Re: [asterisk-users] Question about PCI Slots for DIGIUMs Boards
On Sat, 29 Mar 2008, Al Baker wrote: Detailed specs for the types of PCI slots on the system were posted each and every time I posted int the line Actually, your description wasn't 100% clear at all. _PCI Express_*: _two x8 slots*_, _two x8 low profile slots*_; *_PCI-X: 64-bit/100MHz_* 1) This description seems to IMPLY that there are 5 slots total. Do you know if this is in fact correct? It implies there are 2 PCI-E x8 slots, 2 PCI-E x8 low profile slots, and 1 PCI-X slot. I wouldn't rely on that however without talking to the vendor. 2) The first PCI Express: heading would normally imply that the slots listed afterwards are ALL PCI Express slots, however PCI-X is not PCI Express, so the vendor's description is confusing and misleading. 3) All these damn *'s you keep inserting, are those all done by you, or are some of them from the web page description? Most of the time when something has a * by it that means it's conditional on a footnote that appears at the bottom of the section or page. Are there footnotes we need to know about to clarify this? 4) Is this a rackmount server or a tower case? If rackmount, is it a 1u server or a 2u server or a 4u server? Just because the motherboard has 5 slots doesn't mean the case it is installed in will support 5 cards. A 1u case rarely supports more than 1 or 2 cards, and always requires a riser card. A 2u server rarely supports more than 2 cards unless they are low-profile. A 4u server might allow 5 cards, IF the case is designed with 5 slot openings in the back. 5) Are all the card slots open and available to you at time of shipping? Many options a customer orders with a server, such as a RAID controller or additional network ports will fill one or more of the available slots. You need to be sure all the slots you need are available to you when you get the server. As for types of cards. As others have already said, PCI-X is not PCI-Express and they are not interchangeable. A PCI-Express card slot can accomodate any PCI-Express card with the same number of lanes or less. So an x8 slot (8 lanes) will support an x1, x2, x4, or x8 card, but not an x16 card. I believe the Digium PCI Express cards are only x1 (one lane) so they should fit in any PCI Express slot, but you should check with Digium's web site to be 100% sure the card you are buying is a x1 card. Unless you specifically buy a low-pofile card, a normal PCI or PCI Express Card will NOT fit in a low-profile slot. So assuming Digium's cards are full height, you only have 3 possible options. The 2 PCI Express full height slots, and the 1 PCI-X slot, assuming all those slots are open and will be available with the case you're using. I would NOT base my purchasing decision on that vague description given by the vendor that you have listed in your messages. I would contact the vendor and clarify the total number of slots, types of slots, whether they are open or not, and whether the case will support them all. Many vendors will use a motherboard with 3-5 slots on the board, in a 1u rackmount case that only supports 1 physical card. One final word of warning... don't try to stick too many cards in one box without double checking with someone who can tell you if it will handle that capacity or not. There were a lot of problems with early Digium 4-port T1 cards where you couldn't use more than 1 or 2 cards at a time because of the interrupts. The newer cards, especially PCI Express, may not have that problem anymore... but I would double check before proceeding with more than 2 cards in one box. Can anyone out there clarify (for me as well) whether you can put, say, 4 or 5 4-port T1 cards in a single box now and have it work ok? Assuming enough RAM and a fast enough CPU of course. -- Nick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Had it with Dell Garbage
I haven't bought from them recently, but I also have bought many servers and desktop systems from J N. I have at least 3 servers they built that are over 8 years old and still running in production. I've bought like 8 servers, and a half-dozen desktop systems from them since around 1996 or 1997. Again, I can't speak for anything recently, but they have been a legitimate company, reliable, and honest to deal with. I used to deal directly with the owner (Jerry Jacobsen) -- now they have a lot more employees so you're likely better off just dealing with whoever answers the phone because they're a lot bigger now. They're definitely not a fly-by-night company... it should be safe for you to give them a try if you like what they offer. Lately I buy Supermicro 1U servers and add my own cpu/ram/hd. I had a batch of bad motherboards, but other than that I like the quality so far. One word of warning... their warranty is stupid. The warranty starts the day they sell the product to the reseller/warehouse company... NOT to the end user. So if you buy from Tech Data or Newegg and it sits on the shelf in their warehouse for a month, you only get an 11 month warranty because the clock starts ticking when it leaves Supermicro's plant. They consider Tech Data or Newegg to be their customer, not you. If it sits in the warehouse for a year, then you get NO WARRANTY. Based on my dealings with JN however, I think they would honor the full warranty even if Supermicro doesn't, if you buy it from JN. They're definitely not perfect (in the early days they were slow to ship, but that seems to be better now... and last time I requested a custom quote they never got back to me) but they won't cheat you and they stand behind what they sell. I'd deal with them again if I felt they had what I needed at the right price (or even reasonably close). -- Nick On Tue, 26 Feb 2008, John covici wrote: I had a server built for me by J and N Computer Services http://www.jncs.com which is using a Super Micro c2sbe MB which I think has what you need plus 4 PCI-32 slots! Its a nice MB and I have an e8400 cpu in it. on Tuesday 02/26/2008 Matt([EMAIL PROTECTED]) wrote I've had it with Dell server garbage.They seem to change RAID controllers as much as I change socks, and then the controllers don't work with Linux, unless you load a new driver.They sell servers with a PCI-e slot in them, but then you get it and find out the RAID controller is using the PCI-e slot! Their sales folks are dumber than rocks, and they change them more often than I change underwear. [end rant]. Can anyone recommend an IBM or Gateway server that you have used with Asterisk and are happy with, and which will support RAID-1 or RAID-5 and has room for one or two PCI-express interface cards? I#39;ve had it with Dell server garbage.nbsp;nbsp;nbsp; They seem to change RAID controllers as much as I change socks, and then the controllers don#39;t work with Linux, unless you load a new driver.nbsp;nbsp;nbsp; They sell servers with a PCI-e slot in them, but then you get it and find out the RAID controller is using the PCI-e slot!nbsp;nbsp; Their sales folks are dumber than rocks, and they change them more often than I change underwear.br [end rant].brbrCan anyone recommend an IBM or Gateway server that you have used with Asterisk and are happy with, and which will support RAID-1 or RAID-5 and has room for one or two PCI-express interface cards?br ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.8 breaks tor2 support on CentOS 5.1? (kernel panic)
Thank you very much!!! What was the one line fix? Also, what file was the problem in? Also, if you know the line number or function it was in, that would be nice too. I'd do a diff, but I assume there has been other changes since 1.4.9 was released. Thanks again! -- Nick On Thu, 21 Feb 2008, Kevin P. Fleming wrote: Kevin P. Fleming wrote: I've just located an E400P from our graveyard of old cards... if it works, I'll be able to solve this problem in the morning. This has been fixed in revision 3863 of the 1.4 branch; it's a one line fix that you should be able to easily apply to existing Zaptel source code, or you can wait for the next Zaptel release which should happen later today. Sorry for the breakage. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel 1.4.8 breaks tor2 support on CentOS 5.1? (kernel panic)
The system won't boot at all if the tor2.ko file exists. Period. It doesn't matter what is in the startup scripts. Even if I DELETE the startup scripts, it still crashes until I go in with a rescue CD and delete tor2.ko from the modules directory for the kernel. It's a full kernel panic... no recovery... I don't have any method to save or copy/paste the error messages. I tried 1.4.9 and it is broken too... same panic at same point. Who do I need to report this to? Technically I think my cards are out of warranty... will Digium still accept a support call about this issue? I need to make sure this gets fixed. Thanks, -- Nick On Mon, 18 Feb 2008, James Finstrom wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I would try make clean/make/make install. also add tor2 to the black list and remove it from any zaptel init stuff. Finally once your systems up (note asterisk wont be) try loading the module with insmod. If it panics this may give you a better opurtunity to catch the output then probably contact digium support and see if they know of any issues and send the dumps. Nick Seraphin wrote: Hi all... I did some Google searches and didn't find any info on this so I'm posting it here... if this was recently discussed, I apologize for the duplication -- please point me to the appropriate thread. System Description: Supermicro SuperServer 5015M-MF w/ PDSMi Motherboard Intel Pentium 4 2.8 Ghz CPU 2 GB DDR2 Memory Digium T400P 4 Port T1 Card CentOS 5.1 (Final) Kernel: 2.6.18-53.1.13.el5 All yum updates applied. Zaptel 1.4.8 compiles with no errors or warnings. Problem: When I reboot the server, the machine crashes (hard down) with a kernel panic right as the console says Starting udev:. I go in with the rescue disk, delete the tor2.ko file, and it will boot fine. I do a make install again with 1.4.8 and when rebooting the machine locks up again - kernel panic. The panic message has a lot of stuff that I don't understand, a lot of which scrolls off the screen, but I do notice it mentions the tor2 driver several times. It happens EVERY time I boot if the tor2.ko file exists, even if I turn off the zaptel service with chkconfig, and even if I delete all the zaptel files from /etc/sysconfig and the init.d and rc.d directories. I tried loading just ztdummy and deleting tor2.ko and it works fine, so it looks like it's specifically a problem with the tor2 driver. I then tried loading the latest 1.2 zaptel release version, and that works fine - no errors, no crashes, and everything is perfect. (with tor2) So then I tried loading zaptel 1.4.7 and it works fine too. No errors, no crashes, works great. (with tor2) So something was broken between 1.4.7 and 1.4.8 that specifically affects the tor2 driver and the T400P card. What should I do? I obviously can't fix the problem myself, so is there a fix coming? Is this a known issue? Will 1.4.9 fix it when it comes out? Thanks, -- Nick Seraphin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:47b8ece6111601804284693! - -- James Finstrom Rhino Equipment Corp. Tel: 1-800-785-7073 ext. 6344 FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ext 6344 FWD: 633686 ext 6344 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHuYwKdloC7YyaIOoRAhcmAJwLSOwxBe5l11bSwEr2oNzCz3TuEQCdEjkD NBZaBSnkKxWBxUU0yUk95FI= =wQXy -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel 1.4.8 breaks tor2 support on CentOS 5.1? (kernel panic)
Hi all... I did some Google searches and didn't find any info on this so I'm posting it here... if this was recently discussed, I apologize for the duplication -- please point me to the appropriate thread. System Description: Supermicro SuperServer 5015M-MF w/ PDSMi Motherboard Intel Pentium 4 2.8 Ghz CPU 2 GB DDR2 Memory Digium T400P 4 Port T1 Card CentOS 5.1 (Final) Kernel: 2.6.18-53.1.13.el5 All yum updates applied. Zaptel 1.4.8 compiles with no errors or warnings. Problem: When I reboot the server, the machine crashes (hard down) with a kernel panic right as the console says Starting udev:. I go in with the rescue disk, delete the tor2.ko file, and it will boot fine. I do a make install again with 1.4.8 and when rebooting the machine locks up again - kernel panic. The panic message has a lot of stuff that I don't understand, a lot of which scrolls off the screen, but I do notice it mentions the tor2 driver several times. It happens EVERY time I boot if the tor2.ko file exists, even if I turn off the zaptel service with chkconfig, and even if I delete all the zaptel files from /etc/sysconfig and the init.d and rc.d directories. I tried loading just ztdummy and deleting tor2.ko and it works fine, so it looks like it's specifically a problem with the tor2 driver. I then tried loading the latest 1.2 zaptel release version, and that works fine - no errors, no crashes, and everything is perfect. (with tor2) So then I tried loading zaptel 1.4.7 and it works fine too. No errors, no crashes, works great. (with tor2) So something was broken between 1.4.7 and 1.4.8 that specifically affects the tor2 driver and the T400P card. What should I do? I obviously can't fix the problem myself, so is there a fix coming? Is this a known issue? Will 1.4.9 fix it when it comes out? Thanks, -- Nick Seraphin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Door Intercom? (was: Re: Phone with public address functionality)
On a similar note... has anyone ever seen a SIP-based door intercom unit? Functionality I'm looking for is... basically an outdoor rated weather resistant speaker with 1 button and microphone, when the button is pressed, it dials a specified SIP extension. Likewise, from the Asterisk box, someone can call a SIP extension and the call goes to the intercom speaker so you can initiate a conversation with the person at the door if they just rang the bell but didn't push the intercom button. Preferably something with power over ethernet support. Thanks, -- Nick On Tue, 4 Dec 2007, Doug Meredith wrote: I have searched for this without much luck. I want to be able to send public-address-like notices over VoIP phones. The LinkSys SPA-941 auto-answer support comes close to working, except that if you are currently in a call it places that call on hold without warning. I'm willing to consider a more expensive phone to solve the problem if I have to. Thanks for any help you can provide. Doug ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to bridge two connected calls
I spent several months trying to figure out something similar to this myself a while back. The solution I came up with finally really works great, and I think it should work for you too. Once the incoming caller is in the dialplan, issue a Dial() command using both the m option and the M() option, in addition to any other options you would normally be using for Dial(). The m option will play music on hold while the Dial() command does it's thing. The M() option will take an argument in between the ()'s, and that argument should be a Macro that you define in extensions.conf. When the call connects to the on-call operator via Dial(), execution will pass to that Macro, yet the original caller is still on hold listening to music the whole time, unaware of what is happening with your operator. The Macro can basically ask whether the operator is able to take the call or not, and you can even do really advanced stuff like announce the caller ID to the operator, or other call-identifying information, so they can decide if they need to take the call or not. When the Macro exits, have it set the variable MACRO_RESULT depending on whether the operator is going to take the call or not. If you don't set MACRO_RESULT, or set it to , then the call will be bridged and the 2 parties will be talking to eachother. If the operator rejects the call, then you can set MACRO_RESULT to GOTO:context^exten^priority and it will transfer the original incoming caller to that context/exten/priority after the Macro is done executing. There, you can either play a message to the original caller and hang up, send them to voicemail, or try another operator - anything you want really. I also found you can run an AGI script inside the Macro and it still works, so that makes it possible to do some really advanced call menu structures with database lookups, etc. The original incoming caller will hear nothing but music on hold the entire time from when you first call Dial() until the Macro is finished executing, which then means either the call is bridged, or the caller is sent to yet another context due to rejection. This type of routine typically is used for a call announce menu in a find me/follow me scenario, and is very useful considering you don't know if the recipient is going to answer, or their cell phon voicemail picks up, or their 3 year old picks up the phone, etc. You could even set it up to require the operator to enter a PIN code before they can bridge the call, so that the operator's 8 year old can't answer the call and confuse the customer, etc. Very very flexible stuff. I think this will work for what you want to do. Hope it helps... -- Nick On Fri, 23 Nov 2007, Alberto Pastore wrote: Hi everybody. I am in the following scenario: 1 Customer A calls an asterisk box over a Zap channel on a toll free number during night time 2 The incoming call enters an AGI script on the dialplan 3 The AGI script plays back a welcome message, then starts the music-on-hold stream 4 The AGI script originates a calls to a stand-by operator's cell phone (operator B) 5 When the operator B answers the call, he is prompted (via another AGI script in the dialplan) to dial 1 to be recognized as human (the AMD() function is too random to be useful) 6 After being recognized as human, Customer A must be bridged to Operator B Everything is ok from 1 to 5, but I cannot really figure out how to accomplish task #6 I've tried with MeetMe or call parking but with no success. Can anyone point me in the right direction? Thanks -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best wifi IP phone for asterisk
On Tue, 26 Jun 2007, Hendrik Visage wrote: On 6/25/07, Michelle Dupuis [EMAIL PROTECTED] wrote: We're looking at a large wifi phone deployment, and we're looking for wifi phones that: HAve a look at the Linksys WIP 300 (or something) Can be charged from the USB port I have a WIP330 and it doesn't work. Maybe it needs a firmware upgrade. Maybe it's a defective unit and all the others work fine. I haven't called support yet because I haven't had the chance. Audio in one direction cuts out completely for about 4 seconds every 10 seconds during the call. For 10 seconds it works fine... audio in both directions... then 4 seconds of silence in one direction... then 10 seconds of normal, etc etc, repeating forever. Completely unusable as-is. All my other Linksys IP Phones work great, though. I only have one WIP330. I don't have any WIP300's. My recommendation: Whatever you go with, buy ONE first for testing to make sure you're happy with it BEFORE you buy a boatload of them. -- Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best wifi IP phone for asterisk
On Mon, 25 Jun 2007, Marcus Franke wrote: Benny Amorsen schrieb: MM Siemens GigaSet SL75 The SL75 is DECT, not Wifi. Apart from that, was it really necessary to quote 20 lines and add a ridiculous 15 line disclaimer telling me that I'm not allowed to read the message? There is a GigaSet SL75 WLAN. http://gigaset.siemens.com/shc/0,1935,hq_en_0_122755_rArNrNrNrN,00.html Is this strictly a European phone? I can't find anyone who is selling them in the US... at least not a company I've ever heard of or dealt with before. Tried Amazon.com, voipsupply.com, Tech Data, and 3 pages of Google search results, etc. -- Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] got-name
On Fri, 22 Jun 2007, Luki wrote: I don't know how to contact them, but I am having the same problem. The domain is registered to Jed Stafford. If you want the domain contact details you can do a whois. The same Jed Stafford as SellVoip.net? Oh yes, that explains it... see archives. --Luki Oh, you're kidding me!? Oh geez. Guess that's *another* lesson to learn. Always check the whois on a domain and compare it against Google searches for complaints before you do business with a new company. I guess I can kiss my $5.00 goodbye. Luckily it was only $5 and I didn't pay for more yet. What bothers me more than losing the $5 is the fact that I STILL need a CNAM service, and I don't want to pay the huge amount my CLEC wants for it. Anyone know of any other CNAM services, preferably NOT run by this guy? -- Nick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CNAM.
I signed up for www.got-name.com about a week or two ago... seems to work fine with Asterisk, so long as you use 1.2 or 1.4 (doesn't work at all with 1.0). Good pricing, no minimums, no monthly fees, no setup fees. I originally saw them mentioned on one of these asterisk lists... either biz or users... so I gave them a try. Since I'm using 1.0 code on my production box, I can't use it right now... but I tested it on 1.2 and it worked good... so when I finally get my production machine replaced in a month or two I'll start using it full time. So I can't really comment on how well it works under pressure right now... just onesy-twosy seems to work fine. -- Nick On Sun, 17 Jun 2007, Alex Balashov wrote: So, is there anyone out there that provides rather generic but comprehensive CNAM-style directory services via SIP, to end-users? So I can put names to my calling numbers? Thanks! -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CNAM.
Yes... 1.5 cents per dip... you prepay the fees... and they deduct from the prepaid amount. You can start with $5.00 which seems like a low-risk to check it out at least. The CLEC I use is more expensive that that for CNAM, and they want to do it on EVERY incoming call, even wrong numbers, whether it's answered or not, per PRI. So since I get several thousand wrong numbers a month, and only 100 or so calls that I actually CARE what the CNAM is on those calls, I can set it up in Asterisk to only do the dip for certain DNIS numbers. I calculated that instead of $70+/month this will cost me $1.50/month. Nice savings. :-) I just hope it's reliable when the call volume picks up more. -- Nick On Sun, 17 Jun 2007, Alex Balashov wrote: Thanks Nick. Do they charge per directory dip, or in some other unit? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Slightly OT:CSU on Digium cards, and it's requirement
On Thu, 14 Jun 2007, C F wrote: On 6/13/07, Erik Anderson [EMAIL PROTECTED] wrote: On 6/13/07, C F [EMAIL PROTECTED] wrote: This is just weird I wrote it in caps so you can read it but you still didn't read it so here it is again: its a T1 card that does NOT have a CSU in it, and it is working fine and yes it is a T1 providing PRI. sarcasm Dang shmaltz. You've convinced us - we've all been wasting our precious money on CSUs this whole time. We're all idiots! /sarcasm Seriously - if you're so sure about your card not having a CSU, what is the make/model? Pony up, man. It's a Panasonic KX-TA0187 for T1, or KX-TA02290 The docs and technicians say it doesn have one AND that the FCC requires it. Hence my qeustion does the FCC require it. I think what he's referring to is really the KX-TD187... which is a T1 interface module for the Panasonic KX-TD1232 Digital Hybrid Phone System (I have one of these systems, but not the T1 module). Now there is a KX-TA1232 analog system, and maybe there was a KX-TA187 module for it that has since been discontinued... but I think he meant the digital one. http://www.ablecomm.com/t1isdideq.html They do SAY it doesn't have a CSU... but it's beyond my understanding of how it could possibly work without one. They seem to sell a separate CSU module that can go with it. Maybe he's just not seeing the extra little box because there's more wire between that and the demarc? Was this a system that was already installed for you? Or did you install it yourself? Maybe the CSU is external and you just didn't recognize/see it there? When I first started working with T1's, most CSU's were external. I still have several of them in storage in fact... and I still use external CSU/DSU's on my production network today. :-) I'm typing this message and it will be sent over a T1 connected to 2 external CSU's before it reaches the internet. Bottom line is, no matter what the FCC says... and if somehow you managed to get it to work without a CSU... I believe the phone company would have a fit if they knew you connected equipment to their network without a CSU on it. They're very big on standards-compliance and stuff like that. Sometime look into their rules and regs about colocating equipment inside one of their CO's... it's very very strict. -- Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slightly OT:CSU on Digium cards, and it's requirement
On Thu, 14 Jun 2007, C F wrote: Bottom line is, no matter what the FCC says... and if somehow you managed to get it to work without a CSU... I believe the phone company would have a fit if they knew you connected equipment to their network without a CSU on it. They're very big on standards-compliance and stuff like that. Sometime look into their rules and regs about colocating equipment inside one of their CO's... it's very very strict. The last thing you say is why I am asking this question. The compliance doesn't realy bother me that much, what I am afraid is if the provider notices this and decides to cut it because of that. Whether they would actually cut you off or not probably depends on A) if they find out about it, and B) whoever finds out about it is a strict play-by-the-rules kinda guy and/or has a grudge against you or is having a bad day. A lot of telco employees tend to look the other way... especially if it's not their job to care about it. But... they would have every right to terminate the service if you don't have proper equipment connected to their network. So if they DID decide to terminate it, they would legally have the right to do so, and you would have no recourse other than possibly to purchase the correct equipment and maybe pay a reconnect fee to get service turned back on, which may take days/weeks/whatever time frame to do so. So it's basically a question of, can you afford the downtime caused by them shutting you off if/when they ever found out and/or cared enough to follow the rules. The other possibility, considering it is working for you now, is that there IS a CSU built in but they don't want to tell you... maybe for example because it's not FCC certified... or so that they can charge you for an external CSU. -- Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA400 and asterisk
Wow... how did you get it to work with Asterisk? I bought one a few months ago... played with it for 2-3 days... couldn't get it going... so I stuck it in the basement on my pile of projects to work on when I have a bunch of free time to waste, hoping it doesn't become a doorstop. Did you have to upgrade the firmware to get it to work? Or were there any special config settings? Like in the Grandstream 4104, I found out from Support that they require the SIP account usernames to be all numeric, no alpha characters, or it won't work. I followed the instructions I found on a web site for getting the SPA400 to work with Asterisk, but when I duplicated their settings, it didn't work at all for me. I used my own experience to tweak various things, and try different scenarios... until I finally gave up. I'd love to get it working... if you could share a sample config or other advice, I'd appreciate it. Thanks, -- Nick On Tue, 12 Jun 2007, Alberto Sagredo (M) wrote: You could use it as a usually FXO Gateway. I have tested and it works fine. 2007/6/12, MBIT Technologies [EMAIL PROTECTED]: Hi Guys I am just looking to see if you can help me. I have been investigating the SPA400 and it seems to run asterisk for the voicemail system. Does anyone know if it could be programmed to also talk to the FXO ports? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alberto Sagredo RD area Peoplecall Email : [EMAIL PROTECTED] Blog: http://www.voipnovatos.es Office phone : +34 91 120 5080 Direct phone : +34 91 120 50 39 Peoplecall Network : 700 757 139 Fax number : +34 91 661 9460 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA400 and asterisk
On Wed, 13 Jun 2007, Gergo Csibra wrote: Wednesday, June 13, 2007, 9:44:08 AM, Nick wrote: I'd love to get it working... if you could share a sample config or other advice, I'd appreciate it. http://www.voip-info.org/wiki/view/Linksys-Cisco+SPA400 http://voxilla.com/voxilla-tips/voxilla-tips/the-linksys-spa400-and-asterisk-886.html http://www.justfuckinggoogleit.com/ How about http://www.justfuckingreadtheoriginalmessagebeforebeingasmartass.com/ If you would have read my message, I already said I tried sample configs I found on the web and they didn't work for me. But I guess it was just more fun for you to try to look superior. -- Nick -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE... 2 down, 1 to go (3 questions - variables, upgrading, and IRC)
On Fri, 8 Jun 2007, Jared Smith wrote: On 6/7/07, Nick Seraphin [EMAIL PROTECTED] wrote: Still need an answer to this one. I wrote a response yesterday, but it looks like it didn't come through for some reason. The answer is to use the DumpChan() application and watch the CLI when it's called. Yeah... Thanks... I got your first reply after I sent the second message. I guess the mail list server was backed up. Unfortunately, DumpChan didn't appear until 1.2, so I'm going to have to upgrade anyway. Thanks, -- Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE... 2 down, 1 to go (3 questions - variables, upgrading, and IRC)
On Fri, 8 Jun 2007, Steve Edwards wrote: On Fri, 8 Jun 2007, Jared Smith wrote: On 6/7/07, Nick Seraphin [EMAIL PROTECTED] wrote: Still need an answer to this one. I wrote a response yesterday, but it looks like it didn't come through for some reason. The answer is to use the DumpChan() application and watch the CLI when it's called. I interpreted this question as how do I see the variables for this channel using the CLI? -- show channel foo. That's true... that's exactly what I wanted. But DumpChan does provide the functionality too. The problem is, both DumpChan and the enhancement to show channel that you describe (listing the variables) weren't added until at least 1.2. My version, when you do a show channel whatever doesn't show any of the variables. I checked that long before I sent my first message. :-) Thanks, -- Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 3 questions - variables, upgrading, and IRC
1) How can I get a list of currently set channel variables for a specific channel in Asterisk, including custom variables set by the dialplan? I don't want a static list of variables from a web site, I need the current dynamic list that shows custom variables that are specific only to this channel. 2) Where can I find a comprehensive list of problems and incompatibilities when upgrading from Asterisk 1.0 to 1.2 or 1.4? I have a production machine that has been running on a CVS HEAD version from around the 1.0 days (definitely long before 1.2) and I understand a lot of things changed, like variables, functions, and syntax that were either deprecated or removed. I need to prepare myself for the upgrade in advance to limit the amount of downtime to an absolute minimum, but this means I need to know what is going to break when I try to use my old configs (extensions.conf, sip.conf, etc.) with a 1.4 system? I really don't want to put out the fires one by one, testing things until it finally works... I want to fix everything in advance before the upgrade. 3) When I tried to go to the IRC network to ask these questions online, I found myself unable to get into #asterisk because I can't remember the password I used to register my nickname. Is there a way to reset my IRC nickname/nickserv password? Or is there a person in charge of it that I can email to get this taken care of? Obviously, I would ask on IRC if I could, but I can't even get into the channel. Thanks, -- Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UPDATE... 2 down, 1 to go (3 questions - variables, upgrading, and IRC)
On Thu, 7 Jun 2007, Nick Seraphin wrote: 1) How can I get a list of currently set channel variables for a specific channel in Asterisk, including custom variables set by the dialplan? I don't want a static list of variables from a web site, I need the current dynamic list that shows custom variables that are specific only to this channel. Still need an answer to this one. 2) Where can I find a comprehensive list of problems and incompatibilities when upgrading from Asterisk 1.0 to 1.2 or 1.4? I have a production machine that has been running on a CVS HEAD version from around the 1.0 days (definitely long before 1.2) and I understand a lot of things changed, like variables, functions, and syntax that were either deprecated or removed. I need to prepare myself for the upgrade in advance to limit the amount of downtime to an absolute minimum, but this means I need to know what is going to break when I try to use my old configs (extensions.conf, sip.conf, etc.) with a 1.4 system? I really don't want to put out the fires one by one, testing things until it finally works... I want to fix everything in advance before the upgrade. Found UPGRADES.TXT in the source tar balls. I was hoping for something a little different, but this should work, so long as I read everything in both the 1.2 and the 1.4 versions of the file. I assume these files contain all the known incompatibilities with upgrading... if not, please let me know. 3) When I tried to go to the IRC network to ask these questions online, I found myself unable to get into #asterisk because I can't remember the password I used to register my nickname. Is there a way to reset my IRC nickname/nickserv password? Or is there a person in charge of it that I can email to get this taken care of? Obviously, I would ask on IRC if I could, but I can't even get into the channel. I finally figured out my password, so this is no longer an issue. -- Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Provisioning Linksys PAP2T ATA's
On Fri, 8 Jun 2007, Mattt wrote: Doug, We just pre-provision the Linksys CPE (including PAP2(T)-NA's) in the lab over TFTP after barcode-scanning the relevant information for that unit into a web management interface and, once the unit is deployed onsite, it continues to pull it's config from our prov server over HTTP. The provisioning service itself is just a PHP engine which pulls the relevant settings for that CPE from a database - this way, we can have individual parameters for certain customers (who might be, for instance, having issues with echo or latency, etc, etc, or some need for a different config to our norm). Does your PHP engine just generate a plain text XML file to HTTP via stdout? Does it work the same for both PAP2's and the IP Phones (SPA942/SPA962, etc)? I've successfully provisioned an SPA942 via plain text XML from a tftp server, but I've never tried a PAP2 remotely, nor have I tried with HTTP. Do you find any problems with security with it being in plain text? Do you disable the restore to factory defaults thing? Do you upgrade to the latest firmware before shipping out your PAP2's? (more below) Works a treat, is easy as (once the coding is done), and takes about 10 seconds to pre-prov (also provides the opportunity to ensure the unit isn't a DOA). Customer simply receives the device and plugs it in, waits a few seconds (the pre-prov doesn't configure the unit, just prepares it for remote provisioning from the target site), then starts making calls. Oh - and, unless you can locate some new, old stock, you won't find the PAP2-NA (with the blue LEDs) anywhere. They were discontinued many months ago... What's the difference(s) between a PAP2 and a PAP2T? I've only got PAP2's, and I've got several spares in inventory that I'm hoping I won't regret having. :-) Thanks, -- Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Semi OT - Cable products suppliers
My favorite place for all cable infrastructure products is the local Graybar warehouse. I'm lucky to have one only about 20-25 minutes away. Check www.graybar.com to see if they have one near you. On the web, www.ablecomm.com has some nifty and hard to find telecom-related products and tools, but they are expensive. I've ordered from them several times... they're reputable. But expensive. Graybar is the cheapest place I've found. If someone knows a good web-based store with better pricing and good selection, I'd love to see it for my own use. :-) -- Nick On Fri, 8 Jun 2007, [EMAIL PROTECTED] wrote: Anyone have a good recommendation for a supplier of punch blocks, 25 pair connectors and cables, etc? Thanks BEN BROWN ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 3 questions - variables, upgrading, and IRC
On Thu, 7 Jun 2007, Jared Smith wrote: On 6/7/07, Nick Seraphin [EMAIL PROTECTED] wrote: 1) How can I get a list of currently set channel variables for a specific channel in Asterisk, including custom variables set by the dialplan? Use the DumpChan() dialplan application. Thanks... guess I'll have to bite the bullet and upgrade to 1.2 (probably 1.4, since 1.2 is soon to be end-of-life'd. Will 1.4 run on a 2.4 kernel? Or am I going to have to upgrade that too? -- Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] click to call
This is exactly what I understand click-to-call to be in every case I've ever seen it implemented. Two outbound calls, bridged together from a central server. On Sat, 2 Jun 2007, Joseph Bajin wrote: You shouldn't need a softphone to do Click to Call.. The idea is pretty simple, and maybe I am missing something since I am haven't worked with Asterisk enough, but basically you start off by making the call to the Initial Party, Park the Call, Call the Other Party and then Connect them together.. Seems pretty simple and easy enough to do. On 6/2/07, Steve Totaro [EMAIL PROTECTED] wrote: JIAX client could be modified to do this for free. The project has been stalled for quite a while but the demo works and the source is there and open. I have seen people successfully use it for click to dial for free but they use it internally and do not intend to put it out in the public domain. http://www.hem.za.org/jiaxclient http://forums.vtiger.com/viewtopic.php?t=1636start=40 Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Gordon Henderson Sent: Saturday, June 02, 2007 4:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] click to call On Fri, 1 Jun 2007, Anton Krall wrote: So Guys, no go on this topic? I trialled a click-to-dial application recently. It generated a lot of controversy on the list (search the archives) because various people said it couldn't be done/wouldn't work, etc. Then there were whinges about the commercial nature of the application (it's licensed, not free, and details were being posted to the -users list) and so on. Personally, I didn't see why as the creators of the code were simply replying to questions asked by list members, however... (That's probably why you've not gotten many replies ;-) So the thing I trialled was a button on a web page which downlaoded a soft-phone program written in Java to your browser. The soft-phone uses the IAX protocol to connect to an asterisk server, then depending on the javascript that you write to encapsulate the button on the web page, you have the ability to specify username password (to authenticate back to the asterisk server) and number to dial - the number you dial could even be entered via more javascript on the webpage, and the asterisk server at the back-end can then do what it needs to do with the number - dial an extension in a closed system, or even initiate a dial-out to the PSTN, if the server as such a connection and the connection is authorised. The end-user pushing the button doesn't need to see any of this at all - it can all be embedded in the javascript behind the button. You can specify callerId too, or dial different numbers, so the person answering the call could use this information to know what web page you are on for example. You can even embed it into an email signature with a different number then you could tell if they are calling you in reply to an email, and so on. (And much as I hate big HTML based email signatures, if done correctly this could be quite effective - and it doesn't need to download the Java - about 120KB until you click on the button) (They have a demonstration client which works with the Tesco VoIP service - you enter your Tesco username/password, then get a phone application with buttons, etc. The Tesco VoIP system unusually uses IAX rather than SIP as their transport mechanism!) I tried the application on a WinXP box, Linux box and Mac, and as long as the sound system was setup to work with the headset microphone, it just worked - At last, Java doing what it was supposed to be doing, working correctly cross platform! Some of the whinges to the list were that a soft-phone couldn't possibly be written in Java as Java was too heavyweight - well, this is the latter part of the first decade of the new millennium and Java has come a long way since it was first released, and they couldn't be further from the truth - in use on my 2GHz Linux box, it was using about 2-3% CPU, and at 120KB to download, is no worse than your average mid-resolution camera image these days. If this is what you're after, then go to http://www.mexuar.com/products_connect.shtml They were happy to give me a time-limited trial of the software, which I used, and found worked really well. You will need to write some html and javascript to encapsulate it into your own web page, but that's not hard to do and examples are provided. Now all I need is some clients to sell it to ;-) Gordon -Original Message- From: [EMAIL PROTECTED]
RE: [asterisk-users] click to call
On Sat, 2 Jun 2007, Steve Totaro wrote: That is a totally different concept than we have been discussing. You are talking about actual phones and the person clicking, then entering their phone number having to pick up a physical phone. This is as trivial as generating a .call file and dialplan magic. The concept we are discussing is clicking a link that connects the clicker to whatever via the computer using a headset or speakers and a mic. No phone or numbers involved, at least to the clicker. The problem is, the only people who will be able to use that link are geeks that have a headset/mic on their computer. Most normal people don't have those devices, and even if they did, they feel much more comfortable with the concept of making phone calls using a telephone. We all often forget that the vast majority of the outside world is not technically-inclined in any way, and that unless your web site is only targetted towards computer geeks, you're creating a huge barrier for the average customer. Everyone has a phone, though. If the analog FXS adapter had not been created and reduced to an affordable price, VOIP would still only be about as popular today as it was in 1995. -- Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] click to call
On Sat, 2 Jun 2007, Dean Collins wrote: Hi Nick, Totally disagree with you. We the prevalence of skype, im and MP3's You'd be surprised how many users have headsets. Well, I'm probably one of the geekiest guys in this geographic area. I've been online since 1986, been on the internet since around 1991/92, and been running an ISP since 1995. I love electronic gadgets, computers, networking, telephony, etc. I write software. I build things. I've even spent hundreds of hours reading various ILEC and CLEC tarriff filings for fun. I'm about as geeky as they come, outside of silicon valley. Yet, I do NOT have a headset/mic thingy for using voip from a computer, and I've never used skype in my life. Never even been to their web site. Oh, and my very large MP3 collection is ripped from legally purchased CD's, not downloaded from the internet (unless you count purchased iTunes songs). Anyway, other than possibly a few local ISP-owners/techs I know personally in the area, I don't know of ANYONE who has a headset/mic setup for voip over the net, nor anyone who uses skype or has ever used skype and mentioned it to me. Now, I don't know alot of 14-24 yr olds, which may be the demographic that embraces those things... I mostly know people who are between 25 and 65. Yet every single person I know, from age 5 to age 95, has a phone or has access to a phone. Now maybe in the big cities... or out in silicon valley... the numbers are quite a bit higher in your favor. But in flyover country I'd say the number of people who have a computer with internet access and a headset/mic setup for voip is much less than 5%. Compared with 99.999% that have a phone or access to a phone. And even if they HAVE the device, most people feel far more comfortable talking on a phone than they do at a computer console. Again, VOIP wouldn't be very popular at all today if it weren't for the ability to use your regular phone, and/or phone-like devices (IP phones). I guess it depends on where you are and who you know as to what seems to be the average behavior of people. For me, it seems obvious. But maybe the people you know, in your area, are completely different than my situation. If your site markets to people outside your area, however, then it's important you know what the people around here are like, because I'd bet this is more average nationwide. (I could be wrong) What you are missing here is the additional functionality you get from using a browser delivered call then a pstn call. If you have the right business drivers allowing your users to reach you for free via IP Click-to-Talk is a huge plus over the older generation Click-to-Talk. Well obviously it's cheaper, since you don't have to pay for PSTN termination. And certainly, if your target market is people who have those devices, then it makes perfect sense. Or likewise, if you offer it as an additional option to the standard methods click-to-call methods, and phone/email like everyone else has. There's definitely a place for what you describe. But the typical non-geek consumer won't have a clue what to do with it. Like I said before, it just creates a barrier that doesn't need to be there. Everyone has a phone, so using a phone is going to allow you to reach the most potential customers. It's kind of like companies that build their web sites so that they only work with IE. They alienate a huge base of users that either refuse to use IE, can't use IE, or really don't want to use IE. I'm the latter. So I have to make a choice... do I want to visit that site bad enough to open up an IE window (and possibly move to a different computer terminal if I don't have IE on the one I'm using), risk the security problems of IE, and be generally inconvenienced? Or will I just go to a competitor's site, or find another source for the info/product? If you are selling a product or service, and make money from your site, you NEVER want to give the customer ANY reason to even THINK about going somewhere else. Especially if there are easily implementable ways to avoid it. -- Nick P.S. If this came across at all as being irritable, sorry, it wasn't meant that way. I think this hot weather is getting to me. :-) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] High Port Count ATA
Those 24+ port devices get *really* expensive, *really* fast. I would probably go with the new Linksys SPA-8000 (I think that's the model number)... its an 8-port FXS adapter and is only around $300 or so. So 110 ports, you would need 14 of them... $4200 with the advantage that if one dies, you only lose 8 ports, not 24 or 48. Using high-density solutions could easily run you over $10,000+ depending on the model/manufacturer/density. Plus, at $300 you can afford to keep 2 or 3 spares on the shelf just in case... where as with a high density solution, even keeping 1 spare would be very costly. -- Nick On Thu, 31 May 2007, Douglas Garstang wrote: I'm trying to find a high port count ATA device. We have a lot (up to 110) analog devices that we need to bridge to IP. Rather than go out and buy 110 ATA's, it would make more sense to buy a single chassis type device with some number of ports and blades. Anyone know if such a device exists? Thanks, Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Patton gateways (was: stream file not working but get data and exec background work)
On Wed, 23 May 2007, Patrick Fortin wrote: Here is the setup Pri SIP Gateway(Patton) - Asterisk (ztdummy) - SIP Phone Wow... you're the first one I've seen who is using a Patton gateway, so I hope you don't mind me asking you... why did you go that route? From what I saw a while ago (I may be wrong) I thought Patton's SIP gateways were way overpriced compared to other similar solutions. Now I'm going to have to go and double-check... maybe I misunderstood. Or was there some sort of other reason why you chose Patton that made price un-important? Do you like their sip gateways? What port density (or model?) do you have? -- Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Dial Command to a non-Asterisk url
The 2 most common problems I've seen for no audio in one or both directions is usually either a firewall (which you already said you don't have) or a CODEC problem. Make sure both sides are negotiating the same CODEC. I've often seen situations where something like the Asterisk server will allow gsm, g711, etc. and the phone is set for g711, but because gsm was first in the list on the asterisk side, asterisk was trying to do gsm and the phone wanted g711 and they wouldn't sync up. It wasn't until I did a: disallow=all allow=g711 in sip.conf that it finally started working for me. That may not be your exact problem, but my guess would be a CODEC issue if it's not your firewall. -- Nick On Wed, 23 May 2007, Gavin Henry wrote: Dear All, I have a tiny dial plan like: [testing] exten = 454,s,Ringing() exten = 454,n,Wait(4) exten = 454,n,Dial(SIP/[EMAIL PROTECTED]:5605,10) exten = 454,n,Hangup This connects fine when I dial 454 from any extension in my system, but there is never any audio? Where can I start to look for debugging this? It's all internal so no NAT problems? Thanks, Gavin. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [*Win32 0.60] Sending call notification by e-mail/web?
On Tue, 15 May 2007, Vincent Delporte wrote: Hello, In case there are other users of the AsteriskWin32 port... I haven't really used the AGI feature of Asterisk to run an application from extensions.conf. *Win32 supports Perl, which I don't know. Apparently, it's also possible to write AGI applications as EXE's (there's a eagi-test.exe file installed by default). = When a call comes in, I'd like an AGI application to send an e-mail and send CID name/number to a script on a web server. Is this the correct way to do it in Perl, with the modules available in AsteriskWin32? Could I rewrite this in Delphi instead? ALL AGI scripts are basically just programs that read from stdin and write to stdout. They can therefore be written in almost any language. So yes, Delphi should work fine. (I have very fond memories of Delphi, and before that, Borland Pascal w/ Objects for DOS, and before that, Turbo Pascal... one of these days I'll have to get the latest version of Delphi and take a walk down memory lane. These days everything is C this or Perl that. I loved Pascal. :-)) -- Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Web based call control
On Mon, 14 May 2007, Jordan Novak wrote: Does anyone know if it is possible to use a manager command to answer an incoming call and not consider it answered unitl it is received. Here is an example, I am deivering a call in the dialplan to a home telephone number. I don't want his voicemail to answer and I have no idea how long it will take to go to their home phone voicemail, but I don't want to deliver the call there, I want it to go to the next priority in asterisk. So I was thinking that it would be nice to build a web interface that they could have a button to answer with. This would send a manager command to the server telling it to answer the channel, any thoughts on how to do this. Most companies that I've seen who want this type of behavior have 2 available options. 1) Ask the person with the destination home phone number to cancel their voicemail from the phone company, so that the phone just rings until it times out. Usually they won't want to do this, because other incoming calls to their home number would not go to voicemail then... so unless it's a line dedicated as a home-office number that only receives calls from your system, they won't be happy. 2) The most popular option is have the caller listen to music while waiting, and then when the destination picks up, play an announcement to them and wait for DTMF input. Hi, you have a call from the PBX system... press 1 to accept this call. If they press 1, you connect the two parties. If after X number of seconds you don't get a response, i.e. voicemail picked up, or their 4 yr old child picked up, then no DTMF will be received and you set it to time out and either go on to the next priority or send it to voicemail. As far as I know, there is absolutely no way to prevent the destination's phone company voicemail (or answering machine) from answering the call unless you have a reliable way to know exactly how many rings it is set to before it will answer, AND you have assurance that the number of rings will NOT be changed. Granted, probably 90% or more of all voicemail systems out there default to answer after the 4th or 5th ring, you could always set it to time-out after 3 rings and have decent success rates... but if the caller is already on the phone, unless they have call waiting, the call will go to voicemail on the first ring. I guess theoretically you could always use voicemail detection to see if voicemail answered the call instead of a human, and then somehow grab the call back and transfer it to another extension... but this would probably require a custom Application or modification of the Asterisk source code. If I'm wrong, and there's a better way to do this, someone PLEASE let me know... because I'm planning a project that will have the exact same problem. (That's why I've researched this problem already.) -- Nick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI Silence detection
On Thu, 17 Jul 2003, Stuart Hirst wrote: Does anyone how you might detect a period of x milliseconds of silence using AGI ? I added silence detection to the Record() application and to the record function in the AGI interface in asterisk. It's based on dsp.c, like someone else said. Basically, it waits for X milliseconds of silence and then stops recording, and then deletes those X milliseconds of silence from the end of the recording. If you look at the record function in AGI, you should be able to see how to do this pretty easily. The part about detecting silence shouldn't be difficult, but I'm not sure how you would tell the AGI program that it was detected. -- Nick EagleQuest, Inc. Metro Telephony, Inc. Rochester, Michigan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
