 - 60 seconds of the conversation was normal. I'm wondering if maybe it has to do with network congestion. I called the SIP phone from a separate landline and listened and waited for the static...it never came. I wish the problem was less sporadic. Thanks again for your post. Cheers, Paul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damian Funnell Sent: Tuesday, April 12, 2005 14:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Line Noise HELP! Hi Rich, Hear your point about the trace and all, will try and figure something out. Will also look at logging debug messages. We did the unthinkable and purchased a support incident through Digium and they have zeroed in on the zttest output, as per the info below (I've pasted in an excerpt from their email in case anyone else finds it useful). Not sure if this is the cause of our problem or not, as I don't understand whether the TDM card is used as a timing source for calls over CAPI, but will look at getting zttest output up regardless. Have to say that I am pretty impressed with Digium support so far - the engineer even rang me in New Zealand to follow up on their email and to inform me that I still have 45 minutes of my hour left to use. Cheers, Damian. Excerpt of email from Digium support: Digium does not support CAPI or BRI, either freely or commercially. We can only provide support for you Digium TDM400P card. Your zttest output is extemely low. We are looking for nothing less than 99.98%. If you are receiving 99.96% this will cause major problems. Have a best output of 99.975% is really low. Please ensure that you are running either Asterisk Release 1.07, Zaptel Release 1.07, and Libpri Release 1.07 (Libpri only required if using a PRI) from www.asterisk.org or Asterisk CVS Stable/HEAD, Zaptel CVS Stable/HEAD, and Libpri CVS Stable/HEAD (Libpri only required if using a PRI) as of the current date from Digium's CVS server. You may obtain instructions on downloading CVS Stable/HEAD from Digium's CVS server by visiting the download area on www.asterisk.org. Please verify that your Digium hardware is not sharing an IRQ on your system. You can accomplish this by running cat /proc/interrupts. Do not solely rely on cat /proc/interrupts to determine whether your Digium hardware is sharing an IRQ on your system. Make sure your Digium hardware is on its own IRQ by itself and that it is taking interrupts. You can determine whether it is taking interrupts from the 2nd column of output from cat /proc/interrupts. This should be something other than zero. You will also need to verify that your Digium hardware is not sharing an IRQ by examining the output after runninglspci -v and lspci -vb. Using lspci is the best way to determine whether or not your Digium hardware is sharing an IRQ on your system. Please verify that all Digium hardware is on its very own IRQ by itself. You may need to disable unnecessary hardware on your machine such as sound controllers, USB controllers, extra ethernet controllers, firewire, parallel ports, and/or serial ports. You should try moving and swapping our card to different PCI slots in order to get it on it's own IRQ. Some BIOS's will allow you to specify an IRQ for each PCI slot and/or onboard devices. If you are running an IDE hard drive please verify that you are using DMA mode with a UDMA setting of no lower than 2 or higher than 3. UDMA mode 2 is ATA33. UDMA mode 3 is ATA44. This can be done using hdparm. We suggest using hdparm -d 1 -X udma2 -c 3 /dev/[IDE Device]. You can check the status using hdparm /dev/[IDE Device] and hdparm -i /dev/[IDE Device]. If you make modifications to your IDE hard drive settings they will only be kept until you reboot. You will need to add the hdparm command you executed to one of your distribution's startup scripts. This way the IDE hard drive settings will be updated on each reboot. You can check whether or not your Digium hardware on your system is experiencing IRQ misses by using the zttest application which should be located in yourzaptel source directory. Do not solely rely on zttest to determine whether you are having IRQ misses with your Digium hardware on your system. Optimally,we are looking for output of 100% from zttest. Our cards will function properly as long as they do not report back less than 99.98%. Some people have reported no apparent problems with output as low as 99.975%, while others will have many apparent problems with an output as low
Re: [Asterisk-Users] Line Noise HELP!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Monday 11 April 2005 10:48 am, Damian Funnell wrote: Hi Paul, there was a thread yesterday in regards to a few of us experiencing a very similar problem - a problem that (if the same for all of us), doesn't seem to have been properly diagnosed yet. One thing that appeared to be common to all of us was the version of Asterisk that we are running (1.0.6), is this the version that you are running? What FXO card are you using? Does the noise that you are complaining about occur on every call and, if so, always after exactly a minute, or is it more random? Starting to wonder if there isn't a problem with Stable, interested to hear what version you're running. Afraid not. We were having the same problem. It occurred with Snom220 and analogue phones hooked up to Sipura 2000's and 3000's. It happened whether we were doing internal calls, or outgoing calls. Eventually, we had to pull our machine out. We were using: Asterisk CVS-v1-0-04/04/05-02:14:16 built by [EMAIL PROTECTED] on a i686 running Linux The Linux distribution was: Fedora Core release 2 (Tettnang) We had 256MB's of memory, 1GB swap space, 866MHz 64KB cache VIA Nehemiah (1714 bogomips). What would happen is that after a minute or less, we would start getting voice distortion (slight vibrational sound to the voice), which would eventually worsen until it was so bad after a couple of minutes that you couldn't understand what the person on the other end of the line was saying. Also, the volume through the snom220 was very low, sometimes so low that if there was background noise in your surroundings, you wouldn't be able to hear the person on the other end of the line. During the whole time (from start of connection), a slight echo was also discernible on the line. Other aspects of this environment that might have some effect: - installation was originally ground-start; we switched it over to loop-start - there were 14 sipura 2000's - 3 sipura 3000's - 3 outbound lines - 31 analogue phones - 1 snom220 voip phone w/extra keys - 1 DLink 24-port switch - 1 power outlet - 5(?) power bars (to power DLink and all sipura units) Additionally, the wiring was relatively old in this building (it was located in a small town about 1-1/2 hours from the nearest city). Perhaps with this information, you might be able to get some other ideas on what the problem is. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iD8DBQFCXDu4gYKvkeyp3F4RAgtiAJ46FAywh4S02wlHJpduYgO65f1kuwCeKErP 3V7Z+D0t9yDSC96Sg+8+DyY= =RQwn -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line Noise HELP!
Hi Paul, there was a thread yesterday in regards to a few of us experiencing a very similar problem - a problem that (if the same for all of us), doesn't seem to have been properly diagnosed yet. One thing that appeared to be common to all of us was the version of Asterisk that we are running (1.0.6), is this the version that you are running? What FXO card are you using? Does the noise that you are complaining about occur on every call and, if so, always after exactly a minute, or is it more random? Starting to wonder if there isn't a problem with Stable, interested to hear what version you're running. Damian. Paul wrote: I recently hooked up my sipura IP phone and set it up as an SIP device to connect to asterisk. I am able to dial a number on the SIP phone, connect to an external number via the PSTN connected to asterisk and begin the conversation. At first, the audio quality is PERFECT, in both directions. I can hear the person clearly and they claim to hear me like im on a regular POTS line. After approximately 1 minute, the quality turns horrible and the person can no longer hear me, but I can faintly here them. There is a lot of static on the line, it almost sounds like an electronic device is interfering with it. I thought maybe it was a wireless phone or router, so I disconnected all those and put my cell phone in the other room. Still no change. Anyone have any ideas, this is really getting to be a problem. Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Line Noise HELP!
Damian, pbx*CLI show version Asterisk CVS-HEAD-03/23/05-00:44:07 built by [EMAIL PROTECTED] on a i586 running Linux There is my version info. Someone on the list has suggested that its my SIPura phone. It could very well be the phone, but it just seems unlikely that the conversation would be perfectly clear for some time before the static starts. I tried it today and was able to go approximately two minutes before it started. The FXO card is a generic x100P. Im going to try to get another IP phone and test it to see if its the phone. Let me know if you come up with any ideas. Paul From: Damian Funnell [mailto:[EMAIL PROTECTED] Sent: Monday, April 11, 2005 12:49 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Line Noise HELP! Hi Paul, there was a thread yesterday in regards to a few of us experiencing a very similar problem - a problem that (if the same for all of us), doesn't seem to have been properly diagnosed yet. One thing that appeared to be common to all of us was the version of Asterisk that we are running (1.0.6), is this the version that you are running? What FXO card are you using? Does the noise that you are complaining about occur on every call and, if so, always after exactly a minute, or is it more random? Starting to wonder if there isn't a problem with Stable, interested to hear what version you're running. Damian. Paul wrote: I recently hooked up my sipura IP phone and set it up as an SIP device to connect to asterisk. I am able to dial a number on the SIP phone, connect to an external number via the PSTN connected to asterisk and begin the conversation. At first, the audio quality is PERFECT, in both directions. I can hear the person clearly and they claim to hear me like im on a regular POTS line. After approximately 1 minute, the quality turns horrible and the person can no longer hear me, but I can faintly here them. There is a lot of static on the line, it almost sounds like an electronic device is interfering with it. I thought maybe it was a wireless phone or router, so I disconnected all those and put my cell phone in the other room. Still no change. Anyone have any ideas, this is really getting to be a problem. Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line Noise HELP!
Hi mate, Interesting - you're using a different version of Asterisk than I am, but your problem sounds identical. We thought it was the SIP phones that we were using as well, but then it started occurring on the analogue phones as well. Post again when you've tried a new phone, will you? Let us know if the problem goes away. Cheers, Damian. Paul wrote: Damian, pbx*CLI show version Asterisk CVS-HEAD-03/23/05-00:44:07 built by [EMAIL PROTECTED] on a i586 running Linux There is my version info. Someone on the list has suggested that its my SIPura phone. It could very well be the phone, but it just seems unlikely that the conversation would be perfectly clear for some time before the static starts. I tried it today and was able to go approximately two minutes before it started. The FXO card is a generic x100P. Im going to try to get another IP phone and test it to see if its the phone. Let me know if you come up with any ideas. Paul From: Damian Funnell [mailto:[EMAIL PROTECTED]] Sent: Monday, April 11, 2005 12:49 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Line Noise HELP! Hi Paul, there was a thread yesterday in regards to a few of us experiencing a very similar problem - a problem that (if the same for all of us), doesn't seem to have been properly diagnosed yet. One thing that appeared to be common to all of us was the version of Asterisk that we are running (1.0.6), is this the version that you are running? What FXO card are you using? Does the noise that you are complaining about occur on every call and, if so, always after exactly a minute, or is it more random? Starting to wonder if there isn't a problem with Stable, interested to hear what version you're running. Damian. Paul wrote: I recently hooked up my sipura IP phone and set it up as an SIP device to connect to asterisk. I am able to dial a number on the SIP phone, connect to an external number via the PSTN connected to asterisk and begin the conversation. At first, the audio quality is PERFECT, in both directions. I can hear the person clearly and they claim to hear me like im on a regular POTS line. After approximately 1 minute, the quality turns horrible and the person can no longer hear me, but I can faintly here them. There is a lot of static on the line, it almost sounds like an electronic device is interfering with it. I thought maybe it was a wireless phone or router, so I disconnected all those and put my cell phone in the other room. Still no change. Anyone have any ideas, this is really getting to be a problem. Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Line Noise HELP!
Hi, I'm having very similiar problems.. However, I'm running a development version, and it happens on both SIP phones, and on Analog phones connected via Sipura SPA-2000's (I have 2 different SPA2000's, and 4 analog lines.. Seems to happen on all of them as well).. The problem seems to be EXACTLY as described. THe call seems fine at first, then within minutes the call degrades to the point that neither end can hear each other.. First, the volume seems to lower, and then static, breaking up, etc.. I have both DIGIUM X100 cards for my pots lines (3 of them), and BROADVOICE for outgoing calls. It seems to happen no matter if I'm on an analog line (I.E. someone called me), or if it was me that initiated the call (BROADVOICE outbound). I do have a 'remote' SIPURA SPA2000 located at a friends house in a different state -- he is an extension on my pbx so he can call me, and he can call his friends locally (He just moved away) via my POTS or BROADVOICE line.. He experiences the same problems as I described above, unless he calls me directly at my 'internal' extension, or I call him at his 'internal' extension.. I.E. If it doesn't touch POTS or BROADVOICE, the problem doesn't seem to occur..?? The other interesting thing that has happened of recent development is that some people are complaining that they are hearing the 'electronic beep' sound as if the call is being recorded, but I am not recording the call. This has occured with my friend as well as incoming and outgoing POTS/BROADVOICE calls. If anyone has an idea, I'd love to hear it.. The problem is driving me (and others who talk to me) crazy!!! - Andre -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Damian FunnellSent: Monday, April 11, 2005 3:08 PMTo: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Line Noise HELP!Hi mate,Interesting - you're using a different version of Asterisk than I am, but your problem sounds identical. We thought it was the SIP phones that we were using as well, but then it started occurring on the analogue phones as well.Post again when you've tried a new phone, will you? Let us know if the problem goes away.Cheers,Damian.Paul wrote: Damian, pbx*CLI show version Asterisk CVS-HEAD-03/23/05-00:44:07 built by [EMAIL PROTECTED] on a i586 running Linux There is my version info. Someone on the list has suggested that its my SIPura phone. It could very well be the phone, but it just seems unlikely that the conversation would be perfectly clear for some time before the static starts. I tried it today and was able to go approximately two minutes before it started. The FXO card is a generic x100P. Im going to try to get another IP phone and test it to see if its the phone. Let me know if you come up with any ideas. Paul From: Damian Funnell [mailto:[EMAIL PROTECTED]] Sent: Monday, April 11, 2005 12:49 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Line Noise HELP! Hi Paul, there was a thread yesterday in regards to a few of us experiencing a very similar problem - a problem that (if the same for all of us), doesn't seem to have been properly diagnosed yet. One thing that appeared to be common to all of us was the version of Asterisk that we are running (1.0.6), is this the version that you are running? What FXO card are you using? Does the noise that you are complaining about occur on every call and, if so, always after exactly a minute, or is it more random? Starting to wonder if there isn't a problem with Stable, interested to hear what version you're running. Damian. Paul wrote: I recently hooked up my sipura IP phone and set it up as an SIP device to connect to asterisk. I am able to dial a number on the SIP phone, connect to an external number via the PSTN connected to asterisk and begin the conversation. At first, the audio quality is PERFECT, in both directions. I can hear the person clearly and they claim to hear me like im on a regular POTS line. After approximately 1 minute, the quality turns horrible and the person can no longer hear me, but I can faintly here them. There is a lot of static on the line, it almost sounds like an electronic device is interfering with it. I thought maybe it was a wireless phone or router, so I disconnected all those and put my cell phone in the other room. Still no change. Anyone have any ideas, this is really getting to be a problem. Paul
RE: [Asterisk-Users] Line Noise HELP!
Ooops, sorry folks.. A correction.. I don't have digium X100 cards, I have Digit Networks X100P clone cards.. Don't know if it matters, but wanted to get the facts straight :-) -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Andre NormandinSent: Monday, April 11, 2005 5:06 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] Line Noise HELP! Hi, I'm having very similiar problems.. However, I'm running a development version, and it happens on both SIP phones, and on Analog phones connected via Sipura SPA-2000's (I have 2 different SPA2000's, and 4 analog lines.. Seems to happen on all of them as well).. The problem seems to be EXACTLY as described. THe call seems fine at first, then within minutes the call degrades to the point that neither end can hear each other.. First, the volume seems to lower, and then static, breaking up, etc.. I have both DIGIUM X100 cards for my pots lines (3 of them), and BROADVOICE for outgoing calls. It seems to happen no matter if I'm on an analog line (I.E. someone called me), or if it was me that initiated the call (BROADVOICE outbound). I do have a 'remote' SIPURA SPA2000 located at a friends house in a different state -- he is an extension on my pbx so he can call me, and he can call his friends locally (He just moved away) via my POTS or BROADVOICE line.. He experiences the same problems as I described above, unless he calls me directly at my 'internal' extension, or I call him at his 'internal' extension.. I.E. If it doesn't touch POTS or BROADVOICE, the problem doesn't seem to occur..?? The other interesting thing that has happened of recent development is that some people are complaining that they are hearing the 'electronic beep' sound as if the call is being recorded, but I am not recording the call. This has occured with my friend as well as incoming and outgoing POTS/BROADVOICE calls. If anyone has an idea, I'd love to hear it.. The problem is driving me (and others who talk to me) crazy!!! - Andre -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Damian FunnellSent: Monday, April 11, 2005 3:08 PMTo: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Line Noise HELP!Hi mate,Interesting - you're using a different version of Asterisk than I am, but your problem sounds identical. We thought it was the SIP phones that we were using as well, but then it started occurring on the analogue phones as well.Post again when you've tried a new phone, will you? Let us know if the problem goes away.Cheers,Damian.Paul wrote: Damian, pbx*CLI show version Asterisk CVS-HEAD-03/23/05-00:44:07 built by [EMAIL PROTECTED] on a i586 running Linux There is my version info. Someone on the list has suggested that its my SIPura phone. It could very well be the phone, but it just seems unlikely that the conversation would be perfectly clear for some time before the static starts. I tried it today and was able to go approximately two minutes before it started. The FXO card is a generic x100P. Im going to try to get another IP phone and test it to see if its the phone. Let me know if you come up with any ideas. Paul From: Damian Funnell [mailto:[EMAIL PROTECTED]] Sent: Monday, April 11, 2005 12:49 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Line Noise HELP! Hi Paul, there was a thread yesterday in regards to a few of us experiencing a very similar problem - a problem that (if the same for all of us), doesn't seem to have been properly diagnosed yet. One thing that appeared to be common to all of us was the version of Asterisk that we are running (1.0.6), is this the version that you are running? What FXO card are you using? Does the noise that you are complaining about occur on every call and, if so, always after exactly a minute, or is it more random? Starting to wonder if there isn't a problem with Stable, interested to hear what version you're running. Damian. Paul wrote: I recently hooked up my sipura IP phone and set it up as an SIP device to connect to asterisk. I am able to dial a number on the SIP phone, connect to an external number via the PSTN connected to asterisk and begin
Re: [Asterisk-Users] Line Noise HELP!
Can't help but wonder if this isn't a bug in Asterisk or one of it's modules, as there seems to be a lot of people experiencing the same problem, seemingly with different hardware and software configurations. Anyone know how (or if it's possible) to submit a bug report to Digium regarding this type of problem? I for one am going to have a customer return their Asterisk box for good if we can't get to the bottom of this soon. Andre Normandin wrote: Hi, I'm having very similiar problems.. However, I'm running a development version, and it happens on both SIP phones, and on Analog phones connected via Sipura SPA-2000's (I have 2 different SPA2000's, and 4 analog lines.. Seems to happen on all of them as well).. The problem seems to be EXACTLY as described. THe call seems fine at first, then within minutes the call degrades to the point that neither end can hear each other.. First, the volume seems to lower, and then static, breaking up, etc.. I have both DIGIUM X100 cards for my pots lines (3 of them), and BROADVOICE for outgoing calls. It seems to happen no matter if I'm on an analog line (I.E. someone called me), or if it was me that initiated the call (BROADVOICE outbound). I do have a 'remote' SIPURA SPA2000 located at a friends house in a different state -- he is an extension on my pbx so he can call me, and he can call his friends locally (He just moved away) via my POTS or BROADVOICE line.. He experiences the same problems as I described above, unless he calls me directly at my 'internal' extension, or I call him at his 'internal' extension.. I.E. If it doesn't touch POTS or BROADVOICE, the problem doesn't seem to occur..?? The other interesting thing that has happened of recent development is that some people are complaining that they are hearing the 'electronic beep' sound as if the call is being recorded, but I am not recording the call. This has occured with my friend as well as incoming and outgoing POTS/BROADVOICE calls. If anyone has an idea, I'd love to hear it.. The problem is driving me (and others who talk to me) crazy!!! - Andre -Original Message- *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of *Damian Funnell *Sent:* Monday, April 11, 2005 3:08 PM *To:* [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] Line Noise HELP! Hi mate, Interesting - you're using a different version of Asterisk than I am, but your problem sounds identical. We thought it was the SIP phones that we were using as well, but then it started occurring on the analogue phones as well. Post again when you've tried a new phone, will you? Let us know if the problem goes away. Cheers, Damian. Paul wrote: Damian, pbx*CLI show version Asterisk CVS-HEAD-03/23/05-00:44:07 built by [EMAIL PROTECTED] on a i586 running Linux There is my version info. Someone on the list has suggested that its my SIPura phone. It could very well be the phone, but it just seems unlikely that the conversation would be perfectly clear for some time before the static starts. I tried it today and was able to go approximately two minutes before it started. The FXO card is a generic x100P. Im going to try to get another IP phone and test it to see if its the phone. Let me know if you come up with any ideas. Paul From: Damian Funnell [mailto:[EMAIL PROTECTED] Sent: Monday, April 11, 2005 12:49 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Line Noise HELP! Hi Paul, there was a thread yesterday in regards to a few of us experiencing a very similar problem - a problem that (if the same for all of us), doesn't seem to have been properly diagnosed yet. One thing that appeared to be common to all of us was the version of Asterisk that we are running (1.0.6), is this the version that you are running? What FXO card are you using? Does the noise that you are complaining about occur on every call and, if so, always after exactly a minute, or is it more random? Starting to wonder if there isn't a problem with Stable, interested to hear what version you're running. Damian. Paul wrote: I recently hooked up my sipura IP phone and set it up as an SIP device to connect to asterisk. I am able to dial a number on the SIP phone, connect to an external number via the PSTN connected to asterisk and begin the conversation. At first, the audio quality is PERFECT, in both directions. I can hear the person clearly and they claim to hear me like im on a regular POTS line. After approximately 1 minute, the quality turns horrible and the person can no longer hear me, but I can faintly here them. There is a lot of static on the line
RE: [Asterisk-Users] Line Noise HELP!
And, what asterisk version are you running?
Ooops, sorry folks.. A correction..
I don't have digium X100 cards, I have Digit Networks X100P clone cards..
Don't know if it
matters, but wanted to get the facts straight :-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andre Normandin
Sent: Monday, April 11, 2005 5:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Line Noise HELP!
Hi,
I'm having very similiar problems.. However, I'm running a development
version, and it
happens on both SIP phones, and on Analog
phones connected via Sipura SPA-2000's (I have 2 different SPA2000's, and
4 analog lines..
Seems to happen on all of them as well)..
The problem seems to be EXACTLY as described.
THe call seems fine at first, then within minutes the call degrades to
the point that
neither end can hear each other.. First, the volume
seems to lower, and then static, breaking up, etc..
I have both DIGIUM X100 cards for my pots lines (3 of them), and
BROADVOICE for outgoing
calls. It seems to happen no matter if I'm
on an analog line (I.E. someone called me), or if it was me that
initiated the call
(BROADVOICE outbound).
I do have a 'remote' SIPURA SPA2000 located at a friends house in a
different state -- he
is an extension on my pbx so he can call me, and
he can call his friends locally (He just moved away) via my POTS or
BROADVOICE line.. He
experiences the same problems as I described
above, unless he calls me directly at my 'internal' extension, or I call
him at his
'internal' extension.. I.E. If it doesn't touch POTS or
BROADVOICE, the problem doesn't seem to occur..??
The other interesting thing that has happened of recent development is
that some people
are complaining that they are hearing the
'electronic beep' sound as if the call is being recorded, but I am not
recording the call.
This has occured with my friend as well as incoming
and outgoing POTS/BROADVOICE calls.
If anyone has an idea, I'd love to hear it.. The problem is driving me
(and others who
talk to me) crazy!!!
- Andre
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Damian Funnell
Sent: Monday, April 11, 2005 3:08 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Line Noise HELP!
Hi mate,
Interesting - you're using a different version of Asterisk than I am,
but your problem
sounds identical. We thought it was the SIP
phones that we were using as well, but then it started occurring on
the analogue
phones as well.
Post again when you've tried a new phone, will you? Let us know if
the problem goes
away.
Cheers,
Damian.
Paul wrote:
@page Section1 {size: 8.5in 11.0in; margin: 1.0in 77.95pt 1.0in
77.95pt; }
P.MsoNormal { FONT-SIZE:
12pt; MARGIN: 0in 0in 0pt; COLOR: #66; FONT-FAMILY: Times
New Roman }
LI.MsoNormal {
FONT-SIZE: 12pt; MARGIN: 0in 0in 0pt; COLOR: #66;
FONT-FAMILY: Times New
Roman }
DIV.MsoNormal { FONT-SIZE: 12pt; MARGIN: 0in 0in 0pt; COLOR:
#66; FONT-FAMILY:
Times
New Roman } A:link { COLOR: blue; TEXT-DECORATION: underline }
SPAN.MsoHyperlink
{ COLOR:
blue; TEXT-DECORATION: underline } A:visited { COLOR: blue;
TEXT-DECORATION:
underline }
SPAN.MsoHyperlinkFollowed { COLOR: blue; TEXT-DECORATION:
underline }
P.MsoPlainText {
FONT-SIZE: 10pt; MARGIN: 0in 0in 0pt; FONT-FAMILY: Courier New
} LI.MsoPlainText
{
FONT-SIZE: 10pt; MARGIN: 0in 0in 0pt; FONT-FAMILY: Courier New
}
DIV.MsoPlainText {
FONT-SIZE: 10pt; MARGIN: 0in 0in 0pt; FONT-FAMILY: Courier New
} PRE {
FONT-SIZE: 10pt;
MARGIN: 0in 0in 0pt; COLOR: #66; FONT-FAMILY: Courier New }
SPAN.EmailStyle18 {
COLOR: navy; FONT-FAMILY: Arial; mso-style-type: personal }
DIV.Section1 { page:
Section1 }
Damian,
pbx*CLI show version
Asterisk CVS-HEAD-03/23/05-00:44:07 built by [EMAIL PROTECTED] on
a i586 running Linux
There is my version info. Someone on the list has suggested that
its my SIPura
phone. It could very well be the phone, but it
just seems unlikely that the conversation would be perfectly
clear for some time
before the static starts. I tried it today and was
able to go approximately two minutes before it started. The FXO
card is a generic
x100P. Im going to try to get another IP
phone and test it to see
Re: [Asterisk-Users] Line Noise HELP!
Can't help but wonder if this isn't a bug in Asterisk or one of it's modules, as there seems to be a lot of people experiencing the same problem, seemingly with different hardware and software configurations. Anyone know how (or if it's possible) to submit a bug report to Digium regarding this type of problem? I for one am going to have a customer return their Asterisk box for good if we can't get to the bottom of this soon. If I had the problem (which I don't with CVS-HEAD-04/07/05 and several different types of sip devices), I'd start with an ethereal trace that could be shared with those of us that can analyze it. Needs to include packets from when the audio goes to hell. Then, I'd turn on debugging (in logger.conf) and see if any messages are relative to the problem. If you can capture the results of 'sip show channels' and 'sip show channel ' for the bad conversation, that might be helpful to see as well. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line Noise HELP!
Thanks for that Rich. Etheral trace is going to be almost impossible for various reasons, but will try the other two options. Can't find much online re. debugging - any chance of killing the box by turning this on? SIP show channels and the various CAPI show commands do not show anything out of the ordinary when the problem occurs. Cheers, D. Rich Adamson wrote: Can't help but wonder if this isn't a bug in Asterisk or one of it's modules, as there seems to be a lot of people experiencing the same problem, seemingly with different hardware and software configurations. Anyone know how (or if it's possible) to submit a bug report to Digium regarding this type of problem? I for one am going to have a customer return their Asterisk box for good if we can't get to the bottom of this soon. If I had the problem (which I don't with CVS-HEAD-04/07/05 and several different types of sip devices), I'd start with an ethereal trace that could be shared with those of us that can analyze it. Needs to include packets from when the audio goes to hell. Then, I'd turn on debugging (in logger.conf) and see if any messages are relative to the problem. If you can capture the results of 'sip show channels' and 'sip show channel ' for the bad conversation, that might be helpful to see as well. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Line Noise HELP!
On Mon, 2005-04-11 at 17:06 -0400, Andre Normandin wrote: Hi, I'm having very similiar problems.. However, I'm running a development version, and it happens on both SIP phones, and on Analog phones connected via Sipura SPA-2000's (I have 2 different SPA2000's, and 4 analog lines.. Seems to happen on all of them as well).. The problem seems to be EXACTLY as described. THe call seems fine at first, then within minutes the call degrades to the point that neither end can hear each other.. First, the volume seems to lower, and then static, breaking up, etc.. I have both DIGIUM X100 cards for my pots lines (3 of them), and BROADVOICE for outgoing calls. It seems to happen no matter if I'm on an analog line (I.E. someone called me), or if it was me that initiated the call (BROADVOICE outbound). I do have a 'remote' SIPURA SPA2000 located at a friends house in a different state -- he is an extension on my pbx so he can call me, and he can call his friends locally (He just moved away) via my POTS or BROADVOICE line.. He experiences the same problems as I described above, unless he calls me directly at my 'internal' extension, or I call him at his 'internal' extension.. I.E. If it doesn't touch POTS or BROADVOICE, the problem doesn't seem to occur..?? The other interesting thing that has happened of recent development is that some people are complaining that they are hearing the 'electronic beep' sound as if the call is being recorded, but I am not recording the call. This has occured with my friend as well as incoming and outgoing POTS/BROADVOICE calls. If anyone has an idea, I'd love to hear it.. The problem is driving me (and others who talk to me) crazy!!! - Andre I'm not sure about your other problem... but I have heard others complain about beeping with a Sipura ATA. Had something to do with it incorrectly detecting DTMF when there was none. I believe it was fixed in the newer firmware. You might try updating the firmware and see if that fixes it. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
