RE: [Asterisk-Users] ATTN: Keith - Seriously OT
On Friday, June 10, 2005 3:16 AM, Andrew Kohlsmith [SMTP:[EMAIL PROTECTED] wrote: On Friday 10 June 2005 04:08, Terry H. Gilsenan wrote: Received: from source ([81.56.129.44]) by exprod5mx8.postini.com ([64.18.4.10]) with SMTP; Fri, 10 Jun 2005 00:29:16 PDT Your MTA claimed it was called SOURCE but rDNS tells the recipient MX that it is called: mail.linuxautrement.com I too will block emails with a non-FQDN HELO or EHLO. I feel, however, that reverse should not have to match forward lookups for mail exchangers. It's an assinine requirement (my box does web, mail, dns and a host of other services, why should I need it to be called 'mail' for both forward and reverse lookups just to get mail flowing? Assinine. -A. Your server does not have to be called 'mail' for DNS and rDNS to work properly for mail delivery. All that is required is that a reverse lookup returns whatever the actual name of the server is and the server needs to use that same name when it issues HELO. My server at home is called 'fs-1' and the one at work is 'troutdale'. Both systems work properly just because I set up the DNS and rDNS records to match the names of the servers. There are a lot of broken rDNS records on the internet, and that's not likely to change anytime soon. I only have control of a very tiny portion of DNS and rDNS space, but I still feel obligated to make my part work properly. It's what makes the internet work. Would you feel OK driving around in your car, knowing that some large percentage of the street signs were not correct? ___ 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 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] VOIP-INFO
I queried the name servers and it looks like this site may be hosted on a cable or DSL system which doesn't allow static IP addresses. It's possible that the DHCP and the dynamic DNS fell out of sync. All of this is just a guess based on the name server results. On Thursday, June 09, 2005 2:49 PM, Wiley Siler [SMTP:[EMAIL PROTECTED] wrote: Been that way a couple of hours I think W -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Coulthurst Sent: Thursday, June 09, 2005 2:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] VOIP-INFO Anyone else unable to get to www.voip-info.org? Site is returning 'connection refused' here. Chris Coulthurst [EMAIL PROTECTED] ___ 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 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 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] VoiPSupply Dot Com
Karl has already stated more that once that this was NOT a credit card purchase. If a credit card was not used for the purchase, why would you need a Ship To address on file with the credit card company? Cory Andrews from VOIPSupply has also admitted that the sales rep who took the order made a mistake and failed to notice that a Ship To address had been supplied. On Thursday, June 02, 2005 9:04 AM, The VoIP Connection [SMTP:[EMAIL PROTECTED] wrote: I know I'm running the risk of fanning the flames on an already belabored thread here, but there is some misinformation flying around. Credit card fraud is an unfortunate fact of life, and it costs everyone who isn't perpetrating it money. There is no single universally agreed on process that will guarantee a merchant protection. If there was, somebody would figure out how to game it. Different banks have different merchant account requirements, e-businesses use different procedures to protect themselves, and of course different businesses tolerate different levels of fraud. Some vendors require that items be shipped to an address on file to protect themselves. Others (like us) do not. We have a process for validating the card for these cases which our bank has agreed is adequate in most cases. It's a little more time consuming but it is something that many of our customers require. There was a misunderstanding, let's move on. I am really tired of seeing VoiPSupply Dot Com every time I open a digest email... Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728 ext. 611 sip:[EMAIL PROTECTED] -Original Message- From: Matt [mailto:[EMAIL PROTECTED] Sent: Thursday, June 02, 2005 8:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VoiPSupply Dot Com He is right Karl. Without the ship-to being on file with the bank.. the company can be held responsible for fraudulant purchases. On 5/31/05, Karl J. Vesterling [EMAIL PROTECTED] wrote: I'm amazed that this thread keeps going... About the claim of Ship-To being on file with bank... CDW doesn't have a problem with it... Ingram Micro doesn't have a problem with it. Merisel doesn't have a problem with it. Digi-Key doesn't have a problem with it... Why would Voip-Supply??? We accept packages every day with the same Ship-To address specified to Voip-Supply... Additional comments dispursed throughout At 02:32 PM 5/27/2005, you wrote: On 5/27/05, Karl J. Vesterling [EMAIL PROTECTED] wrote: At 08:59 AM 5/27/2005, you wrote: [ snip for brevity ] I just wanted to clarify ... this isn't a voipsupply.com problem at all, butrather a courier screwup... which happens anywhere and at anytime... right? TWO screw ups in the shipment. 1.) It was shipped to the Bill-To address. Since there is no one there during the day I had to sit and wait for it lest it not be delivered. This screw up has to do with the person that ordered it, because they didn't have the ship to address on file with their bank. This was not a paypal transaction. The PO had BIG BOLD LETTERS - Ship To: I'm unaware of any practices with the bank that requiring Ship-To addresses to be on file with them. Perhaps your financial institution is a bit different? 2.) when an order is placed on a Tuesday AM (or) Monday PM, and it's priority overnight, and it's across town, and the tracking number was supplied on Wednesday one would expect that it would show up Thursday, not Friday. See above, again this is a screw up that happened because of the one that ordered it, by NOT having the ship to address on file with their bank. Where do you get this Ship-To on file w/ Bank idea? Anyhow, you were already answered before that it had to do because YOU didn't have the address on file with your bank. Why are you repeating this lie that it is voipsupply.coms fault? Be repeating it you make yourself look more like a politician or media person, but certainly not someone that is in the electronic engineering business. No I will not believe it because I read it twice, so stop it. No lie... Fact. There is a difference... So, what we have here is one problem compounded by another, none on behalf of the courier. Exactly, but on behalf of the ordered. If you give a Ship-To address that is NOT on file with your bank, you will NOT get it to that address, and it WILL delay shipping. Gosh dang spin doctors... Where does it state this??? Prove it. Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Telephone: Washington DC: (202) 448-3009 Extension 0 Annapolis MD: (240)
RE: [Asterisk-Users] C files of Asterisk
Modifying the C files of asterisk is very different from modifying the config files. If you are not currently a programmer, this will be a very long and slow process. First, a good understanding of the C language programming statements and language syntax is needed. Next, you will need to learn about the algorithms and data structures of programming. There are many good books available at libraries and book stores which cover these subjects. Next, you will need to decide which features of asterisk need to be modified. Then you will need to read a large portion of the existing asterisk C code to learn how the program works to determine how to make the modifications. Programming is very exciting and rewarding, but requires years of dedication to the subject to become proficient. If you were wanting more immediate results, you may want to hire a programmer that already understands asterisk. -Original Message- From: Bharat M. Sarvan [SMTP:[EMAIL PROTECTED] Sent: Monday, April 25, 2005 12:57 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject:[Asterisk-Users] C files of Asterisk Hello Everybody, I was going thru the C code of Asterisk. Does anybody know how does one go about modifying the C code of Asterisk? Please do reply. Regards, Bharat M. Sarvan EZZI BPO Pvt Ltd., PUNE. File: ATT00250.htmlFile: ATT00251.txt ___ 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] LOOKING TO HIRE
I completely agree with this! This entire argument is like saying that a screwdriver is a better tool than a hammer, and 'good' mechanics use screwdrivers while 'bad' mechanics use hammers. In reality, the 'best' mechanic knows when to use a screwdriver and when to use a hammer. -Original Message- From: Jean-Michel Hiver [SMTP:[EMAIL PROTECTED] Sent: Thursday, May 19, 2005 3:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:Re: [Asterisk-Users] LOOKING TO HIRE Ergo, the assertion Good Programmer = Compiled Languages is *pure bull*. ___ 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 of E1s? (one E1 input -- 2 x E1 output)
Many channel banks have two T-1 connectors and support a feature called 'drop and insert'. This allows some of the DS0 channels to be cross connected from one T-1 connection to the other. The first T-1 connection can go to the telco or an interface card in a computer, and the second T-1 can go to another channel bank. Some of the channels can be dropped off at the first channel bank while the rest can continue on to the second channel bank. You are asking about E-1 and PBX instead of T-1 and channel bank, but if I understand the 'drop and insert' correctly, and if your hardware supports it, this may work for you. Il giorno ven, 29-04-2005 alle 17:33 +0200, [EMAIL PROTECTED] ha scritto: Hi, Assume I have one E1 digium card to which I want to plug two distinct E1 PABXs , one with 15 channels and the other with 15 channels; Is there a sort of E1 multiplexer devise that allows me to plug in one hand the E1 port of the Digium card and on the other hand the two PABXs? In this same devise, I should be able to say that 15 channels need to go to first Interface and 15 other channels need to go to other interface. Or is it necessary to acquire a another E1 card although I don't need to process more channels (30 channels are ok). Any help is greatly appreciated. ___ 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 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-h.323
I believe that you have to start openh323gatekeeper before starting asterisk. Regards, Neal -Original Message- From: [EMAIL PROTECTED] [SMTP:[EMAIL PROTECTED] Sent: Thursday, April 28, 2005 7:05 AM To: asterisk-users@lists.digium.com Subject:[Asterisk-Users] asterisk-h.323 Hi I've a problem with the registration of the openh323gatekeeper. First I've downloaded and installed the pwlib and openh323 libraries successfully. Then I've downloaded the package openh323gk.tar.gz,executed the binary file, but the gatekeeper is not registred on asterisk! Then I've also downloaded and installed the pwlib and openh323 over the Asterisk's pc, and launch the make command in the directory /../asterisk/channels/h323, as suggested by README file, in order to compile h323. I've several compilation errors related on ast_h323.o. Can someone help me about it?Are the installation steps correct? Thanks for all ale ___ 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 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.
Hi, You could try disabling the USB in the BIOS setup to see if it stops sharing the interrupt. Regards, Neal -Original Message- From: Ian Pattison [SMTP:[EMAIL PROTECTED] Sent: Sunday, April 17, 2005 6:52 PM To: asterisk-users@lists.digium.com Subject:Re: [Asterisk-Users] TDM400P Revision question. I don't know how everyone else is doing but my woes are continuing. I'm really starting to dislike these Digium cards. Hardware: Digium TDM400P (REV G according to the silk screening on the board) 2xFX0, 2xFXS purchased in August/September 2004 Dell Precision 420 (PIII-733, 512MB RAM nothing fancy but not doing too much either) Software: Zaptel, Libpri and Asterisk (v1-0) downloaded and re-compiled from CVS today (April 17) SuSE 9.1 (Kernel 2.6.4-52-default) configured as a life-support system for Asterisk only... no other apps running. Here's are my issues: 1. dmesg reports the card as Revision E/F although Rev G visually confirmed (see below) Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 11 for device :03:05.0 PCI: Sharing IRQ 11 with :00:1f.3 Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXO (FCC mode) Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXS/DPO Module 3: Installed -- AUTO FXS/DPO Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) 2. Low ringing voltage still (~44V AC). I have used the boostringer=1 option when loading wcfxs, did I miss something at compile time? 3. Rogue on-hook 109V AC voltage (11V AC off-hook) on both FXS ports. I have conformed that it is being generated by the card itself. I repeat, it is not being induced on the wire. After finding it a the wall jack I was able to sample the same 109V AC at the card itself with no cables attached. 4. Random calls dropped on the FXO ports from both FXS and SIP clients. The drop is usually preceded by a 2-3 second buzzing sound on the line. This occurs with both incoming and outgoing calls. It should be noted that the card is sharing an IRQ with another device (the USB controller to be exact). No matter what slot the card is inserted in it ends up sharing an IRQ. To that end I made sure it was sharing with an unused device (no USB devices attached). Looking for help here... Thanks, Ian ___ 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 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
Hi, It's not probable that the delay is just in the sidetone. It is more probable that the echo is caused by reflected energy somewhere very far downline, perhaps even at the far terminating end of the call. Yes, I know that the person at the far end of the call does not hear an echo, but that doesn't mean that the far end is not the cause of the problem. Think about this: If I scream and the energy is reflected by the wall of a building, I will hear an echo. If you stand at that wall and listen, you will only hear me scream but not the echo. You will never hear an echo at the point where the reflection occurs because there will be no delay at that point. You will only hear an echo somewhere away from the reflection point so that the signal will have some time delay caused by the travel path. If you hear an echo, the reflection is somewhere away from you so that there can be a delay caused by the signal going somewhere and then coming back. Try this experiment: call a number that is answered directly by the asterisk box and see if you get the echo. If you do, it is definitely a local problem. Regards, Neal -Original Message- From: Adam Goryachev [SMTP:[EMAIL PROTECTED] Sent: Tuesday, April 12, 2005 9:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:Re: [Asterisk-Users] Local Echo On Tue, 2005-04-12 at 17:13 -0700, Noah Silverman wrote: Thanks Jeff, Your explanation helps. You are correct. There is delay in the sidetone. It annoys me, but the other party doesn't her it. (You're right that the other party is on a POTS line.) I assume that the echo must be between the SIP phone and Asterisk. Since the actuall call sounds fine to both me and the other party, then the Zapata stuff must be working fine. Right?? No, thats what everyone keeps telling you. Everything is working fine on both ZAP and SIP sides, just that there is some delay, and therefore you hear echo. So start reading the advice that other people are offering. One thing that other people haven't mentioned, is that if you are not in the US, then you should also set the OPERMODE value to your country. Interesting, If I call someone who doesn't pick up right away, I can still hear myself echo really badly if I talk into the phone while it is still ringing at the other end. Does this help?? Strange/interesting, but I personally don't know enough about this to comment further. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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 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: Ebay listing selling Asterisk @ Home and AMPfor over 1000 dollars
Hi, The GPL is about _distributing code_. If someone takes some hardware and installs GPL code and lists the item on ebay, that is not the same as _distributing code_. If someone bids on the item, and the hardware containing the code is shipped to that person, then the code was distributed and the GPL notice must be sent with the code. I don't think there is any obligation to distribute the GPL notice _in advance_ of distributing the actual GPL code. The notice and the code can be distributed together at the same time. The source code must be provided on request, but doesn't have to be shipped with the binaries. Regards, Neal -Original Message- From: Robert Webb [SMTP:[EMAIL PROTECTED] Sent: Tuesday, April 12, 2005 12:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:Re: [Asterisk-Users] RE: Ebay listing selling Asterisk @ Homeand AMPfor over 1000 dollars On Tue, 12 Apr 2005 15:04:26 -0400 David Brodbeck [EMAIL PROTECTED] wrote: -Original Message- From: Jean-Michel Hiver [mailto:[EMAIL PROTECTED] I don't think the GPL obliges you to give credit to anybody. In fact, I think that's a key difference between the GPL and the BSD license. Actually, the GPL does require credit to be given, at least in the sense that the source code is modified and not necessarily in the advertisement as I understand it. Please take a look at the Developers archive as there is an ongoing discussion about another distro that is for Windows that is also dealing with the GPL issue. I am not completely sure that this falls under the exact same category, but I believe it is really close. Just in this case it would probably deal more with AMP than Asterisk directly. But not having this sellers code, I connot confirm nor deny that this is the case. Can anyone chime in on whether or not this seller must provide the source code for what he is selling?? ___ 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 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] sangoma S508/FT1 ISA
Hi, Does anyone have any experience with the Sangoma S508/FT1? I can't seem to find very much information on it, and Sangoma has not responded to my e-mail. The Sangoma wanpipe driver doesn't seem to support TDM on this card, but I feel certain that the card can handle it. I hope someone knows how to make this card work so I don't have to hack the firmware. Regards, Neal ___ 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
