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