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-requ...@lists.digium.com>wrote: > Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > asterisk-users-requ...@lists.digium.com > > You can reach the person managing the list at > asterisk-users-ow...@lists.digium.com > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-users digest..." > > > 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.diasda...@gmail.com> > Subject: Re: [asterisk-users] Asterisk Support from Digium > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: > < > ca+d0ut_xh_bh3g2mk1k8anqghbcs3tro94cn3f+tlt0ie6j...@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 <lath...@gmail.com> > > > On Sat, Nov 3, 2012 at 2:16 PM, Danny Dias <ing.diasda...@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 lath...@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:* da...@voice.danntel.net <http://www.danntel.net/?page_id=189> > *Web: *http://www.danntel.net > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.digium.com/pipermail/asterisk-users/attachments/20121104/dacdf299/attachment-0001.htm > > > > ------------------------------ > > Message: 2 > Date: Sun, 04 Nov 2012 22:33:39 +0000 > From: Chris Bagnall <aster...@lists.minotaur.cc> > Subject: Re: [asterisk-users] Asterisk Support from Digium > To: asterisk-users@lists.digium.com > Message-ID: <5096ed43.5060...@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....@officegeneral.com> > Subject: Re: [asterisk-users] PRI got event HDLC Abort > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <50974aff.1010...@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....@officegeneral.com > > <mailto:edwin....@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 <t...@ovm-group.com> > Subject: Re: [asterisk-users] PRI got event HDLC Abort > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <5097b9cc.2060...@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 <ge...@pagestation.com> > Subject: [asterisk-users] play wav file > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <5097eebe.6040...@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" <da...@debsinc.com> > Subject: Re: [asterisk-users] play wav file > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@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-boun...@lists.digium.com > [mailto:asterisk-users-boun...@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 <ch...@acsdi.com> > Subject: Re: [asterisk-users] play wav file > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: > < > cajlbxekhmmufgn9snuyctt8bxohwxcqqqocaswfcq7fqj1u...@mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > On Mon, Nov 5, 2012 at 10:52 AM, Jerry Geis <ge...@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 > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.digium.com/pipermail/asterisk-users/attachments/20121105/a35675f5/attachment-0001.htm > > > > ------------------------------ > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > AstriCon 2010 - October 26-28 Washington, DC > Register Now: http://www.astricon.net/ > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > End of asterisk-users Digest, Vol 100, Issue 6 > **********************************************
-- _____________________________________________________________________ -- 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