You're HDLC error is evident of timing slips.<div><br></div><div>Use "cat /proc/dahdi/1" or 2 or 3</div><div>Also "cat /proc /interrupts" </div><div><br></div><div>--</div><div>Vincent Swart<br><br>
<div class="gmail_quote">On Mon, Nov 5, 2012 at 8:00 PM, <span dir="ltr"><<a href="mailto:asterisk-users-request@lists.digium.com" target="_blank">asterisk-users-request@lists.digium.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
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Today's Topics:<br>
<br>
1. Re: Asterisk Support from Digium (Danny Dias)<br>
2. Re: Asterisk Support from Digium (Chris Bagnall)<br>
3. Re: PRI got event HDLC Abort (Edwin Lam)<br>
4. Re: PRI got event HDLC Abort (Thorsten G?llner)<br>
5. play wav file (Jerry Geis)<br>
6. Re: play wav file (Danny Nicholas)<br>
7. Re: play wav file (Christopher Harrington)<br>
<br>
<br>
----------------------------------------------------------------------<br>
<br>
Message: 1<br>
Date: Sun, 4 Nov 2012 21:37:27 +0100<br>
From: Danny Dias <<a href="mailto:ing.diasdanny@gmail.com">ing.diasdanny@gmail.com</a>><br>
Subject: Re: [asterisk-users] Asterisk Support from Digium<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Message-ID:<br>
<<a href="mailto:CA%2Bd0Ut_xh_BH3g2Mk1K8AnQgHbcS3trO94cn3f%2BtLT0ie6jbbA@mail.gmail.com">CA+d0Ut_xh_BH3g2Mk1K8AnQgHbcS3trO94cn3f+tLT0ie6jbbA@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="iso-8859-1"<br>
<br>
Thanks Andrew,<br>
<br>
But i'm quite confuse with the following:<br>
<br>
*Q: Does Digium offer SLA guaranteed support for Asterisk?*<br>
*A:* Yes. Digium offers SLA guaranteed support, to SLA-entitled customers,<br>
for the Certified Asterisk branches. Digium does not offer SLA guaranteed<br>
support for other branches or releases.<br>
<br>
Just for Certify Versions of Asterisk? What does SLA means "exactly"?<br>
<br>
For example, if i install a FreePBX/Elastix (i'm not a good friend of these<br>
systems, but customers always ask for a web interface for management) to a<br>
customer, can i buy support from Digium for the Asterisk Release used? It<br>
would be nice to now the scope and limits of this support<br>
<br>
Thanks<br>
<br>
<br>
<br>
2012/11/3 Andrew Latham <<a href="mailto:lathama@gmail.com">lathama@gmail.com</a>><br>
<br>
> On Sat, Nov 3, 2012 at 2:16 PM, Danny Dias <<a href="mailto:ing.diasdanny@gmail.com">ing.diasdanny@gmail.com</a>><br>
> wrote:<br>
> > Hello,<br>
> ><br>
> > I wonder if Digium provides support for Asterisk OpenSource versions as<br>
> an<br>
> > anual fee or something?<br>
> ><br>
> > For example, if i download Asterisk 1.8.X (Certified or not...) can i buy<br>
> > support from Digium to maintain and help on possible future problems in<br>
> my<br>
> > configuration?<br>
> ><br>
> > Thanks<br>
><br>
> Yes<br>
><br>
> Please review<br>
> <a href="http://www.digium.com/en/supportcenter/custom-communications-solutions/" target="_blank">http://www.digium.com/en/supportcenter/custom-communications-solutions/</a><br>
> for more information.<br>
><br>
><br>
> --<br>
> ~ Andrew "lathama" Latham <a href="mailto:lathama@gmail.com">lathama@gmail.com</a> <a href="http://lathama.net" target="_blank">http://lathama.net</a> ~<br>
><br>
> --<br>
> _____________________________________________________________________<br>
> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
> New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
><br>
> asterisk-users mailing list<br>
> To UNSUBSCRIBE or update options visit:<br>
> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
><br>
<br>
<br>
<br>
--<br>
*SIP:* <a href="mailto:danny@voice.danntel.net">danny@voice.danntel.net</a> <<a href="http://www.danntel.net/?page_id=189" target="_blank">http://www.danntel.net/?page_id=189</a>><br>
*Web: *<a href="http://www.danntel.net" target="_blank">http://www.danntel.net</a><br>
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<br>
Message: 2<br>
Date: Sun, 04 Nov 2012 22:33:39 +0000<br>
From: Chris Bagnall <<a href="mailto:asterisk@lists.minotaur.cc">asterisk@lists.minotaur.cc</a>><br>
Subject: Re: [asterisk-users] Asterisk Support from Digium<br>
To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
Message-ID: <<a href="mailto:5096ED43.5060804@lists.minotaur.cc">5096ED43.5060804@lists.minotaur.cc</a>><br>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br>
<br>
On 4/11/12 8:37 pm, Danny Dias wrote:<br>
> For example, if i install a FreePBX/Elastix<br>
<br>
I'd be very surprised (no, actually, I'd be *amazed*) if Digium were<br>
prepared to provide support on a product from a third party, which is<br>
what FreePBX and Elastix effectively are.<br>
<br>
Kind regards,<br>
<br>
Chris<br>
--<br>
This email is made from 100% recycled electrons<br>
<br>
<br>
<br>
------------------------------<br>
<br>
Message: 3<br>
Date: Sun, 04 Nov 2012 21:13:35 -0800<br>
From: Edwin Lam <<a href="mailto:edwin.lam@officegeneral.com">edwin.lam@officegeneral.com</a>><br>
Subject: Re: [asterisk-users] PRI got event HDLC Abort<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Message-ID: <<a href="mailto:50974AFF.1010507@officegeneral.com">50974AFF.1010507@officegeneral.com</a>><br>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br>
<br>
On 11/2/2012 10:06 PM, Liban Abdi wrote:<br>
> is there static on the line??<br>
<br>
no. there were customer complains about sound cutting in and out.<br>
however i wasn't noticing and bad sound quality when i was testing it.<br>
<br>
> is there timing slips and crc4 errors?<br>
<br>
no. the only messages i have are the HDLC abort warning.<br>
<br>
> are they increasing throughout the day?<br>
<br>
they happen randomly, and quite frequently.<br>
<br>
> are you getting timing slips during the day when users are using the phones and<br>
> not off-peak hours?<br>
<br>
no timing slips related messages in either Asterisk's logs<br>
or syslog.<br>
<br>
> are you getting hdlc abort erros when you hear a static noises??<br>
<br>
that i don't know. however there was once it happened<br>
while i was in the middle of a call but i couldn't hear<br>
any sound drop off or any static.<br>
<br>
> is the card sharing irq?<br>
<br>
no. this the only card that uses IRQ 30<br>
1b:00.0 Network controller: Digium, Inc. Device 1420 (rev 14)<br>
Subsystem: Device 0005:0000<br>
Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr+<br>
Stepping- SERR+ FastB2B- DisINTx-<br>
Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort-<br>
<MAbort- >SERR- <PERR- INTx-<br>
Latency: 64 (8000ns min, 32000ns max), Cache Line Size: 64 bytes<br>
Interrupt: pin A routed to IRQ 30<br>
Region 0: Memory at 97a00000 (32-bit, non-prefetchable) [size=32K]<br>
Kernel driver in use: wct4xxp<br>
<br>
> is your system plugged directly into an outlet without ups?<br>
<br>
good question. i don't know.<br>
<br>
><br>
> On Fri, Nov 2, 2012 at 8:40 PM, Edwin Lam <<a href="mailto:edwin.lam@officegeneral.com">edwin.lam@officegeneral.com</a><br>
> <mailto:<a href="mailto:edwin.lam@officegeneral.com">edwin.lam@officegeneral.com</a>>> wrote:<br>
><br>
> hi folks.<br>
><br>
> recently some of our customers complained about bad voice<br>
> quality on the phone system. i looked at the logs and found<br>
> a lot of these:<br>
><br>
> [2012-11-03 08:26:38] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort<br>
> (6) on D-channel of span 1<br>
> [2012-11-03 08:26:45] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort<br>
> (6) on D-channel of span 1<br>
> [2012-11-03 08:26:54] NOTICE[11305] chan_dahdi.c: PRI got event: HDLC Abort<br>
> (6) on D-channel of span 1<br>
><br>
> i upgraded Asterisk/dahdi/libpri. tried turn on/off echo canceller etc.<br>
> nothing seems to help. call the phone company to check out the line<br>
> (which they said it's working fine)<br>
><br>
> any idea? do i have a hardware issue here? i've check syslog<br>
> there was no dahdi errors.<br>
><br>
> here's my system.conf:<br>
> span=1,1,0,esf,b8zs<br>
> bchan=1-23<br>
> dchan=24<br>
> span=2,0,0,esf,b8zs<br>
> bchan=25-47<br>
> dchan=48<br>
> span=3,0,0,esf,b8zs<br>
> bchan=49-71<br>
> dchan=72<br>
> span=4,0,0,esf,b8zs<br>
> bchan=73-95<br>
> dchan=96<br>
><br>
> and here's my chan_dahdi.conf:<br>
> [channels]<br>
> switchtype=national<br>
> pridialplan=unknown<br>
> prilocaldialplan=unknown<br>
> internationalprefix = 001<br>
> nationalprefix =<br>
> unknownprefix =<br>
> signalling=pri_cpe<br>
> usecallerid=yes<br>
> usecallingpres=yes<br>
> echocancel=no<br>
> echocancelwhenbridged=no<br>
> group=1<br>
> callgroup=1<br>
> pickupgroup=1<br>
> faxdetect=incoming<br>
> context=defaultspan1<br>
> channel => 1-23<br>
><br>
<br>
<br>
<br>
<br>
------------------------------<br>
<br>
Message: 4<br>
Date: Mon, 05 Nov 2012 14:06:20 +0100<br>
From: Thorsten G?llner <<a href="mailto:tg@ovm-group.com">tg@ovm-group.com</a>><br>
Subject: Re: [asterisk-users] PRI got event HDLC Abort<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Message-ID: <<a href="mailto:5097B9CC.2060609@ovm-group.com">5097B9CC.2060609@ovm-group.com</a>><br>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br>
<br>
<br>
><br>
>> is the card sharing irq?<br>
><br>
> no. this the only card that uses IRQ 30<br>
> 1b:00.0 Network controller: Digium, Inc. Device 1420 (rev 14)<br>
> Subsystem: Device 0005:0000<br>
> Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop-<br>
> ParErr+ Stepping- SERR+ FastB2B- DisINTx-<br>
> Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort-<br>
> <TAbort- <MAbort- >SERR- <PERR- INTx-<br>
> Latency: 64 (8000ns min, 32000ns max), Cache Line Size: 64 bytes<br>
> Interrupt: pin A routed to IRQ 30<br>
> Region 0: Memory at 97a00000 (32-bit, non-prefetchable)<br>
> [size=32K]<br>
> Kernel driver in use: wct4xxp<br>
><br>
>> is your system plugged directly into an outlet without ups?<br>
<br>
Please give us a complete "lspci -vvv".<br>
<br>
Did you read this?<br>
<a href="http://alexrrr.blogspot.de/2007/10/solving-asterisks-hdlc-abort-issue.html" target="_blank">http://alexrrr.blogspot.de/2007/10/solving-asterisks-hdlc-abort-issue.html</a><br>
<br>
<br>
<br>
------------------------------<br>
<br>
Message: 5<br>
Date: Mon, 05 Nov 2012 11:52:14 -0500<br>
From: Jerry Geis <<a href="mailto:geisj@pagestation.com">geisj@pagestation.com</a>><br>
Subject: [asterisk-users] play wav file<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Message-ID: <<a href="mailto:5097EEBE.6040204@pagestation.com">5097EEBE.6040204@pagestation.com</a>><br>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br>
<br>
I have an mp3 that is 128K, 44.1K stereo.<br>
I convert that to wave 16 bit, stereo, 44.1K<br>
<br>
The "sound" alike at this time.<br>
<br>
I want to play them (not just over my sound port) but through asterisk<br>
on select devices/machines that are also running asterisk over the<br>
Console/dsp.<br>
<br>
I converted the wave file to 8K, mono and it doesn't sound very good, I<br>
am also<br>
using 1.4.43 and ulaw,alaw,gsm allowed.<br>
<br>
What format will give me the best sounding output and how do I get that?<br>
Do I need somethink like g722?<br>
<br>
Thanks,<br>
<br>
Jerry<br>
<br>
<br>
<br>
<br>
------------------------------<br>
<br>
Message: 6<br>
Date: Mon, 5 Nov 2012 11:03:27 -0600<br>
From: "Danny Nicholas" <<a href="mailto:danny@debsinc.com">danny@debsinc.com</a>><br>
Subject: Re: [asterisk-users] play wav file<br>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Message-ID: <00e801cdbb77$79c701c0$6d550540$@<a href="http://debsinc.com" target="_blank">debsinc.com</a>><br>
Content-Type: text/plain; charset="us-ascii"<br>
<br>
If you're going to stay with 1.4.X probably g722 would be best for you. If<br>
you work a while with SOX, you should end up with 8K files that sound<br>
"almost as good" as the 44K wav files.<br>
<br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Jerry Geis<br>
Sent: Monday, November 05, 2012 10:52 AM<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
Subject: [asterisk-users] play wav file<br>
<br>
I have an mp3 that is 128K, 44.1K stereo.<br>
I convert that to wave 16 bit, stereo, 44.1K<br>
<br>
The "sound" alike at this time.<br>
<br>
I want to play them (not just over my sound port) but through asterisk on<br>
select devices/machines that are also running asterisk over the Console/dsp.<br>
<br>
I converted the wave file to 8K, mono and it doesn't sound very good, I am<br>
also using 1.4.43 and ulaw,alaw,gsm allowed.<br>
<br>
What format will give me the best sounding output and how do I get that?<br>
Do I need somethink like g722?<br>
<br>
Thanks,<br>
<br>
Jerry<br>
<br>
<br>
--<br>
_____________________________________________________________________<br>
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To UNSUBSCRIBE or update options visit:<br>
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<br>
<br>
<br>
------------------------------<br>
<br>
Message: 7<br>
Date: Mon, 5 Nov 2012 11:04:36 -0600<br>
From: Christopher Harrington <<a href="mailto:chris@acsdi.com">chris@acsdi.com</a>><br>
Subject: Re: [asterisk-users] play wav file<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
Message-ID:<br>
<<a href="mailto:CAJLBXEkHmmUFGN9snuYCtt8BXOhWXCqqqoCASWfCQ7FQj1UaOw@mail.gmail.com">CAJLBXEkHmmUFGN9snuYCtt8BXOhWXCqqqoCASWfCQ7FQj1UaOw@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="utf-8"<br>
<br>
On Mon, Nov 5, 2012 at 10:52 AM, Jerry Geis <<a href="mailto:geisj@pagestation.com">geisj@pagestation.com</a>> wrote:<br>
<br>
> I converted the wave file to 8K, mono and it doesn't sound very good, I am<br>
> also<br>
> using 1.4.43 and ulaw,alaw,gsm allowed.<br>
><br>
><br>
This has been covered just recently, try searching for "mp3" on the mailing<br>
list.<br>
<br>
What format will give me the best sounding output and how do I get that?<br>
> Do I need somethink like g722?<br>
><br>
><br>
Keep in mind that you are going to be using codecs and hardware that are<br>
optimized for speech, so anything that isn't speech is not going to sound<br>
good. In that case, "best" is really going to depend on what the content is<br>
and will probably require you to simply test all of the permutations and<br>
find the one that sounds the "least bad".<br>
<br>
--<br>
-Chris Harrington<br>
ACSDi Office: 763.559.5800<br>
Mobile Phone: 612.326.4248<br>
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