Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
shadowym wrote: > I hope I am not opening a can of worms here but IMHO there is > ABSOLUTELY NO REASON TO USE SCSI anymore! For sure not for this > application but most other things too. SATA is mature now, does > command queuing, and works well on 2.6 kernels. Oh, there is the > issue of cost as well. This is just not true. If you want the best performing drives out there today, you'll be using SAS (Serial Attached SCSI) or Fibre Channel. There are still 3.5" LVD SCSI drives (the old parallel style) that beat the pants off the fastest competing SATA drives because they spin at 15K RPM and have longer MTBFs. Yes, these drives cost more than SATA, but there are many reasons to use them. -- Shaw Terwilliger <[EMAIL PROTECTED]> SourceGear LLC ___ --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] VoipSupply? [Semi-Urgent] (Big Apology)
Shaw Terwilliger wrote: > If you search the archives from a few months ago you'll find a few > unhappy voipsupply customers (including me). They never shipped what I > ordered, didn't respond to any e-mail or calls. The president saw the > list traffic and sent me a long apology (stating his commitment to > service) and offered to send me an extra component that I had cancelled > the order for--free of charge--as a show of good will. > > It's been two or three months since that promise, and I never received > the part. He hasn't responded to my follow-up "did you really mean it?" > e-mail either. I must offer a HUGE apology to VoipSupply in regards to my first reply. VoipLink.com, *NOT* VoipSupply, was the company I had problems with (as described in my first message). Except for sending me some spam after I ordered from them, I have had no problems with VoipSupply. I confused the vendors as I wrote my reply, since I have ordered from both of them. Sorry for the confusion, and best wishes to the VoipSupply team. -- Shaw Terwilliger <[EMAIL PROTECTED]> SourceGear LLC signature.asc Description: OpenPGP digital signature ___ --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
Re: [asterisk-users] VoipSupply? [Semi-Urgent]
Matt wrote: > Hi, > Does anyone know what is going on with voipsupply? My sales guy > hasn't been online in several days, their 800 number is fasy busy, as > are their direct lines. And the canadian store website is down. What > the heck is going on? If you search the archives from a few months ago you'll find a few unhappy voipsupply customers (including me). They never shipped what I ordered, didn't respond to any e-mail or calls. The president saw the list traffic and sent me a long apology (stating his commitment to service) and offered to send me an extra component that I had cancelled the order for--free of charge--as a show of good will. It's been two or three months since that promise, and I never received the part. He hasn't responded to my follow-up "did you really mean it?" e-mail either. -- Shaw Terwilliger <[EMAIL PROTECTED]> SourceGear LLC signature.asc Description: OpenPGP digital signature ___ --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
Re: [asterisk-users] Warning - Voiplink.com doesn't deliver - stuck in a hole
Mr. Jones wrote: > I have had the same experience with a Grandstream order from them - 7 > days and no product. > > They even told me it was shipping Monday, but couldn't produce a > tracking number on Tuesday. I ordered a T1 card from them on July 17. No trace of it. I've sent them four e-mails, the last two asking them to simply cancel the order and give me a refund. The last e-mail "half-bounced" (I got bounces from some of the addresses the original recipient address must have forwarded to). I never got a single response from them, and they still have my money, and I still have no card (or even a tracking number). I'll initiate a charge-back via my credit card company today--it's been almost a month! Stay far away from voiplink.com. -- Shaw Terwilliger <[EMAIL PROTECTED]> SourceGear LLC signature.asc Description: OpenPGP digital signature ___ --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
Re: [asterisk-users] Clocking Multiple T1 Cards
Bruce Reeves wrote: > It will cause issues if you are using fax/modems on the channel bank and > trying to send out via the PRI. We had a great deal of problems with > timing sync between 2 spans on a Sangoma A104D until the latest beta > drivers were released. No faxes here. After reading dozens of FAX threads on asterisk-users, I've just kept a dedicated POTS line for the fax machine. -- Shaw Terwilliger <[EMAIL PROTECTED]> SourceGear LLC signature.asc Description: OpenPGP digital signature ___ --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
Re: [asterisk-users] Clocking Multiple T1 Cards
Andrew Kohlsmith wrote: > What I was trying to state was that if you have two data streams that are > solidly clocked but out of phase, you will not encounter any of these issues. > > If the clock period of either (or both) drifts then yes, you will run into > trouble. So it sounds like Asterisk can't synchronize the clocks between the Digium and Sangoma boards (or any two PCI boards), and this just may be a limitation of the T1-peripheral-on-PCI architecture. But it really shouldn't matter because of the nature of my setup: errors caused by timing mismatch between the PRI and channel banks won't cause noticeable quality issues. Do I have it right? -- Shaw Terwilliger <[EMAIL PROTECTED]> SourceGear LLC signature.asc Description: OpenPGP digital signature ___ --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
Re: [asterisk-users] Clocking Multiple T1 Cards
Andrew Kohlsmith wrote: > Correct. Since you're using an entirely different card for the incoming PRI, > you don't need to change these spans. Just add the third, and use '1' for > timing to specify that you want the clock recovered from that span to be the > primary clock source. As the other spans are on a different card, it won't > make a whit of difference to them. :-) Thank you; this is the kind of information I was looking for. The wiki and other documents told me exactly what the configuration options did, but I didn't know what kind of timing configuration was right for multiple cards. -- Shaw Terwilliger <[EMAIL PROTECTED]> SourceGear LLC signature.asc Description: OpenPGP digital signature ___ --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] Clocking Multiple T1 Cards
[ I originally sent this to the list last week but it never arrived; it may have been stuck in moderation because the sending address is not my subscribed address. I apologize if you get this twice. ] I have a Digium TE205P connected to two channel banks in my Asterisk. PBX. I will be installing a Sangoma A101 to be connected to a PSTN PRI in the same box. How should I go about configuring the T1 timing for these spans? Right now my zapata.conf contains span definitions like the following: span=1,0,0,esf,b8zs span=2,0,0,esf,b8zs If I understand the timing comments correctly, using 0 for both spans causes Asterisk to supply the timing for them, which I can't do with a PSTN PRI (right?). Should I anticipate any problems with this kind of configuration? Will I have to configure the Sangoma card in a special way, or reconfigure the Digium card? -- Shaw Terwilliger <[EMAIL PROTECTED]> SourceGear LLC signature.asc Description: OpenPGP digital signature ___ --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
Re: [Asterisk-Users] Voipsupply - my experience
On Thu, Dec 15, 2005 at 03:03:04PM -0500, Omar A. Sabek wrote: > I also have nothing but wonderful things to say about voipsupply.com, > they are first on my list for equipment needs. They have great methods > of communication and they free up my time to better serve my clients. I'm happy to hear that everyone's had good service from them, but I've felt burned by them twice. The first time, I ordered over $4000 worth of hardware from them and as soon as my order is processed, I get spammed with all their monthly specials (all in caps, of course). I sent them an e-mail asking them why they treat their customers this way, but I never received a response. Also, this order was split up and shipped at different times, but their order status pages didn't explain this (and I had no idea where my parts were for a week). I gave them another chance recently--just last week, in fact. I ordered two IAXy adaptors, so one of my co-workers could take one back to Europe. I ordered on the 6th, paid for 2 day shipping, and got an e-mail that afternoon telling me my order status was "shipped." Well, three days passed, and my co-worker flew back home without an IAXy. This week I asked voip-supply for a FedEx tracking number, and the tracking information shows the package actually shipped on the 13th, not the 6th like they told me! Their order status calculations are 7 days too optimistic. I sent them an e-mail about the status discrepancy, but I haven't received a response. I would consider their communication methods inadequate, unless you don't care when your order will arrive. -- Shaw Terwilliger <[EMAIL PROTECTED]> SourceGear LLC signature.asc Description: Digital signature ___ --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
Re: [Asterisk-Users] Calibrating both RX and TX gain?
On Fri, Oct 14, 2005 at 11:38:00AM -0600, Rich Adamson wrote: > A milliwatt generator creates an audio signal at 1,004 hz and 0db. It > has nothing at all to do with a T1 signalling, etc. You can yell into > a analog telephone set and create audio levels greater then 0 db. > > Whoever is feeding you the above words apparently has no knowledge of > telephony whatsoever. I got a similar run-around from my telco (SBC). The customer and technical service people had no idea that this type of line existed. They sent me to the tech dispatch people, who seemed to know what kind of line this was, but said, "we don't have these kind of numbers... that we give out to anybody." It's just a number that plays a tone at a reference volume when you call it. Maybe you can get the number from some area telco consultants/installers. -- Shaw Terwilliger <[EMAIL PROTECTED]> SourceGear LLC signature.asc Description: Digital signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Calibrating both RX and TX gain?
On Wed, Oct 12, 2005 at 12:05:32PM -0400, [EMAIL PROTECTED] wrote: > I am in the middle of trying to get a milliwatt test line to calibrate the > rxgain properly. However, this won't help me with the txgain, will it? > How can I properly calibrate the txgain? By ear? Or is there a more > scientific method? Maybe I can help. I had a similar problem. I was using a Digium TE205P card and two Rhino channel banks, and every call that was bridged from a phone on an FXS interface to a PSTN line on an FXO interface was (1) loud and (2) had an echo with a tiny delay (maybe 30ms). The echo sounded almost like excess sidetone, but was delayed enough to phase shift the speech and make things sound hollow. I could verify that what was being transmitted was coming back on the RX channel of the PSTN interface (using ztmonitor). I'm using Nortel analog, wall-powered phones (pretty nice models). I had echo cancellation on, and had tried all possible configuration settings for taps, etc. Nothing killed my echo. I had tried adjusting all the gains down in Asterisk for all the interfaces, but that didn't work. I contacted Rhino to see if they had any suggestions, and they were able to give me a few. What finally worked was setting the Asterisk gains back to 0 for all channels, then adjusting the gains down on the channel banks themselves for the phone (FXS) interfaces only. A huge improvement! My current adjustements are the following: On the Rhino channel banks: For FXS (phones) interfaces: rx -4 dB tx -4 dB For FXO (PSTN lines) interfaces: rx 0 dB (default) tx 0 dB (default) In Asterisk's zaptel.conf: context=phones rxgain=3.0; This is to compensate for the drop in volume because of ; the -4 dB setting on the channel bank for rx. txgain=3.0; This is to compensate for the drop in volume because of ; the -4 dB setting on the channel bank for tx. context=pstn rxgain=1.4; This was bumped up last, as a result of a milliwatt test. ; txgain=1.4; This was also bumped up, because it makes the outbound ; calls a bit louder, and doesn't seem to overdrive the ; line. I figure the gain loss on rx (which was calibrated ; with the milliwatt test) should be similar to tx gain lost, ; although I couldn't directly test this. Now, when I turn on echo cancellation for all my interfaces, the echo is completely gone. After compensating for the gain drop on the channel banks with asterisk boost, the call volumes sound good too. -- Shaw Terwilliger <[EMAIL PROTECTED]> SourceGear LLC signature.asc Description: Digital signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Auto-assign CallerID for all my FXS Interfaces
As far as I can tell, in order to have caller ID show up for calls from other internal phones, I have to set the caller ID on each channel in zapata.conf. This is tedious, and redundant (since Asterisk knows which extension is making the call, and it could look up the name from the voicemail configuration--if the extension matched the mailbox). Is there a way to set the caller ID for these calls from the voicemail information so I don't have to duplicate all the names? Is there some variable or function that will return a voicemail name for an extension? Would this functionality be pretty trivial to implement? Also, the stutter dialtone mapping to mailboxes requires the same duplication. However, I can't see how Asterisk would know that a zap channel "belongs to" a single extension, so I can understand this requirement. -- Shaw Terwilliger <[EMAIL PROTECTED]> SourceGear LLC signature.asc Description: Digital signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Tiny Echo on PSTN via Zaptel
On Thu, Sep 29, 2005 at 04:22:39PM -0500, Neil Lewis wrote: > FXOTune is only for TDM cards. I suggest you try one of the other > methods listed. I've tried them all so far. What I'm hearing sounds a lot like (too loud) sidetone, but there's just the smallest bit of delay. It's only noticeable when I'm bridged to a PSTN line. If I plug a phone directly into one of those PSTN lines, there is no problem. Changing the echo settings has improved it a bit, so it probably is some sort of echo. -- Shaw Terwilliger <[EMAIL PROTECTED]> SourceGear LLC signature.asc Description: Digital signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Tiny Echo on PSTN via Zaptel
On Thu, Sep 29, 2005 at 12:10:47PM -0500, Neil Lewis wrote: > If you are using CVS-Head you can also use the fxotune utility located > in /usr/src/zaptel to tune > out echo in the FXO module. Execute this command: './fxotune -i 4' > It will automatically configure the FXO modules for echo. Is fxotune supposed to work with the TE210P? When I run it, it outputs the following for all of my FXO (and FXS) ports: Tuning module 1Skipping non-TDM / non-FXO Failure! > You can also try using an different Echo Canceler. We've recently added > a new EchoCan to Asterisk: KB1. To utilize it, just uncomment its > define statement in zconfig.h, and comment the other EchoCan out. I'll give this a shot, then play with the agressive suppressor. The echo situation is better than it used to be, so I'm making progress. -- Shaw Terwilliger <[EMAIL PROTECTED]> SourceGear LLC signature.asc Description: Digital signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Tiny Echo on PSTN via Zaptel
On Thu, Sep 29, 2005 at 08:33:03AM -0400, Tony Nichols wrote: > I have had problems between the sip/FXO lies and was able to "kill" the echo > by trying different combinations of the echocancel line to 64 (I think it > has settings in 32 bit increments) > Just kept trying different ones till it went away. Here is my config: I think I finally found the problem (but I'm not sure). The rxgain on my phone ports was set to -8. When I turned them back up to 0.0 (to boost the outgoing signal on the POTS lines when a call was going on), the echo canceller seems to work pretty well. I've been using ztmonitor to verify this: previously the echo was 20-40% of the original signal. Now it's hardly ever there. I thought I had my gains set correctly, but it seems like louder is better (except for distortingly loud) for the echo canceller. -- Shaw Terwilliger <[EMAIL PROTECTED]> SourceGear LLC signature.asc Description: Digital signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Tiny Echo on PSTN via Zaptel
I'm using Asterisk 1.0.9, a Digium TE210P dual T1 card, and two Rhino channel banks (one 12FXO/12FXS, the other 24 FXS). So it's an analog phone on the inside connected to one of the FXS ports, and PSTN line connected to one of the FXO ports. My problem is that as soon as I hear the _first_ ring when I dial out through the PSTN line, I hear a tiny echo on the phone (I estimate between 20ms and 40ms), which never goes away for this call. It's just loud enough to bug the heck out of me when I'm talking (I could estimate the gain relationship with ztmonitor if it would help). The sound on the recipient end of the connection is perfect. If I make a call from the phone to the another internal extension (another phone on an FXS port), there is no echo. If I call into Comedian mail, there is no echo. I've checked all my gains. The "internal" gains were a bit loud to start with because of the powered phones, but now they all fall comfortably within ztmonitor's dynamic range display. The PSTN line is pretty good at tx 0 and rx 0, so I left it. I've tried turning them down, but that didn't kill the echo. My zapata.conf includes these lines at the bottom: echocancel=yes echocancelwhenbridged=no echotraining=yes context=companyA-pstn txgain=0.0 rxgain=0.0 signalling=fxs_ks group=1 channel=1-7 context=companyB-pstn txgain=0.0 rxgain=0.0 signalling=fxs_ks group=2 channel=11-12 context=internal txgain=-12.0 rxgain=-8.0 signalling=fxo_ls callerid=asreceived group=3 channel=13-48 When the calls are connected, I can use "zap show channel 11" and verify that the echo cancellation is ON. But I can still hear one. I've also tried echocancelwhenbridged=yes, but it didn't make any difference. -- Shaw Terwilliger <[EMAIL PROTECTED]> SourceGear LLC signature.asc Description: Digital signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Re: ztcfg Kills My Dial Tone [solved]
On Wed, Sep 07, 2005 at 11:50:28AM -0500, Shaw Terwilliger wrote: > Right after I reboot, and modprobe wct4xxp, my analog phone connected > to port 13 of the first channel bank (first FXS port) gets a dial tone. > Asterisk is not running yet, and I have NOT run ztcfg. I've solved my own problem, with a little help from Rhino (and some poking around). I think my problem was that an older version of the wct4xxp module was sitting around in the kernel's module dirctory, so I had an inconsistency between the userspace tools and the kernel drivers. Also, I had my signalling misconfigured the first time I brought up the channel banks, and even though I had corrected this in zaptel.conf, I had to tell the channel banks to "Auto T1" themselves. Now I get nice, strong dial tones on my FXS interfaces, and my FXO interfaces happily listen to POTS lines. -- Shaw Terwilliger <[EMAIL PROTECTED]> SourceGear LLC signature.asc Description: Digital signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] ztcfg Kills My Dial Tone
startup! Sep 7 11:46:22 localhost kernel: Putting 0 in register 2f on span 1 Sep 7 11:46:22 localhost kernel: Putting 0 in register 30 on span 1 Sep 7 11:46:22 localhost kernel: Putting 0 in register 31 on span 1 Sep 7 11:46:22 localhost kernel: Unassigning channel 0/1! [it's all the same for the other channels up to 1/23] Sep 7 11:46:22 localhost kernel: Putting 0 in register 2f on span 2 Sep 7 11:46:22 localhost kernel: Putting 0 in register 30 on span 2 Sep 7 11:46:22 localhost kernel: Putting 0 in register 31 on span 2 Sep 7 11:46:22 localhost kernel: Unassigning channel 1/24! Sep 7 11:46:22 localhost kernel: Registered tone zone 0 (United States / North America) Sep 7 11:46:27 localhost kernel: wct2xxp: Clearing yellow alarm on span 1 Sep 7 11:46:27 localhost kernel: wct2xxp: Clearing yellow alarm on span 2 Sep 7 11:46:28 localhost kernel: Zaptel: Master changed to TE2/0/1 -- Shaw Terwilliger <[EMAIL PROTECTED]> SourceGear LLC signature.asc Description: Digital signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
