Re: [Asterisk-Users] sending a DTMF tone before hangup
On March 13, 2005 09:57 am, Nigel Burgess wrote: [door] exten = s,1,Dial (SIP31,15) exten = s,2,Playtones(dtmf) However the call hangsup before trying to play the DTMF tone. Make sure you use the 'g' flag in the Dial command to go on in the context after a hangup. Now whether the tone will be played or not is still a question, as I'm not sure when SIP declares the connection closed, but it's worth a try. -A. ___ 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] Problem with TE405P and Slackware 10.0
On March 14, 2005 06:50 am, pixer wrote: 3: 0 XT-PIC t4xxp Without loading the module the LED glows in red colour, but the moment we load module, it goes off. (No red or green). We ran zttool and tried to run a loop test, but zttool simply hung with the message 'Looping UP Span 1...'. We had to terminate zttool with 'kill'. What's wrong ? This is a hardware or BIOS issue -- your card is unable to generate interrupts. Try shuffling the card around to a different PCI slot and/or adjusting your BIOS interrupt settings. Also you might want to try the pci=noacpi or even noapic kernel options. I run the same card in a similar box with Slackware 10.0 (and formerly 9.1) without any issue whatsoever. This is a specific hardware issue. -A. ___ 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] E1/T1 back to back ??
On March 14, 2005 06:43 am, Brett, Gary wrote: Just a quick question, I will be building some servers in a lab utilizing Digium E1 cards. I would like if possible to avoid the expense of installing an e1/ISDN30 in my lab. I have two questions really, first does anybody know of an effective simulation tool I can use to replicate a real world PRI but without the telco line being installed. And secondly, can I have a scenario with 2 asterisk servers with digium e1 cards 'back to back' one configured as the network side and the other configured as the client side (can I just use a single cat5 straight through cable between them ?? and cant the Digium e1 cards operate ok in both modes?) A standard Cat5 ethernet cable won't work, but a T1/E1 crossover cable made from Cat5 should work just fine. I do this all the time with T1/PRI, I don't see why it wouldn't work with E1/PRI. One side is set up as pri_cpe, and hte other as pri_net. -A. ___ 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] Skype - Bandwidth
On March 14, 2005 04:20 pm, Eric Wieling wrote: Skype does not interface with Asterisk in any way whatsoever. You could just as well have asked if someone knows what RNA sequence 42 in the turnip genome is for. About as many people on this list would be familiar with that as would Skype. Don't be silly. RNA doesn't exist in turnips, RNA only exists in animal cells. :-) -A. ___ 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] OT: Best DB
On March 15, 2005 01:00 pm, Giudice, Salvatore wrote: MySQL: Speed, Power and Precision Now *that* is funny. Thank you for bringing some humour to the list. Now take the rest of this email and file it under FUD and exaggeration on MySQL's capabilities, especially the benchmarks. -A. ___ 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] OT: Best DB
On March 15, 2005 02:21 pm, Giudice, Salvatore wrote: Sticks and stone still break my bones, but PostgreSQL is still a dog. Until you actually show some benchmarks where the tests are clearly documented and Postgres is properly tuned, you're spreading FUD. Your testing should also demonstrate real world performance (hundreds of connections, complex queries, etc.) or it's just marketing fluff, which is exactly what your links, including this one on market share are. It's been stated time and time again that Postgres' default values are *very* conservative. There's a reason that most people who actually try Postgres after years of using MySQL continue using Postgres, and it isn't because Postgres is a dog, as you state. -A. ___ 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] OT: Best DB
On March 15, 2005 06:04 pm, Giudice, Salvatore wrote: commercial licensing AND has a real enterprise class support structure behind it, or are you going to run with PostgreSQL (bow wow) distributed under a BSD license with some mom and pop support shops and some mailing It's time to put up or shut up. Can you please give supporting evidence that MySQL AG has no more oomph in commercial support than companies like Command Prompt, Fujitsu, Red Hat, or even PostgreSQL, Inc.? Every single one of those organizations has commercial support available for PostgreSQL. I'm genuinely curious if you consider MySQL AG more of a company than Red Hat or Fujitsu. Seriously. You're frothing at the mouth and tripping over yourself trying to make your point, and you're so far off base to begin with that you couldn't possibly be more wrong. As far as your benchmark points go, until you can show me properly organized and open benchmarks, your point is totally invalid. In my cursory check (hint: try locating the open database bake-off from a couple years ago, phpbuilder's evaluation a few years back, http://benchw.sourceforge.net, or locate anything done by independent testing groups) it appears that under real-world load, Postgres trounces MySQL handily and can handle FAR more concurrent connections than even a tuned-out MySQL server can handle. Yes, Postgres needs some tuning out of the box, this has been hashed over repeatedly and nobody's denying it. Yes, MySQL is fast for the simplest queries and inserts. And my personal favourite, Yes, MySQL will take artistic license with your data. These are all facts that everyone (MySQL AG included) but you seems to be able to agree upon. The only benchmarks you'll speak of are those found with mysql-bench, but those results are generally held as a practical joke with zero relevance in real-world applications. Your comment on licensing is also interesting. I wonder, do you also have problems with Apache because it too is released under a BSD license? How about the BSD Unixes themselves? How is BSD less good than GPL? Honestly I'd love to know! Hey, it's your choice. Do you want to eat American Grade A American beef or that strange meat flavored tofu? As long as it meets your needs, choose whatever you have the ability to handle. Exactly my point. This is *exactly* why I run PostgreSQL over MySQL. At any rate I've participated in this offtopic thread enough. Unless you post some practical examples to back up your points I will let you have the last word. The list archives will no doubt commemorate this particular thread. :-) -A. ___ 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] Re: Problem with TE405P and Slackware 10.0
On March 16, 2005 05:57 am, pixer wrote: I have following your advice and I have put this into /etc/lilo.conf append = pci=noacpi 20: 0 IO-APIC-level t4xxp modules (COM port, serial ports, etc), and shuffling the card around to a different PCI slot, but unfortunately he does yet not work equally :/ Can you put this card in a totally separate machine with your slackware HDD just to see if it comes up properly in another machine? This is very unusual. -A. ___ 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] Problem with TE405P and Slackware 10.0 (reply this)
On March 16, 2005 07:12 am, pixer wrote: Unfortunately I have already also tried this, without results. I do not know what to do any more.. Was it an entirely different motherboard (different manufacturer)? If so, it's time to call Digium and open a ticket. It sounds like the card is DOA. They will likely want you to go through all these same steps, but be patient. -A. ___ 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] OT: Best DB
On March 18, 2005 07:08 pm, Mike Sander wrote: But Budwieser tastes like water to most Australian beer drinkers. No, it tastes like piss to pretty much everyone. They just have a great marketing budget. -A. ___ 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] who has purchased a V400 card from Varion ?
On March 20, 2005 07:52 pm, [EMAIL PROTECTED] wrote: who has purchased a V400 card from Varion ? I need some help . please help me . Does Varion not provide any support for their products? I'm interested to know why you chose them over Digium... -A. ___ 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] why even use SIP
On March 21, 2005 01:07 pm, Kevin P. Fleming wrote: Well, let's see.. 99.99% of the available VOIP hardware only support SIP, MGCP and H.323, but not IAX2. Is that a good reason? No. 95% of the marketplaces uses Windows. Drive the marketplace to use better protocols. IAX2 calls between servers carry the signaling and media in the same connection, which is good for NAT issues, but bad for CDR and traffic control issues. SIP handles them separately, so you can keep complete CDR without forcing the media to follow the same path. Is that a good reason? Yes, but the dynamic port allocation and ABSOLUTELY INSANE WORDINESS and WASTAGE of the SIP control protocol is reason enough for me to never want to support it. While perhaps not worth much on my own, I am voting with my wallet and my feet. I will not support SIP, nor will I purchase products or services which require it. -A. ___ 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] Major problems with TDM400 and specific telephones: suggestions?
On March 22, 2005 03:08 pm, [EMAIL PROTECTED] wrote: The phone in question is what I would consider to be a good-quality GE two-line cordless telephone. Digium's guess is that it is putting power on the telephone line and the card doesn't like that. They have given me zero solution other than to use a different telephone. I have a Panasonic 900MHz digital cordless phone that also causes the TDM card to have fits. I've sent it to Digium to try and figure out what's going on, as every single other phone and fax (probably two dozen brands between the two) I have ever hooked up has worked just fine. This is not a normal thing and it may just be that the actual POTS system is able to handle their particular brand of yuck. I certainly don't blame Digium for this, but they have been more than willing to help me correct it, especially since I am willing to get the phone to them to test with since they seem to be unable to recreate it in their lab. My 5.whateverGHz Panasonic digital cordless phone works great, and my 900MHz non-digital (cheapass) cordless phone works great. As an aside, why is it that just about *any* other device with an analog interface you can buy today more robust than the TDM cards? I've used countless different ISDN NT-1's without problems, from $100 cheapo models to $1000 high-end devices and tons in between and none have had problems like this. Now there's a ton of SIP gateway devices. They don't seem to have these issues. Why do the TDM cards? And most importantly, can an end user do anything about this? As I said, I've hooked up countless devices to the TDM cards and this particuar phone is the ONLY one I've had trouble with. It is perhaps a corner case in the TDM design, but as I said Mark has personally been more than willing to help fix this. As an electronics designer myself, I know how unbelievably frustrating it is to have a customer with an issue and not be able to recreate it myself such that a fix can be found. -A. ___ 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] ADIT 600 Dynamic Impedance matching
On March 23, 2005 08:25 am, Matt Schulte wrote: Has anyone ever heard of this so called Dynamic Impedance matching on the ADIT 600? I called their support and they've never heard of it. We That's odd, I have always had excellent support from CAC. And FWIW I've never had echo problems with their channel banks. Ever. I have echocancel turned off in the Zapata driver. The only clue to the dynamic impedance is that the 5g and ver8 of the FXS cards can hardcode the impedance according to country. Well that's fine and dandy but so can a Rhino CB-24 in the rating of milliamps.. You don't tune impedance in milliAmps. That's a current measurement. The Rhino can probably alter the amount of current it can source and this is what they're talking about. Not having used Rhino's stuff, I can't say for certain, but you simply don't alter impedance by changing mA. (yes, IAAEE). Does anyone have suggestions regarding these issues? Please hold back the flaming comments. I'm not here to flame, but to resolve and very tiring issue. :-) You can start by giving us a connection diagram between the Adit600 and whatever you're hooked up to, including grade of cable, how long it is, what it's terminating to (make and model) and whether you've tried replacing some runs with other cable to test. Invariably my Adit600 analogue runs are always under 50 feet since I'm terminating to a PBX or KSU nearby. These devices are able to terminate very long (km) runs, so I am curious as to why you're having such issues. Do you have the gains on the Adit600 or Zapata turned way up? -A. ___ 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] Digium support quality: Excellent
On March 23, 2005 07:37 am, Michael George wrote: I was told to try changing PCI slots (I haven't had a chance to do that yet), but since the TDM cannot share IRQs with anything else, changing slots might just put it into a conflict situation. This one could be sticky... As I am learning more and more of the zaptel code I think the *right* solution is to have the driver recognize that it already has a zaptel timing source and turn off the timer on subsequent cards, using the first card detected as the sole generator of interrupts. I've got a few other things on my plate, however, so I haven't been able to really test this. -A. ___ 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] who has purchased a V400 card from Varion ?
On March 23, 2005 08:05 am, Ernest Stokes wrote: We are a small shop that had one t100p card, when it came time to expand to a second card we found the price had been raised to $599 from $499 for the single port. The 4 port cards from varion are $699 on special. I believe that I can get it working without any support from varion. Perhaps so, but Varion's got a good deal going -- defer all support to the lists. :-) It would be irresponsible of me to buy a 1 port for the office at that price. Plus I think 1 card instead of 2 would be a better solution in my server. I support Digium any way I can ( t100p, plus 2 TDM cards/x100-non clone when I was first starting out last year) but $100 buys me 3 more ports. I am not bashing your choice, as it was a judgement call. I was just curious as to why you chose to contact the list first instead of the people you bought the card from. (I'm not withholding help or anything... you didn't give any information to start, but secondary to that is the fact that I don't have any experience with the Varion cards. I was merely curious.) -A. PS - it is considered bad ettiquette to CC the author as well as the list, I am already subscribed so I get two copies. Others may disagree and prefer to be CC'd as well but I believe that they're the minority and should include the specific request in their .sig. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chanisavail and IAX2
On March 23, 2005 09:59 am, Anton Krall wrote: But I get that the chan is unavailable eventhough I can make calls to that channel. Is there any chatch? The channels is defined as peer and Ialso tried doing a register on iax.conf for that channel. Everything is registering ok and I CAN make the call. Just a guess -- is there a qualify statement for that peer in iax.conf? I typically set my qualify to 500 or 1000ms (acceptable lag between me and them, it does NOT determine how often to ping them) -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Perform Action after X invalid tries
On March 23, 2005 08:53 pm, Josh Alberts wrote:
Hello, I'd like to make it so that after 5 invalid attempts of entering
an extension, the Hangup command will be issued. How would I go about
doing this?
My guess would be a combination of
SetVar($[${VAR} + 1])
and GotoIf($[${VAR} 5 })
Not having ever done it before, that's only a guess.
-A.
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Re: [Asterisk-Users] Echo on my TDM fxo
On March 24, 2005 08:08 am, Rich Adamson wrote: Then try the following in zapata.conf: echotraining=800 echocancel=yes echocancelwhenbridged=yes as a starting point for each fxo channel. Does echotraining *improve* echo cancellation at all? All I've ever found it to do is help the canceller converge faster. i.e. if the echo does eventually go away, echotraining helps it go away faster. If the echo never goes away I have never seen echotraining do anything to help that. -A. ___ 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] ADIT 600 Dynamic Impedance matching
On March 28, 2005 08:15 am, Matt Schulte wrote: Thanks for the response, it's a rather simple setup. What worries me is we're going into an old PBX, the channelbank goes 25pair about 20 feet to a punchdown block. Then from the block goes to another block (standard telco room layout) then to the phone system. The old phone system is a Meridian, about 20 years old. All the phones coming off that are analog from what I gather, the building wiring can range from 5 - 50 years old. Yeah that's all pretty standard. What's unusual is I've never heard this echo personally. I've had the customer call from different phones of course and I've dialed out from these phones to even my cell phone and haven't had a problem. What's odd is this seems to be random, if I could get it to happen everytime on a single phone then I could point fingers at the internal wiring. shrug, else all I have to blame is the cb or the wiring between it and the pbx. Until you are able to recreate it it's going to be hard to nail down... I'd start by testing individual lines -- is it always line 3 that echoes? If the Meridian's hunting you may get the same line 5 times in a row or you may get it only when the moon is in Saturn's realm... And is it only specific destination numbers or ...? There are still too many variables. -A. ___ 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] constant ringing on Zap channels
On March 29, 2005 08:40 am, Richard Reina wrote: This goes on continuously and no phones are ringing. I am using a digium T1 card and ADIT 600. Do you have the Adit600 configured correctly? It's not stuck in a test mode or anything? -A. ___ 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] Troubles with VoIP providers
On March 30, 2005 05:24 am, Obihuan wrote: My calls, depending the hour of the day, have diferent quality. Sometimes I felt cuts in the conversation or lost the sound on one of the end point. All of the providers I tested had any kind of trouble. Sounds like the trouble is on your end then. I use nufone almost exclusively and put about 5000 minutes a month through them, with multiple simultaneous calls (mid-size business) and while I occassionally have some audio problems, I have never had issue with nufone's network. I have been able to (in my mind anyway) prove that the connectivity issue was on my end, as when the problem occurs it occurs with any provider I happen to be using, and they all take wildly different paths once it leaves my (decently connected) internet provider. My internet gateway is an 1 Mb. ADSL conection y I make QOS by the router 70% of bandwidth for SIP and IAX2 protocols and 30% for others protocols. With 3 simultaneus calls. I thing that the problem is in the providers side, cause we make calls between our diferents offices via IAX2 without quality problems, but I am not sure. I said that because when in US the people wake up and start to work, about local time 13:00, our calls get more troubles, like cuts, but before that time our calls goes better than after. Is there any heavy downloading or uploading going on around that time? The unix program 'rate' or even tcpdump or ethereal should be able ot help you determine this. Remember that you can only rate-limit your OUTGOING traffic. Traffic headed for you can be dropped in an attempt for tcp's automatic backoff to slow down the connection, but as the name implies it only works for TCP. Feel free to try my traffic control script: http://www.mixdown.ca/~andrew/dump/rc.tc -- it runs on our upstream router and with it I am able to keep our connection loaded but still have voice traffic pass through as top priority. Again, it tries to limit the incoming traffic but that's more based on luck than anything else. :-) -A. ___ 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] Sangoma VS. Digium
On March 30, 2005 10:26 pm, Kristian Kielhofner wrote: It is obvious that Asterisk/TDM support from Sangoma is (and has been) secondary. Their cards support data like no other. Excellent. Voice, on the other hand, appears to be immature. I respectfully disagree. Sangoma's voice capabilities are no less and no more mature than Digium's voice capabilities. I use cards from both Sangoma and Digium. Both seem to work well but (and it does pain me to say it, it really does) Digium's cards seem FAR more finicky about the type of hardware they'll run reliably on. Sangoma's cards you can pretty much throw into any system and they work. Shared interrupts and oddball PCI chipsets included. I do believe, however, that this is merely a driver issue. If I were a more competent driver programmer I would certainly dive into this headfirst. -A. ___ 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] Many analog lines
On March 31, 2005 08:53 am, David Hajek wrote: how to use Asterisk where I need to have lets say 40 analog lines. Any ideas? A pair of TE110Ps or a TE405P and an Adit600. This will get you any combination of up to 48 ports, in groups of 8. -A. ___ 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] Sangoma VS. Digium
On March 31, 2005 10:17 am, Rich Adamson wrote: I'll second that one for sure. Maybe someone can talk Sangoma into developing a competing TDM04b card? ;) Actually I've found the TDM4XXP very good lately -- FXS and FXO. -A. ___ 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] Are there online forums instead of this email forum??
On March 31, 2005 10:26 am, Chuck Bunn wrote: I am new to Asterisk and the first thing I have noticed about Asterisk and Pingtels open PBX's is that they are using this dinosaur method of running forums. It is a real pain getting every message in the forum and essentially keeping my own database of issues. With that said are there any forums that are well used or that might even convert this email in a true forum that is searchable and that doesn't require me downloading every email. Before you go and rant on me go see how Mambo Server does it at http://forum.mamboserver.com. The forums are easy to use and thus are easy to participate in. I use mozilla Thunderbird and I have setup filters and all but it still is a pain to use this outdated email forum. Actually mailing lists offer numerous advantages over the (in my opinion) idiotic use of web forums. This has been discussed to death over the years, and even on this very mailing list. An asterisk web forum has been attempted several times and has always failed because, for the most part, the people who do the helping and have the knowlege prefer mailing lists to forums. from http://www.google.com/search?q=asterisk+forum+site%3Alists.digium.comie=UTF-8oe=UTF-8 http://lists.digium.com/pipermail/asterisk-dev/2004-February/003102.html http://lists.digium.com/pipermail/asterisk-users/2004-September/thread.html#62899 http://lists.digium.com/pipermail/asterisk-users/2004-November/thread.html#75178 http://lists.digium.com/pipermail/asterisk-users/2003-September/thread.html#22010 http://lists.digium.com/pipermail/asterisk-users/2003-August/thread.html#17720 and so on, and so forth. Call it archaic if you like but mailing lists get the job done faster, better and without all the bullshit that forums bring to the table. -A. ___ 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] Are there online forums instead of this email
On March 31, 2005 01:19 pm, [EMAIL PROTECTED] wrote: Any decent on-line forum would be much better than these digium email lists. The lists are poorly formatted, there is no easy way to post code, you cannot neatly quote anyone, the s earch function in the archive is elementary at best, there is no possibility for active use rs to moderate their area of interest, there is no private messaging, the list goes on and on. Are you on crack? Posting code is simple. Just post it. Quoting? You gotta be kidding, get a BASIC email client, what are you using, telnet? Private messaging? Send the email to the person, not the list. Moderation? Are you a child? Do you need to be moderated? I highly and kindly suggest that Digium transition off these email list and into a good, commercial, on line forum, like vBulletin (www.vbulletin.com). Feel free to do it yourself. Nobody's stopping you. I just joined this list and am amazed at how much traffic there is, how poorly formatted the messages are, how so many different topics are lumped into one list, etc. Use a threading email client. Thunderbird works. Kmail can do it. Stop using Outlook. As someone who has administered the UNIX Forums for many years, and with over 28 thousand registered users, I have seen how great an on-line, forum-based community can become. This is a great community and Digium needs to leverage the energy and the power with a good on-line forum software, like vBulletin. Excellent, you sound like the perfect person to start up a web forum and show us how good it can be. Have at it. -A. ___ 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] Are there online forums instead of this emailforum??
On March 31, 2005 02:34 pm, Tim Bass wrote: Email does not build an on-line community. The search function is primitive at best. Just like Mr. Bunn said in his original email, email lists for support are a dinosaur, and people who have moved beyond dinosaurs are considered intelligent in the evolutionary chain, LOL Uh, yeah. If Digium agrees to decommission this noisy and unorganized list traffic, I will set up a special category for Asterisk support at www.unix.com, free of charge, and since Asterisk runs on Linux/Unix, this is a good fit. I will give moderator rights to anyone Digium chooses. I am speaking for myself only. I do not work for nor pretend to speak for Digium. Blow it out your arse. It seems the only people who bitch about the mailing lists are the people who can't navigate them. Nobody's stopping you from setting up an asterisk forum on unix.com. Competition is healthy. Give it a year. Tell us which is more popular, your forum or the list. My money's on the list for the place to find solid technical answers, good (albeit sometimes offtopic) discussion and the odd humours thread. Again -- nobody's stopping you from proving us pro-listers wrong. In fact, we enourage you to compete with us. Have at it. -A. ___ 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] Online forums vs email list...
On March 31, 2005 01:07 pm, Chuck Bunn wrote: I am curious just what the advantages of an email forum over an online one. Thanks for the search tip, but it is still an annoying way to This has been hashed over time and time again. In a nutshell: - offline access - threaded access - ease of searching (granted this list doesn't fall into that category) - bandwidth considerations - ease of navigation (see threaded and searching points above) - readability (no ads, banners, panels, etc.) - accessibility (no javascript, ads, popups, cutesy icons, etc.) - no single point of failure (voip-info.org, anyone?) oh and did I mention no cutesy colours, themes, icons, avatars, ads, animations, sounds, popups, etc. etc. etc.? If you need more, check the archives. Feel free to set one up but every one that has in the past has killed it off due to lack of use. All the experts hang out on the mailing lists for the reasons mentioned above. -A. ___ 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] Are there online forums instead of this emailforum??
On March 31, 2005 11:58 am, Eric Wieling aka ManxPower wrote: Maybe I can use procmail to send an automated message to anyone that posts a message in HTML. 8-) Until you hit your first out-of-office autoreply that sends HTML... :-) -A. ___ 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] Are there online forums instead of this emailforum??
On March 31, 2005 05:11 pm, Tim Bass wrote: The UNIX Forums have over 28 thousand registered users. I have many years of experience in both email lists and on line forums and I can tell you without a doubt that on-line forums are far superior to email lists. There is no comparison. Prove it to us. You certainly talk the talk. This list, for example, has a one hour lag time between users who recently joined (at the end of the SMTP queue) and posting a message!! That is completely dysfunctional, posting a message and waiting over an hour to see it sent out to other! This is one of the problems with serial email lists. The lag time gets worse and worse for each new member of the community. It's called underspec'd hardware. Certainly someone as clever as yourself can identify the problem and how to overcome it. Forums don't magically fix this. Bigger iron and more bandwidth fix it. Plus, you cannot easily have 10 or more moderators on a busy email list server. However, in a modern on-line community, you can have many moderators sharing the work and can moderate to keep all the profanity, bullying, insults, etc. down to a zero and raise the level of discussion up to facts and knowledge. You don't need moderators. Moderators are for people who have skins too thin to function without a nanny. While I agree that sometimes it gets out of hand, having a bunch of self-appointed rulers who get to say who can and who can't post smacks of the same garbage that you get with new housing communities and their self-appointed housing police who claim they get to say what you can do with your yard and house. As the lead admin for the UNIX forums, I have watched how on-line communities develop for many years. You can't have a strong community when posters use profanity, are impolite to others, etc. Digium and Asterisk are too important to be supported by a broken email list with a one hour lag time between post and delivery, too big to moderate. . A community serves everyone, not just those who dominate with there intimating posts to others. As I said several times today -- use your superior knowlege and help make Digium better, but don't for an instant demand that they shut down the mailing lists in order to placate you. If forums are so much better they would have not only replaced this list by now, but have replaced all lists by now. Flatly put, I simply don't believe your claims about how much better forums are over lists, and I have over a decade of experience in using lists to back up my opinions. You have the bandwidth, hardware and experience to back your opinions up. I'm calling you on it. Show us how much better a well-run forum can be. Digium must fix this. Others, putting up a forum, will not solve the problem because the list will remain.All that has to happen is for Digium to endorse a forum (I recommend someone use vBulletin, but that is just my opinion) and transition off this list to something that benefits the community as a whole, and not just a few dominate individuals who like email. Digium doesn't have to do anything. If putting up a forum without dropping the list won't solve it then you don't have any argument, IMO. Competition's healthy. The wiki survives with the list. IRC survives with the list. Why can't a forum? It's not just a few dominant individuals, either. There really aren't many people who are aching for a forum. It is you, Mr. Bass, who is in the loud minority. If the majority of people wanted a forum there'd be a forum by now. But again, since you won't put your money where your mouth is so to speak, we won't be able to find out. It's kind of a shame, since someone with your kind of experience with forums might just be able to pull it off. Seriously. -A. ___ 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] Are there online forums instead of thisemailforum??
On March 31, 2005 11:28 pm, Tim Bass wrote: The discussion should not be laced with profanity, you should treat this list and others like there are women on the list and try to be polite so everyone is comfortable. Most professionals discuss matters in a way where everyone is comfortable to discuss. There is nothing wrong with being polite, not using profanity, and being respectful of people with different opinions. You would do well to follow your own rules. I believe the only profanity I used in my correspondence with you is the word 'arse' -- if that's enough to get me moderated down in your 28-kilouser-strong community then I want no part of it. Or, better yet, Digium should shut this list down and move it to a commercial vBulletin style forum and get some good moderators to delete posts that do not follow a basic set of social rules of behavior. Here are the rules from UNIX.COM, and they work very well: The rules don't look bad and they're very similar to the implied rules of any mailing list (including this one), with the exception to you reserving the right to remove any post you or any moderator sees fit. No thanks, I don't do well with censorship. You don't happen to be one of those neighbourhood czars who try and enforce what your neighbours can do with their homes in order to protect your own property value, do you? Again, there's no reason for this list to be shut down. Asterisk has a link to voip-info.org on its site and also has links to several other online resources. Why should your forum be any different? If it really is better, everyone will flock to it. -A. ___ 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] Re: Are there online forums instead ofthisemailforum??
On April 1, 2005 09:14 am, Tim Bass wrote: (2) When you registered (if you registered two years ago, for example, you receive mail in a large list before someone, say, who registered a month ago); You really have very little understanding of mailing list technology. Please, do some basic research into how various lists work, including mailman, before posting this incorrect tripe. During peak times on this list, people who have recently registered have a one hour lag time to receive messages and it has little to do with ISPs, etc. The lag varies with time of day and other factors but you are correct, it typically has very little to do with the end-user ISPs. Some simple math. (not completely accurate) If there are 2000 people on the list and it takes 2 seconds to deliver a message, and you are at the end of the list, then it will take 1000 seconds to get mail, or 15 minutes to get mail.If any network congestion, then it could take an hour for some people at the end of the list (which you will not see if you are at the first of the list). Again, a modicum of basic research is expected to participate in this list. Two seconds to deliver a message? Maybe on my father's Altair. Digium just needs some bigger hardware and maybe a fatter pipe, or even better, a few list relays. This is actually a nifty use of multicast, which is a pity it didn't take off. -A. ___ 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] Are there online forums instead of this email
On April 1, 2005 11:07 am, Tim Bass wrote: one, that is not such a serious issue, vice having a bit of profanity laced discussions with women and students in the community. I have to ask -- you keep harping about women and students -- why are they any different from any other person who dislikes profanity? Honestly. Hell, most of the students I know are more profane than even us crusty old buggers! -A. ___ 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] Re: Livevoip still no DTMF?
On April 1, 2005 01:44 pm, Brian Litzinger wrote: Made the suggested changes. Called in via SIP and Cell Phone. Still no response to DTMF. It's time to get lowlevel. iax2 debug and look for received DTMF digit '3' or something. tethereal will also show you the IAX2 IEs for DTMF. If you do not see this, the far side is not sending DTMF, and you need to complain to livevoip. IAX2 DTMF is *always* out of band. -A. ___ 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] Re: Livevoip still no DTMF?
On April 1, 2005 02:44 pm, Brandon Patterson wrote: Level 3 does DTMF inband DTMF. Period. Not on IAX2 it doesn't, and not on any kind of compressed codec with SIP it doesn't. -A. ___ 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] Is this possible?
On April 1, 2005 12:19 pm, Paul wrote: I'd like to setup my Asterisk box to receive a call on the incoming POTS line and immediately redirect back out to connect to another phone number. Im thinking I could use either the threeway feature of that POTS line, or a second POTS connected to a different FXO card. Does ANYONE know if this is possible and if so, how it's accomplished? I don't know if you could trigger 3-way calling very easily, but if the calls' coming in on Zap/1 and you want to call another number with Zap/2 and have them connect if the person Zap/2 called picks up, it's simple: exten = s,1,Dial(Zap/2/somenumber) exten = s,2,Hangup That's it. -A. ___ 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] Re: Are there online forums instead of, this email
On April 2, 2005 07:30 pm, Tore Hansen wrote: Having read a number of mailing list memos on this subject, there is much to be said for having a proper support forum BBS, rather than getting an awkward long memo with a string of messages every 3 to 5 hours. awkward long memo with a string of messages ?? don't use digests. They're evil. 1. Support for message threads - replies to messages are shown right below the original message. 2. Support for subject matter sub forums - different message categories can be established. 3. Built in search engine - messages relevant to the problem you are working on can easily be located. 4. Moderated forums - postings and discussions can easily be supervised. Trouble makers can be banned from posting. With the exception to #3 there is nothing you've mentioned that a mailing list doesn't already do. And #4 I don't see as an advantage at all, as has been discussed ad nauseam in this thread already. It would do Digium well to establish a similar BBS, since it would dramatically ease the support issues for the membership. Running a web based BBS forum is not particularly load intensive, even if it ends up having many thousands of registered users. And it has all the problems we've already discussed on this list many times. You've not given any example of how to overcome any of them. Since Asterisk is here to stay, why not get serious about the support, and do it right? As we've said many times already... You go do it. If it's truly that much better, the subscribers will flock to them. There have been numerous attempts over the last few years but the list still persists. I really, *really* wish that the forum people would see that. The forums that have been attempted must be missing *something* for them to fail. What is it? I believe it's the sheer simplicity, clarity, offline capabilities and semi-decentralized nature that keeps it strong. But hey, feel free to prove me wrong. Competition's healthy. -A. ___ 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] Router with QoS recommendations
On April 3, 2005 08:13 am, Tim Pushor wrote: To someone who has never installed OpenBSD (or FreeBSD + pf for that matter) the learning curve is going to be much much higher than 15 minutes, although one you learn PF you will never go back! I've never seen the great advantage to pf over ip and tc. Perhaps I'm just not that learned though. :-) -A. ___ 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] TE405P takes ~5mins to load.
On April 4, 2005 05:58 pm, Paul Belanger wrote: I have recently purchased a TE405P from Digium and have noticed the board seems to take ~5 mins on a fresh boot (Slackware) to start (IE: Until leds on back start flashing). Is this normal? Can it help speed this up? I have the exact same hardware on slackware without any issues whatsoever. What have you done out of the ordinary? -A. ___ 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] Channel bank question
On April 4, 2005 06:40 pm, Sean Kennedy wrote: If I have 10 copper wires coming in from the phone company, and I want to get a channel bank that will turn those into a t1 to feed into an * box with appropriate hardware, do I want an FXS or FXO channel bank? you want an FXO channel bank, or at least a channel bank with 10 FXO channels, since you'll be wiring it up to the telco. While I'm at it: Are there specific features I should be looking for? Is there a specific company everyone's had good luck with? Any recommendations on this or otherwise? You want CPD (calling party disconnect, also know as far end disconnection, disconnect supervision, etc.). On FXS it doesn't matter but on FXO it's a critical feature IMO. Carrier Access ABI and ABII do not have this feature. CAC's Adit600 does. I don't know about the others. -A. ___ 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] TE405P takes ~5mins to load.
On April 4, 2005 08:01 pm, Derrick Knight wrote: Are you viewing the output to the console as you are booting the system? I suspect that it has nothing to do with the Digium drivers and more to do with other features of Slackware such as attempting to autodetect USB or 1394 devices. If you don't have any of them you can turn off the probing in your kernel. I don't have any USB or 1394 devices on my * box. Total boot time is less than 30-45s. -A. ___ 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] WRT54GP2A-AT Morality
On April 5, 2005 10:30 am, Race Vanderdecken wrote: You have to set asterisk up to look like the Vonage switch. You have to spoof the switch. Sure, if you have their RSA private key. Go for it. If you tweak it out before it ever contacts Vonage you've got a chance, just like you can do with that PAP2s. But surely this is obvious to anyone who understands networking. You see the difference between theory and practise is that in theory, there is no difference between theory and practise. -A. ___ 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] Asterisk .call files
On April 6, 2005 03:47 pm, Gilbert Abboud wrote: I created a .call file as mentioned in the WiKi but when i place it in /var/spool/asterisk/outgoing, the Asterisk console shows unknown keyword for all the keywords used in the .call file (i.e channel, context, extension,...). Any ideas why? http://www.catb.org/~esr/faqs/smart-questions.html Give us some details (hell the .callfile would be handy perhaps) and come back. We can't help you if you won't give us the information we need to assist. -A. ___ 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] [again] Sangoma PRI vs TE410?
On April 7, 2005 05:50 am, Roy Sigurd Karlsbakk wrote: Does anyone have any details on the actual differences of using Sangoma PRI cards as compared to the TE410? How are CPU usage, interrupt load? Are there other diffferences? They are completely different beasts; the details on the actual differences are not obtainable since both are closed hardware and firmware. Suffice it to say that both seem to work well. The Digium cards seem to be more finicky about the type of hardware they'll run on, but I've certainly had no issues with either card (Sangoma A101u vs T100P and TE405P). -A. ___ 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] Re: [again] Sangoma PRI vs TE410?
On April 7, 2005 09:01 am, Tony Mountifield wrote: Do the Sangoma cards use zaptel-compatible drivers or something different? Do they provide a timing source in the same way as Digium cards do? Yes. -A. ___ 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] Issues with ringing on FXS ports
On April 7, 2005 01:44 pm, Ian Pattison wrote: Ok... I've done a bit of emperical testing but don't really know what the results mean. I'm starting to think I need an oscilloscope to measure this properly. All I have is a DMM, I'm measuring on both the AC and DC scales... AC MeasurementDC Measurement On-hook 107V 49V Off-hook 11V6V Ring drops to 44V0V Does this make any sense to anyone? No. You should have no appreciable AC in on or offhook conditions. Ring is the only time you should have AC and it should be around 80VAC. The DC measurements look alright. -A. ___ 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] Sangoma VS. Digium
On April 7, 2005 01:53 pm, Craig Guy wrote: the server they're going into (Dell poweredge 750's). When a GPL'd hardware design costs more than an entire proprietary server (including chassis, motherboard, dual hard disks and remote access card) then there is something very wrong in the market. I do not possibly see how a quarter length PCI card should cost more than an entire rack mount server. IMHO First off, the TExxxP cards are not GPL. Second, it's all economies of scale. How many Dell Poweredge 750s does Dell sell a MONTH compared to how many TExxxPs Digium sells in a year? There's nothing wrong with it; Digium is charging what they believe the market will bear. That's capitalism. If you can do it cheaper, do it and make a fortune. I know MANY PCI boards which are upwards of several dozen thousand dollars. Again, they are charging what the market will bear. Now granted, high-end signal processing and acquisition cards are a slightly different market than what you're talking about. :-) -A. ___ 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] TE405P vs TE410P
On April 7, 2005 04:02 pm, Tony Mountifield wrote: That's a pity and I'm not convinced the assertion is true. Andrew, if you read this, is your hacksawed TE405P board still in a 3.3V slot and still working? I have no intention of hacksawing a board myself, but the findings of a year ago suggested that all Digium would need to do is to respin the PCB with an extra slot and make no other changes. My TE405P is working just fine still. I agree with Digium *not* to do this unless you feel that you are able to accept the risks. It is not Digium's fault that the Xilinx Spartan II cannot be OFFICIALLY used in a 3.3V slot when programmed for 5V I/O; it is a limitation of the chip and until Xilinx can do something official about it, this is all you have. I've spoken to Mark a little with suggestions for ways to get around this without voiding Xilinx's comments but all of these methods have to be tested before they can be accepted, and all of them require a new board layout. Again -- **FOR ME** a modified TE405P works in a 3.3V PCI slot. You *will* void your warranty if you do this, and it may not work for you even if you do. I took an educated guess and used my own card, so even if it didn't work there was nobody to blame but myself. Perhaps you feel you can do the same. -A. ... it does, however present the opportunity where I can provide this service. :-) ___ 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] Re: TE405P vs TE410P
On April 7, 2005 04:42 pm, Tony Mountifield wrote: OK, I'd been told this chip could support both 3.3V and 5V, but from what you're saying, it sounds like it can be set up to support 3.3V OR 5V, but not both at once officially. Of course, when selling product it is prudent only to work within the official specs! That is correct; the chip is both 5V and 3.3V capable but the 5V logic blocks are *not* certified by Xilinx to work in 3.3V systems. I manually looked at the Voh/Vol, Vih/Vil and current specs on the I/O blocks and they seemed fine to me, so I went ahead and did it. So really, Digium could dual-key the TE405P and take the chance, but I know I certainly wouldn't do it from a business perspective. Does anyone know if the new TE411P is compatible with both kinds of slot? I see the Sangoma A104 has both cutouts on the edge connector. You'd have to see the card edge. I couldn't find a picture of it on the web site. No way :-). It was only brought up today because all I have is a TE405P and when I went to try it in a new system the slot was the wrong type. If I really need to put a PRI card in that system, I'll buy a TE410P. Or a Sangoma (but they are considerably more expensive in the UK). That's exactly what happened to me. I got the TE405P since *all* systems have 5V slots but only a few have 3.3V ones. Then my SuperMicro server came and all it had were 3.3V slots. dammit! :-) -A. ___ 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] Answering without ringing from PRI
On April 7, 2005 09:38 pm, Ugur GUNCER wrote: How can i set asterisk for when call came from pri ring once then answer pri call. In now call cames from pri then asterisk directly answering pri call without ringing. Then my carries hangup call because they said your box is answer without ringing Just Wait(4) before Answer()ing. -A. ___ 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] How many FXS/FXO ports do I need?
On April 8, 2005 08:30 pm, Aaron O'Hara wrote: My understanding is that a standard residential/business phone line carries the signal over 2 wires. Your 4-wire RJ11 wiring supports 2 phone lines. Given that each line takes 2 wires, and there are 8 wires in an FXO port, can I conceivably support 4 phone lines on one FXO port? No; the FXO ports only have the middle two pins wired to anything. One port = one line. On the phone/FXS side of things, can you also have multiple lines per FXS port? See above; only the middle two are wired. The TDM400P uses RJ45 because they can use the same backplate with the TE4xxP cards. :-) If I want to hookup 5 phones to my residential phone service with 2 lines, what # of FXO FXS ports do I need? if you want each of the 5 phones on their own 'extension' then you need 5 FXS ports. Irrespective of that, you need two FXO ports. I'd say get a TDM22P; that gives you two internal extensions (say 3 phones on one, and 2 on the other), and access to your two lines. I'd recommend against the X101Ps; not only will you have double the interrupts of a single TDM that can handle twice the number of ports, but you will also have a poorer hybrid interface to the PSTN; the TDM4xxP's FXO modules can be tuned much better. -A. ___ 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] s extension doesn't work with ata
On April 8, 2005 10:59 pm, Carlos Rojas wrote: You have well formed your file zapata.conf? Why would an ATA use anything in zapata.conf? An ATA typically takes an analogue interface and converts it to an IAX or SIP device. I'd suggest looking at his iax.conf or sip.conf, depending on the unit. I think there'd be basic documentaion with the unit to help with this. -A. ___ 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] DS3000P - 20 E1 capacity on single card
On April 9, 2005 02:13 pm, Eric Wieling wrote: izo wrote: I just checked digium's site. Looks like next big thing is coming to town DS3 on single card. Would be nice to know how many channels it can handle. Anybody had his hands on this card or knows some details ? Please God, if you can hear me, don't let them use a TigerJet chipet. I don't think they will; their quad T1/E1/J1 have no such POS on them. -A. ___ 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] DS3000P - 20 E1 capacity on single card
On April 9, 2005 08:25 pm, Eric Wieling wrote: Which specific Digium card does not use the TigerJet chip (as shown in lspci)? TE405P: 05:03.0 Communication controller: Xilinx Corporation: Unknown device 0314 (rev 01) I imagine the TE410 and TE110 are both also similarly lspci'd. -A. ___ 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] How to upgrade safe?
On April 10, 2005 10:50 am, Rich Adamson wrote: One way to do that is simply: [ snippage of simply mv'ing to a backup ] That is *precisely* how I do it for small changes, and for full-out upgrades, I have the old slackware packages standing by. -A. ___ 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] Can you comment on this Qos script? How does one shape RTP?
On April 10, 2005 04:47 am, cmisip wrote: I got this from the voip wiki but the original script didn't seem to work right so I fiddled with it a little bit. I am no expert so maybe someone can look at it for errors. This is for my cable connection. So far asterisk seems to use 1:10 while all other traffic uses 1:102. How does one packet shape RTP? That looks like my rc.tc script. The most up to date version is at http://www.mixdown.ca/~andrew/dump/rc.tc. Please note that it only tries to make things happy for IAX2. It should be fairly easy to add RTP packet detection and to throw them into the same queue. -A. ___ 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] Sipura SPA-841 Phone Review
On April 10, 2005 05:03 pm, John Novack wrote: As to that hold button. What idiot decided it should be in the middle of a row of keys, the same size as the others, and not a bright color? Maybe me; I have no desire for a bright 'hold' button. Give me the Norstar system where 'Rls' (release, hang up) is the bright button. -A. ___ 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] DS3000P - 20 E1 capacity on single card
On April 10, 2005 12:01 pm, Matthew Boehm wrote: I have a TE405P and mine shows up as Xilinx but a lvl 2 tech a digium says it still uses the TigerJet chipset. That's why it won't work in my Dell. I'll paypal you US$100 if you can find a TJ320 chip on either the TE410P or TE405P. It doesn't exist. Now they MAY have incorporated the TJ320 chip logic in the Xilinx Spartan II FPGA but I would be **VERY** surprised if they did that. Just my opinion, but I think that level 2 digium tech is full of shit. -A. ___ 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] Can you comment on this Qos script? How does one shape RTP?
On April 11, 2005 10:08 am, Sean Kennedy wrote: Honestly, the best script I've ever found is the wondershaper script ( google it ). I tried the correct one posted in this thread, tried modifying it, but in the end I just used wondershaper. :-) I started out with wshaper and just didn't like it, which is where rc.tc came from. -A. ___ 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] DS3000P - 20 E1 capacity on single card
On April 11, 2005 03:17 pm, Andres wrote: Can you confirm if there will be some sort of DSP daughther card add on of some sort for the DS3000 so that we can run G729 transcoding? I don't see how the DS3 interface would be usefull unless we could offload transcoding stuff to onboard DSPs. Or is Digium only going to recommend this card for G711 only uses? (Note, I do not work for nor speak for Digium.) G711 only; if you want transcode do that on a cluster of boxes feeding the box with this card in it. -A. ___ 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] DS3000P - 20 E1 capacity on single card
On April 11, 2005 06:43 pm, Bicom Systems wrote: Come June/July an USB/PCI DSP cost effective solution should be available to address this issues. It will transcode nearly all codec's. I am not in position to reveal the company name at this stage unless MN wants to speak up :) secondary card for DSP functions is very inefficient of the PCI bus. I'd be curious to know if the Digium cards can even do PCI-PCI DMA. And USB? I would be *very* curious to see what these products can actually do to help. There's a very good reason why any TDM boards that do off-CPU processing do it on the same card or over a separate bus... -A. ___ 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] How do I reduce echo on the Caller side
On April 12, 2005 11:59 am, Joel Jn-Francois wrote: I get an echo only from the caller end when I am making calls. I only get it for some VOIP providers. I am using asterisk Asterisk CVS-v1-0-03/26/05-16:54:47 and Grandstream HandyTone 486 and 488. My default codec is ulaw. Is there any way I can reduce the echo without comprising quality? Your terminology is confusing. When you place a call through your handytones, do you hear echo, or does the other side hear echo? -A. ___ 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: [Asterisk-Dev] Iax Trunking LD Service
This is not a development question; it's actually a -biz question but I'm not on that list so this'll have to do. On April 12, 2005 05:15 pm, Tom Dickenson wrote: Anyone know a good IAX Long Distance Trunking service that is not monthly? Kind of like a calling card charge up service?! There are plenty of IAX termination providers. I am partial to Nufone myself; they're not the absolute cheapest but I've never had any technical issues with their service. -A. ___ 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] New PRI install with new te110p
On April 13, 2005 02:40 am, Me wrote: == Primary D-Channel on span 1 down Apr 13 01:30:08 WARNING[10128]: chan_zap.c:2054 pri_find_dchan: No D-channels available! Using Primary on channel anyway 24! The telco hasn't turned up your D channel yet. If I change signalling to pri_net the errors go away, either way I can receive calls into Asterisk. When you're pri_net you are creating the D channel, but if you're connected to a telco there is no way you'd receive calls in this configuration. How should the signalling be set, to cpe or net? You're the CPE. Any idea what's causing this error? I am not entirely sure my PRI is 100% up even, * seems to be talking to it because when I pull the cable it starts giving me alerts and such, the alerts go away when I plug the cable back in. The T1 is likely up, which is what makes the LED on the back go green. When you pull the cable, the T1 is down and Asterisk tells you this. PRI is signaling on top of the T1. You can have the T1 up and have no D channel. Wait for your telco to tell you the PRI is provisioned and up (they usually work with you on the phone while they provision it, because there are a few test calls made and so on). Now if you are able to receive calls into asterisk in this state... then colour me confused. -A. ___ 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] invalid extension (need help)
On April 13, 2005 12:35 am, amna saleem wrote: I was wondering if the i extension works ,i mean i have included this in my extensions.conf ie exten = i,1,Answer exten = i,2,Playback(pbx-invalid) exten = i,3,Hangup You've already answered the call; no need to answer again, although it won't hurt. Make sure that these lines are either in the same context that your call is executing within, or that it is included in that context. -A. ___ 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] DS3000P - 20 E1 capacity on single card
On April 12, 2005 11:36 pm, Kevin P. Fleming wrote: Also, keep in mind that a DS3 is _only_ 45 megabits per second. Any PCI bus (even lowly 33MHz 32-bit PCI) can easily handle 90 megabits per Yes, but then what are you doing with it? You're shuttling the new data to/from a network card in a lot of cases. Combined with other traffic over the PCI bus for normal system operation I could see you coming close to the limitations of regular ole PCI. second of traffic. People looking a DS3 cards are also likely to deploy them in servers with multiple independent PCI buses, which would then allow for even more bandwidth. The mind boggles at the possibilities! True enough, but you still need to marshall the data going between PCI busses and to system memory. Certainly not impossible problems to overcome but they do add to the fun of getting a low latency VOIP system together. -A. ___ 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] DS3000P - 20 E1 capacity on single card
On April 13, 2005 10:57 am, Kevin P. Fleming wrote: Very true; realistically, modern PC hardware has more than enough bandwidth to do what is required. The real issue is timing, based on contention for resources, and how that impacts latency. The existing boxes out there (not PCs) that handle DS3 have far lower performance metrics than a 3GHz P4 or similar system :-) Well yes, but they're not a general computing platform either and their I/O design is quite different. They could spank any PC in terms of concurrent I/O without even breaking a sweat. :-) -A. ___ 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] ZAP channel hangs up with no apparent reason
On April 13, 2005 11:20 am, Ezabi wrote: Recently I've been having strange behaviour on my calls to PSTN, when dialing from any extension to the PSTN through ZAP the line hangs up after exactly 3:03 mins., tried to look everywhere for a string defining this timing but of no use, I even set the AbsoluteTimeout in the dialplan to 0 but still the problem persisted, any suggestions? Are you using busydetect or callprogress in zapata.conf? -A. ___ 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] Local Echo
On April 12, 2005 07:21 pm, Matt Fredrickson wrote: If you're using a TDM card, you might see if the fxotune program will help. It does impedance tuning of the card and finds the line impedance that has the lowest mean power (i.e. least echo). I've been working on it for a while and some people have had some success with it. He should also be using ztmonitor and a milliWatt source to make sure his gains are optimally set. Googling for Adjusting txgain/rxgain site:lists.digium.com and looking for Kris Boutilier's lengthy post. I found you don't need any additional patches, just the telco's milliWatt number and some time. I know you're looking for a quick and easy answer, but there isn't one. Echo is a problem that affects all TDM networks, and affects any packetized networks doubly so. -A. ___ 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] IAX introducing huge latency
On April 13, 2005 01:18 pm, chawki hammoud wrote: I placed a call through a voip provider from my console CLI. Then i ran a test iax2 show netast and here what i get when at the beginning of the call EMOTE ChannelRTT Jit Del Lost % IAX2/selectcom-2 1000 183 193-1 -1 -1 -1 0000 0 00 0 RTT: the asterisk ping round trip is 1000 ms,but i perform ping from shell command line, i get around 750ms. after few seconds, i ran iax2 show netast from asterisk CLI again, and here what i got: ChannelRTT Jit Del Lost % IAX2/selectcom-2 4340 779 790-1 -1 few seconds later, i got ChannelRTT Jit Del Lost % IAX2/selectcom-2 17945 258 810-1 Now, the final RTT is tremendous 17942 ms. But the delay is actually about four secs. Each time i check the ping from the shell command line, i pretty much get around 750ms. Where the latency the iax experiencing comming from. I am behind a nat, is there a well known issues for being behind the nat. what should i try You are describing the same type of jitter that I get; it isn't so much so jitter as it is a (very) late packet. There are a few sources of this, one of them being the network layer and the other being inconsitent timestamps coming from asterisk itself. Steve Kann and I (ok mostly Steve g) have been working on the latter. If you can consistently get this, it would be wonderful to get in contact with you offlist to see if we can recreate it between our networks. -A. ___ 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] TDM card periodic buzz
On April 13, 2005 03:42 pm, Trent Tuggle wrote: The symptom is a loud, brief buzz, almost exactly every 6 seconds, on the dot. It is only audible to remote parties, when I use an analog phone connected to my Digium TDM card. All other audio through my Asterisk box is fine, including SIP phones, music on hold, voicemail, etc. But when the TDM400P is bridged to the PSTN through my IAX2 provider, I get this repeating buzz! With it occurring, log in and type zttest and let it run for a minute and tell us the accuracy min/max/avg. -A. ___ 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] Why does this Macro Loop?
On April 13, 2005 03:45 pm, Mystery Glitch wrote:
In my [incoming] context I have something like this:
exten = 8885861575,1,Macro(vrforward,${EXTEN},8136361451)
Make sure you have a 'h' extension defined that just hangs up.
-A.
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Re: [Asterisk-Users] BOUNTY: app_hangup from exten = h
On April 14, 2005 08:31 am, Eric Wieling wrote: This is a bounty for a patch to app_hangup.c to generate an error when Hangup is called from exten = h. You should not call Hangup from exten = h. I disagree; you should use Hangup() WHEREEVER you want to make absolutely sure the dialplan stops. If you do post-hangup processing that has some branching it's far simpler to simply Hangup at the various branches than to Goto(h,end,1). A lot neater, too. A warning perhaps, but it should not error out. -A. ___ 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] BOUNTY: app_hangup from exten = h
On April 14, 2005 09:42 am, Eric Wieling wrote: exten = h will not be called unless the channel has ALREADY hung up. I understand that, which is why I'm still suggesting a WARNING and not an error. Something like No need to execute Hangup from the h exten, line is already hung up -A. ___ 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] Changing IRQ's on TDM
On April 14, 2005 10:44 am, Rich Adamson wrote: Sounds like its time to swap motherboards. :( I just wish that the PCI bridges on the TDM and TExxx cards would allow you to utilize INTA,INTB,INTC or INTD... if the mobo's fucked up at least let the card route around the damage. I'm not sure why nobody allows this. -A. ___ 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] Changing IRQ's on TDM
On April 14, 2005 12:51 pm, Rich Adamson wrote: Maybe because some motherboard designs have copper trace from the interrupt controller to the individual pci slots? There are four INT# lines on every PCI slot. INTA of slot1 is supposed to be routed to INTB of slot2, INTC of 3, INTD of 4. INTB of slot1 - INTC-2, INTD-3, INTA-4, and so on. If I could programmatically select on the PCI interface IC of each chip which INT# to utilize it owuld make interrupt routing far less of a hassle when dealing with assinine motherboards and assinine chipset IRQ routings... Of course, if people just wrote their goddamned drivers correctly interrupt sharing wouldn't be much of an issue on PCI. :-) -A. ___ 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] TDM400P Revision question.
PLEASE!! trim these replies!!! -A. ___ 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] Overheard conversation. Comments please !
PLEASE TRIM YOUR POSTS, it takes less than 30 seconds! On April 14, 2005 04:27 pm, Damon Estep wrote: The user stated that the line is PRI ISDN, not likely to be a physical short as that would take the digital line out, not produce crosstalk, had to be a switching issues with the telco or *, or user (agent) error. It's easy to have crossed lines in totally digital networks. It just occurs at the switch instead of on the physical lines. -A. ___ 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] dial plan
On April 14, 2005 05:48 pm, Michael Di Martino wrote: Call come in over the pots lines however Outbound goes out thru the VOIP provider. However looking at the configs I cannot figure out what controls how call are sent out. In other words where in the config files does it determine that all outbound calls go to the VoIP provider? in extensions.conf. basically your SIP users' default context will be whatever is defined in their context= line of their type=user section in sip.conf, falling back to the context= line in the [general] section of sip.conf if none is specifically assinged to them. Simiarly, incoming calls from the PSTN are likely coming in a Zapata device, and each Zap channel can have a context as defined in zapata.conf. The asterisk handbook (http://www.digium.com/handbook-draft.pdf) goes through most of this and is a fairly easy read. The basic flow is that an incoming call arrives into a context which is defined in extensions.conf and the dialplan starts from there. So when a SIP user makes a call, Asterisk sees an incoming call request from the SIP user and looks for a matching extension in extensions.conf, in their defined context. This context is currently defining the call route to go out through the VOIP provider. Hopefully this makes sense. :-) -A. ___ 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] codec introducing huge latency
On April 14, 2005 06:34 pm, chawki hammoud wrote: I previously posted about the huge latency introduced by iax2. It is a problem introduced by the codec. in iax2.conf, i disllowed=all and allow=gsm and the RTT is the same as I do ping shell command. When i change from gsm to ulaw or alaw, then i have the huge RTT and high jitter and evntually the call get disconnected. when i use gsm, the call doesn't get disconnected. of course, i like to use ulaw, for the internet bandwidth i have, ulaw quality is better. i have asterisk 1.0.6 and i wonder if the older version had the same codec problem. It's not a codec problem, and it doesn't appear to be an asterisk problem. Either you or the far side simply does not have the bandwidth to sustain the communications. ulaw is about 80kbps, and gsm about 28-30kbps. It really sounds like you need to review your network. -A. ___ 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] About Audio Latency from PSTN to SIP
I'm Andrew. On April 14, 2005 10:01 pm, Qiao Yuansong wrote: My asterisk box and sip phone are not behind a nat, the sip phone and asterisk box are connected by LAN, so the delay is not caused by network congestion, and furthermore, there is no delay from sip to pstn. [sip phone]--LAN--[Asterisk with X100P]--[PSTN] sip to pstn (no delay) pstn to sip (half or one second delay) This doesn't make any sense; the streams are identical. Are different codecs being negotiated when the call origination is one side then the other? put disallow=all allow=ulaw in sip.conf, under [general] and comment out all other allow/disallow lines. Restart asterisk and try again. Something basic is not right. -A. ___ 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] About Audio Latency from PSTN to SIP
Please don't post HTML to the list, and PLEASE TRIM your posts! Maybe I'm getting oversensitive to this lately but the sheer volume of bandwidth wasted due to people not taking 30 seconds to trim replies is staggering! My reply is an example of proper reply trimming; only the essential bits from your post are retained, and everything else is deleted. On April 15, 2005 04:12 am, Qiao Yuansong wrote: put disallow=all allow=ulaw in sip.conf, under [general] and comment out all other allow/disallow lines. Restart asterisk and try again. Something basic is not right. I tried your suggestion, and it make no use. So you have [some_sip_user] type=user disallow=all allow=ulaw context=somecontext in sip.conf for that sip phone? Can you post the output from the sip phone dialing a PSTN number, and then the output from a PSTN incoming call ringing the SIP phone? What version of asterisk? Perhaps you should check out http://www.catb.org/~esr/faqs/smart-questions.html while you're at it. We can't help you if you're not willing to help us. -A. ___ 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] T100P frame slips
On December 24, 2004 08:48 am, Patrick wrote: I read somewhere that to be able to hear the fax tones you need to give Asterisk 1 or 2 seconds to be able to pick them up. So put a Wait(1) or Wait(2) in your dialplan (directly after Answer would make sense to me) so Asterisk can figure out it's a fax call and throw it to the fax extension. While this is true, it doesn't apply to my particular case -- I have a DID specific to faxes which is thrown to my faxsterisk box over IAX. Basically PRI - colo* dedicated IAX2 link faxsterisk - TDM430P - faxmachines faxmachines - world = good faxes world - faxmachines = 50+% failure rate world - rx_fax on faxsterisk = good faxes -A. ___ 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] Tie web application to VOIP
On December 24, 2004 09:17 pm, Michael Giagnocavo wrote: MS SQL 2005 Express is probably the best free DB out there? And I run lots of Mono code just fine... *cough* okay. Sure. Whatever you say... -A. ___ 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] Linux Distribution
On December 25, 2004 07:29 am, Jean-Michel Hiver wrote: To answer the real question which is on the back of your head, unless you're lucky you'll probably have to do a lot of fiddling around no matter which distro you choose to get * to work... I have no idea what fiddling you're talking about -- Asterisk will run just fine on any distro I can think of. It certainly takes no fiddling on Slackware. -A. ___ 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] New TDM11B. FXS detach! We failed: 5
On December 25, 2004 02:07 pm, Lane wrote: I can make asterisk run, and I can connect to it using a software SIP phone. I can even hear the demo, but it is wa choppy. So I figure that the choppiness will diminish once I can get the FXS module to load. Remove the card entirely and run the demo -- how is the audio? -A. ___ 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] Sip
On December 26, 2004 07:40 pm, Michael Di Martino wrote: Regards, Michael Di Martino Director of MIS The telx Group Office: 212 480 3300 X.2022 Cell: 646 207 6603 [EMAIL PROTECTED] -- Sent from my BlackBerry Wireless Handheld We're impressed. Really we are. Perhaps next time you'll include some content? :-) -A. ___ 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] ZtDummy vs Hardware
On December 28, 2004 11:44 am, Rich Adamson wrote: I would seriously doubt that you can actually squeeze 12 channels through that dsl and obtain anything reasonable for quality, regardless of which asterisk codec you choose. But, it certainly would not be that hard to test it and validate assumptions. 3.4kB/sec per GSM codec (real live wire speed, including all overhead) 12*3.5 (add some margin): 42kB/sec 80kbps * 12 is 960kbps -- you could almost fit 12 ulaw conversations into my ADSL pipe. My DSL line has an 800kbps upstream so this would easily fit, especially if you have IAX2 trunking turned on (and assuming the conversations are going ot the same endpoint). If that dsl is used for anything else (including hackers/scanners hitting the IP associated with the circuit), quality will vary. Don't forget to add the IP packet overhead to the codec bandwidth estimates. This is true regardless of your internet connection, DSL isn't any different. -A. ___ 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] caller-id blocking
On December 28, 2004 06:32 pm, mohammad wrote: How can a user block his caller-id in Astersik? show application SetCallerPres -= Info about application 'SetCallerPres' =- [Synopsis]: Set CallerID Presentation [Description]: SetCallerPres(presentation): Set Caller*ID presentation on a call to a new value. Sets ANI as well if a flag is used. Always returns 0. Valid presentations are: allowed_not_screened: Presentation Allowed, Not Screened allowed_passed_screen : Presentation Allowed, Passed Screen allowed_failed_screen : Presentation Allowed, Failed Screen allowed : Presentation Allowed, Network Number prohib_not_screened : Presentation Prohibited, Not Screened prohib_passed_screen: Presentation Prohibited, Passed Screen prohib_failed_screen: Presentation Prohibited, Failed Screen prohib : Presentation Prohibited, Network Number unavailable : Number Unavailable You could also use SetCIDName/SetCIDNum as a more brute-force method. Note these likely only work on ISDN BRI/PRI interfaces. -A. ___ 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] Hook/Flash, Hold, Call Waiting, Three Way Calling
On December 29, 2004 06:25 pm, PHP Mechanic wrote: Hi, I have a TDM411B and when I am using asterisk I can't get hook/flash or hold to work when using asterisk, which means I can't use three way calling or the call waiting functions. I've tried using combinations of hook flash button and *0 on three different phones and I dont get a dial tone, the other party is not put on hold, and I don't see the keys I'm pressing in the CLI. Are you trying to use these features in * or on the line? When I take asterisk out of the equasion and plug the analoge phones directly into the telephone line everything works as you would expect. Can someone post an example of a working extensions.conf / zapata.conf where they use hook/flash that I can try. This sounds like you are subscribed to these services via your telco -- this means you need to flash the line, not your phone. To do something like that I imagine you'd have to hit # or hookflash your phone and then have dialplan logic in extensions.conf which would Flash() the proper Zap line. Doesn't sound easy but I've never done it myself. -A. ___ 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] Hook/Flash, Hold, Call Waiting, Three Way Calling
On December 29, 2004 07:05 pm, Richard Reina wrote: For threeway calling (analog phone) I just hit the flash button get a dial tone, dial the number and hit the flash key again. You're missing the point. POTS - Asterisk - Analog phone He's got call waiting/threeway calling on his POTS line -- Asterisk has no way of passing this on to the phone outside of the audible beep you hear. The best thing I can think of for him is something like this *1,1,Flash(Zap/1) So when he hears the beep, he hookflashes, hits *1 and is rejoined... I have no idea if it'd actually work or not though, since I have no phone line at home. :-) -A. ___ 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] PRI Woes continue
On December 29, 2004 07:52 pm, Andrew McRory wrote: System is built on a SuperMicro motherboard with Serverworks chipset, IRQ is not shared. Have a dialplan that worked for 8 months without errors, tried reverting to older release then upgraded to 1.0.3 stable release, currently running on fedora core 1 kernel 2.4.22-nptl.2199 (have tried plain jayne), telco says it's not us, HDLC abort seems to occur when when a Zap channel hangs up... no luck in searching the list, no luck on google, ready to scrap the T400p card, waitin on callback from digium, thought I'd post this log. Anyone make any sense of these errors? 1. Have you reverted back to the EXACT configuration that worked for 8 months? 2. Have you plugged in a cheap T100P to see if the T400P has gone bad? Basic troubleshooting skills here -- it worked, you changed something, and it no longer works. Go back to what works and see if it still works. -A. ___ 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] PRI Woes continue
On December 29, 2004 09:42 pm, Andrew McRory wrote: Well it is hard to go back to a specific configuration since I have used the system to test the rpm packages I compile. Yikes. Nothing like using a production server for testing, eh? I have reverted to a (actually several) pre 1.0 release that worked well, changed the port, moved the PCI slot, changed out the motherboard three times, enabled and disabled onboard devices, tried several kernels, rerun the cabling from the smart jack, checked the powersupply voltages, UPS, power cabling, etc etc etc. Basic troubleshooting? yeah man. That wasn't meant to be flip -- Perhaps I've just been bitten too many times myself by doing the exact same thing you just did -- I back up my config (going as far as to rsync or image the partition if I need) before changing something like that on a production system... especially something as important as our main telephone system. :-) I dont have a T100P lying around so I cant do much in the way of changing the interface. Yet. Before I commit to changing that I want to rule out any other possibilities... How can one determine without a shadow of a doubt that it is the card or otherwise? I have enabled all the debugging I can find BUT the output is foriegn to me... shrug Yeah -- I don't know -- I am the last to blame hardware (10 years as an embedded electronics designer does that to you) but failing everything else it really does seem that this is the issue, does it not? Something else I learned the hard way -- have any criticial hardware available onhand, not at a distributor, even if they can ship overnight -- I have a story about a DS3 MUX that had both controllers die and the manufacturer shipped one overnight but UPS lost it... true story. It's expensive to have hardware sitting on the shelf idle but better that than be without phone service or whatever other critical system you've got. :-) Is there a way to log all communication on the D Channel? Have I missed some critical debugging reference? I'm going crosseyed looking, tweaking and trying the same things over again. pri debug span 1 will show you all q.931 traffic and intense will show you the q.921 traffic too, but this seems deeper than that -- I am not a telco expert but it certainly seems like something very low level is buggered. I am sorry I can't be more help. -A. ___ 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] Problem with Digium TDM04B
On December 29, 2004 11:03 pm, Sudhir Kumar wrote: 1. When it dials out, many times the digits are not properly recognized by telco as I hear the announcement please check the number and dial again although I see on the screen that the dialed number is correct. I would try to stretch out the length of the DTMF digits. I have noticed that PlayDigits' digits are awfully short, I imagine that they're equally short out of the zap interface. The default length appears to be either 100ms or 800ms, I'm not sure which. from zaptel/digits.h: #define DEFAULT_DTMF_LENGTH 100 * 8 I'd perhaps try changing that value to 250 or even 500 -- Note that I have not done this myself before, I am merely guessing. -A. ___ 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] Qs about FXO/FXS cards
On January 1, 2005 04:09 pm, Rich Adamson wrote: b. don't ever post anything to the -dev list regarding a TDM card as that is NOT the forum for digium cards or drivers, Eh? If you're hacking on the code for wctdm, -dev is most certainly an appropriate place to post. If you're just going there to bitch about it well no, that's not the right place. :-) c. digium support is not addressing the issue, and, d. the amount of effort required to support the TDM card (stop *, restart the drivers, start *) in its present condition is far greater then what any reasonable non-technical customer will endure. With regard to c) I think that Digium's doing their best to try and nail down the issue but it's eluding them, and they are keeping very quiet about it. (Head in the sand perhaps?) d) I completely agree with -- I would love to deploy these cards, up to a pair in a system, but I just can't at this point in time. -A. ___ 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] Qs about FXO/FXS cards
On January 1, 2005 06:24 pm, Steven Critchfield wrote: 1. Power alarms. WTF does that mean? Wish I had some support docs. 2. On bootup, Excessive leakage module x, ProSLIC failed Auto Configuration. Again, WTF? Reboot and it's ok. But, just a reboot after driving 100+ miles to the client site is not a good option. 3. On bootup, a LED won't light. When zapata gets to it, it can't find the channel. Usually means a complete power cycle to get it to work. Those first 3 all sound like you have a problem with power supply and consistency. You don't mention what modules you have in the cards, but I bet you have FXS ports and have too light of a power supply for the job. Everyone keeps coming back to this light power supply and I just do not buy it. Period. I'm sorry, Steven, but it's bullshit. I'm speaking as an electronics design engineer and as someone who's been playing with this kind of stuff for the better part of a decade. light power supply is like irritable bowel syndrome -- it's what you call the problem when you haven't been able to isolate the cause and the patient is demanding to know what's wrong with him. Xeon 2.4GHz system, triple-redundant power supplies, Supermicro server motherboard, hot-swap everything. +5 and +12V lines are within +/- 40mV of their target voltages, measured with a 100MHz DSO -- it is *not* a power issue. P3-700 with 12 IDE drives in it, 350 or 450W (but decent make) power supply: Power quality is slightly lower but still what I would call acceptable. I get the issue where two of the three FXS modules will be seen. modprobe pauses for a good 5-6 seconds when the third module disappears -- unload/reload and it will find it, or not. unload/reload until it finds all three and you're good to go. ... Except that I can't receive faxes through it. I can send them just fine (there are two different fax machines connected up to 2 of the 3 ports, and both exhibit the problem.) Use a T100P+Adit600 FXS channel bank and my fax rate (in and out) is 100%. (this is IAX2 to a PRI connected to another system in the same location, btw, so it's not a FOIP issue.) Unplug the T100P+channel bank and swap in TDM430P... can't receive worth a shit, and that's all that's changed. I'm using app_rxfax right now instead of Dial()'ing the fax machine on the TDM port, and my rx rate is 100%. Weird, eh? Perhaps *some* of the TDM issues are power issues but the more I read and experience with them on my own the more unlikely I think it is. Truth be told there isn't a hell of a lot of power required to ring a goddamned phone. we're talking 20mA loops here and 85VAC. There is something more insidious at work than just bad power. -A. ___ 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] TE410P - Normal activity ?
On January 3, 2005 04:15 am, Nathan Alberti wrote: Is it normal for the following to occur hourly on an E1 PRI ? -- B-channel 0/1 successfully restarted on span 1 ... Yes, perfectly normal. -A. ___ 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] Qs about FXO/FXS cards
On January 3, 2005 05:35 am, Bob Goddard wrote: What matters is the volts, amps and the voltage drop when the rails are put under load. You have to ask yourself how many amps does the mb require on each rail and can the psu supply it? The total power supplied by the psu means nothing if it supplies all that power over the 12v line but leave nothing for 5v. As I said, I measured the variation on the relevant lines with a 100MHz DSO when running under load. There isn't any kind of significant droop or swell in the lines -- not when asterisk is ringing a line, not when executing a kernel compile and a find / -name 'somethingthatdoesn'texist', not when doing both. Again, if the TDM400P is drawing more than a couple hundred milliamps over nominal when ringing all four lines, something is wrong in the design, and if your PSU can't source a couple hundred milliamps more than it is under normal load, you've specc'd it too close to your average draw. -A. ___ 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] TE410P - Normal activity ?
On January 3, 2005 07:53 am, Rich Adamson wrote: Not sure why the restart code was added, but it was some time ago (maybe up to a year ago). I'd have to guess that it was added to address an issue back then and probably really isn't needed any more (but that is a 'guess'). IIRC it was added because some switches got a little flaky when not seeing activity on B channels -- IIRC one of the developers saw this kind of activity and mimiced it. It would be nice if it were not at the logging level it's currently at though, it should be more of a debug thing, IMO. -A. ___ 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
