[asterisk-users] Digium TDMXXB and Electronic Noises
I recently moved an installed and working Asterisk system from one PC to another. I moved two Digium TDMXX cards and the OS as well (a live distro). I tuned the hardware on the new PC, but for some reason analog calls periodically have some electronic noise. It's like beeps, but more musical. I do not recall noticing this on the old PC, but immediately noticed it on the new system. Since the hardware and the OS are the same, I'm not sure what could be causing this issue, or how to remedy it. Any ideas? Thanks, Matthew Yingling ___ -- 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] Digium TDMXXB and Electronic Noises
You are probably both correct. I noticed that both of our TDM cards, and the Ethernet card are all sharing the same IRQ. Since we do VOIP internally and analog externally, that IRQ is getting hit twice for any outbound or inbound calls. The system is new, and the OS supports ACPI, so I'm not yet sure what's going on. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tim Nelson Sent: Thursday, January 31, 2008 1:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Digium TDMXXB and Electronic Noises I second that. IRQ issues are more than likely causing the problem. Check your interrupts and see if your TDM cards are sharing IRQs with any other devices. From past experience, I know we would get the same behavior when an analog card was sharing an IRQ with a storage controller. Any amount of disk activity would cause little blips and beeps in the audio stream. Make sure you have all extraneous unneeded devices turned off in the BIOS. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 31, 2008 11:34:18 AM (GMT-0600) America/Chicago Subject: Re: [asterisk-users] Digium TDMXXB and Electronic Noises On Jan 31, 2008 12:28 PM, Matthew Yingling [EMAIL PROTECTED] wrote: I recently moved an installed and working Asterisk system from one PC to another. I moved two Digium TDMXX cards and the OS as well (a live distro). I tuned the hardware on the new PC, but for some reason analog calls periodically have some electronic noise. It's like beeps, but more musical. I do not recall noticing this on the old PC, but immediately noticed it on the new system. Since the hardware and the OS are the same, I'm not sure what could be causing this issue, or how to remedy it. Any ideas? Thanks, Matthew Yingling IRQ issues i would suspect. Also,just because two machines are the same make and model absolutely does not mean that they have the same hardware. I have seen exact server models ordered from CDW at the same time have very different chipsets. Thanks, Steve Totaro ___ -- 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 ___ -- 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 ___ -- 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] Unified Messaging On Thin Client / Terminal Server
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Wednesday, December 26, 2007 1:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Unified Messaging On Thin Client / Terminal Server Jeffrey Thompson wrote: (Preferably with no other buttons needing to be pushed). The idea is that playing the voicemail over the Thin Client / RDP session won't work or will provide poor sound quality, but instead, to use the VOIP Why would the thin client be unable to provide a good experience doing that? If it was going to provide poor sound quality due to network latency for example, wouldn't it stand to reason that the display would be garbled or slow to respond as well? Just a thought. And would using the VOIP phone lighten the need for network resources? There is the possibility that audio will not be available via thin client, such as with Windows 2000 Terminal Server. Optional audio support was added to WinXP/2003. ___ --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 ___ --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] Soundcard necessary on an asterisk server to get output of playback()??
Search for ztdummy, zttest and Zaptel Issue 11153 in the Dev Mailing List. You might have a buggy kernel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stefan Guenther Sent: Monday, December 03, 2007 5:00 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Soundcard necessary on an asterisk server to get output of playback()?? Hi, My quick guess would be that it's a timing issue. You didn't mention whether you are using a Zaptel device or ztdummy. I'm using ztdummy, and yes, I guess your're right - it seems to be a timing problem, because I found the following messages in /var/log/messages: Dec 3 22:51:36 asterisk kernel: [25713.830465] printk: 249 messages suppressed. Dec 3 22:51:36 asterisk kernel: [25713.830468] rtc: lost some interrupts at 1024Hz. BTW: name -a Linux asterisk 2.6.22-14-386 #1 Sun Oct 14 22:36:54 GMT 2007 i686 GNU/Linux But what does lost some interrupts at 1024Hz tell me? And if it is related to my problem, how do I solve it? Thanks for your help, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ___ --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 ___ --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] Remote Office, Centrally Shared Voicemail
Hi, I'm trying to set up a remote office with its own Asterisk Server they'll have a dedicated land line, but we'll still want them connected to the main office via VOIP (IAX2 via VPN). I've tested using IAX2 to bridge between the offices based on extensions, since the extensions we want to share are in isolated blocks of numbers. I'm not sure how to handle voicemail though. I'd like to link the voicemail so that local calls to either office will call extensions and leave voicemail with the appropriate parties. I'd like to avoid Please call a new number messages. I have some ideas: 1. Use central network storage for both offices - if the remote VPN goes down, the remote office can't connect to the voicemail storage, so they can't see old voicemail, and may lose new voicemail. 2. Use local storage for all voicemail. Only the local office can see or receive voicemail. This would require a Please call a new number message, I think. 3. Implement some sort of backup script - use local storage for each office, then periodically sync voicemail folders over the VPN. Can anyone suggest an approach to this problem? Thanks! Matthew Yingling ___ --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] Caller ID matching
We use this macro, which works quite well:
[macro-checkuservoicemail]
; ${ARG1} - Device extension(s) to check for mail
; Usage
; in main context do exten = 1000,1,Macro(checkuservoicemail,101)
exten = s,1,NoOp(Entering CheckUserVoiceMail for ${MACRO_EXTEN})
exten = s,n,GotoIf($[${MACRO_EXTEN} = ${ARG1}]?:NoMatchVM)
exten = s,n,Playback(beep) ; Hack for UIP200 clipping bug
exten = s,n,VoicemailMain([EMAIL PROTECTED]) ; Check vmail
exten = s,n,Hangup ; Hangup after checking vmail
exten = s,n(NoMatchVM),NoOp(End checkuservoicemail)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mike Hammett
Sent: Tuesday, May 22, 2007 9:37 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Caller ID matching
Yeah, I was trying to have it match the caller ID with what they're dialing
so that I don't have a separate entry for every customer.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rizwan Hisham
Sent: Tuesday, May 22, 2007 5:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Caller ID matching
I did it anyway. i used another way around to do it:
suppose 88777 is your number
exten= 88777,1,Dial(SIP/you)
exten= 88777/88777,1,VoiceMailMain()
but in this case you will have to make a separate vm extension for every
user.
On 5/22/07, Rizwan Hisham [EMAIL PROTECTED] wrote:
well i have tried to solve your problem, making your extensions in my
dialplan and reloading dialplan gives me segmentation fault. im afraid i
cant help u :)
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
On 5/20/07, Mike Hammett [EMAIL PROTECTED] wrote:
What's going on here? 555* seems to indicate that the number is being
passed as the callerID because NoOp says the phone number.
I'm trying to emulate cell phone voicemail where you call your own number
to check your voicemail.
-- Accepting AUTHENTICATED call from 65.182.165.XXX:
requested format = gsm,
requested prefs = (),
actual format = ulaw,
host prefs = (ulaw),
priority = mine
-- Executing NoOp(IAX2/815748-16, 815748) in new stack
-- Executing Hangup(IAX2/815748-16, ) in new stack
== Spawn extension (outbound-scripted, 555*, 2) exited non-zero on
'IAX2/815748-16'
May 20 11:10:34 ERROR[3286]: cdr_addon_mysql.c:144 mysql_log: cdr_mysql:
cannot connect to database server localhost.
-- Hungup 'IAX2/815748-16'
May 20 11:11:00 NOTICE[3275]: chan_iax2.c:7323 socket_read: Rejected
connect attempt from 65.182.165.XXX, request '[EMAIL PROTECTED]'
does not exist
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},1,Answer
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},2,NoOp(It's here)
exten =
${CALLERID(NUM)}/${CALLERID(NUM)},3,VoicemailMain(${CALLERID(NUM)})
exten = ${CALLERID(NUM)}/${CALLERID(NUM)},4,Hangup()
exten = 555*,1,NoOp(${CALLERID(num)})
exten = 555*,2,Hangup
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
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--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
--
Rizwan Hisham
Software Engineer
AXVOICE Inc.
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RE: [asterisk-users] Iaxy clicking
Hi, Can any one suggest if asterisk-users is the best mailing list for questions on Digium Iaxy (S101I) hardware, or a different one if not? I found this link on Digium's site: http://kb.digium.com/entry/15/120/ However, I assume that if this was the case, all of my Iaxys would click, and only one of mine does. Is Digium referring to clicking coming from the FXO/FXS hardware or the Iaxy device? My constant clicking is coming out of the Iaxy, whether or not it's connected to the VOIP network. Thanks, Matthew Yingling -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matthew Yingling Sent: Thursday, May 10, 2007 5:29 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Iaxy clicking Hi, I have three Iaxy devices (s101i) parts. Two of them seem to work fine. The third plays a loud repeating click sound when an analog phone is plugged in. I can provision all of them, and make calls to all of them. The clicking one will blink when a call is incoming, but no audio from the call can be heard on the handset, and the caller only hears silence. The same handset works on the other Iaxys, and other handsets have the same clicking issue. Resetting the Iaxy doesn't seem to fix the problem. Does anyone have any ideas on how to fix this problem, or whether the Iaxy is broken and unfixable (for me as an end-user). Thanks, Matthew Yingling ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Iaxy clicking
Hi, I have three Iaxy devices (s101i) parts. Two of them seem to work fine. The third plays a loud repeating click sound when an analog phone is plugged in. I can provision all of them, and make calls to all of them. The clicking one will blink when a call is incoming, but no audio from the call can be heard on the handset, and the caller only hears silence. The same handset works on the other Iaxys, and other handsets have the same clicking issue. Resetting the Iaxy doesn't seem to fix the problem. Does anyone have any ideas on how to fix this problem, or whether the Iaxy is broken and unfixable (for me as an end-user). Thanks, Matthew Yingling ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
