Re: [Asterisk-Users] Re: www.openpbx.org
On Sat, Oct 08, 2005 at 06:50:54PM -0300, Doug Meredith wrote: Dinesh Nair [EMAIL PROTECTED] wrote: too much divergence and we have two pieces of software competing for each other. My guess is that if they succeed, they will diverge significantly. We will have two pieces of software that work with each other at well-defined interfaces. The development of internal workings may diverge. -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: www.openpbx.org
On Sat, Oct 08, 2005 at 10:43:28PM -0400, Paul wrote: Steve Underwood wrote: It's not harder. It's just different. A number of things have similar requirements. The ISDN4Linux folk have certain versions of their software approved by the telecoms bodies in Europe. They need to tie down exactly what was approved, so any other versions emit a notice that says they are unapproved versions. They do this with a signature on the approved version. It seems to work out OK. From the ISDN4Linux FAQs: Actually, since April 2000 the rules for certification have changed. Now the producer of an ISDN card has to do only hardware tests, the driver is not part of the certification anymore. This applies to the whole European Community. http://www.isdn4linux.de/faq/i4lfaq-25.html If this is true then perhaps the ruling telecoms have improved their protocol violation defenses and dispensed with the certification process. This would be a Good Thing (tm). I think that the important thing to remember is that a good reverse engineer can take the object code from a rom and produce source files that are better commented than the original source ever was. Reversers are mundane scribes and relics. Their services were valued in the past when software was expensive and poor quality. Free and competitively valued software has devalued their efforts. Besides, it's hard to build a community or support organization around stolen merchandise. Further evidence of their insignificance is the lack of coverage by the media. -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: www.openpbx.org
On Sun, Oct 09, 2005 at 01:51:41PM -0400, Paul wrote: Mike M wrote: Mike, the context was regarding security by obscurity. It has nothing to do with stealing a product to sell to others. The only reverse engineering I ever did had nothing at all to do with bootlegging or counterfeiting software. The closest I ever came to that was reversal for the purpose of proving it contained stolen goods. By the way, I am not a mundane scribe or a relic by any means. Closest I ever came to being a scribe is putting a signature of mine in pcb copper and some silicon. I also left my signature in the leftover gates of some array logic. Calling me a scribe or relic is a rather hefty insult, don't you think? The context of reversing was difficult to discern from repeated readings. The message seemed to be to not bother closing software because it can be reversed easily and the source can be better than the original. I supposed you were describing hypothetical abstract possibilites and not actual occurences. My responses were similarly abstract. I admit there can be legally justifiable reasons for reversing, or that it could be a form of archaelogy, but the original statement did not suggest these cases. Now that your context, meaning, and intent are clearly defined, it's evident you should not take umbrage with the description of reversers as scribes and relics as those terms do not apply to you. Besides, illegitimate reversers can't complain about being insulted because they run the risk of being exposed. And then their contacts can be investigated for possible license violations. Reversing to exploit security weakness is most likely very effective. I agree with you that securing by keeping software closed is folly. Opening the software does not make it secure either. I return to my original point: Keeping software closed is done only when you can't figure out how to have it open. The point that launched this sub-discussion was that Asterisk has a dual license and OpenPBX does not. The underlying assumption is that the commercial license for Asterisk is for a closed source super-implementation of the project. Could this be a competitive advantage? As you point out, there are certainly no security advantages. There could be some commercial advantages that currently exist for Asterisk that might be altered with the presence of OpenPBX. -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: www.openpbx.org
On Sat, Oct 08, 2005 at 09:20:07AM -0400, Paul wrote: I find that amusing. I have a lot of experience with disassembly. I have even reverse-engineered machine language code that ran on custom processors which means you have to reverse-engineer the instruction set as part of the task. I think your argument is: Don't require or offer closed source applications since they can be cracked. Similarly we shouldn't lock our doors when we leave home because they can be overridden. Locks, like closed source, are legal barriers that work most of the time for their intended purpose. The discussion of licensing issues on forking Asterisk should assume everyone understands and follows the applicable legal guidelines on software licensing. The earlier point was Asterisk with its commercial license option, and presumably closed source traits, will be required some situations. Having closed source as component of a certified solution is topical ointment enjoyed by purveyors of certificates. If this is true then OpenPBX, lacking a similar license option could be at a competitive disadvantage. But what if OpenPBX attains features that are desireable but uncertifiable because the closed source option does not exist? Then we'll be living in interesting times ( http://www.noblenet.org/reference/inter.htm ). Closed source might delay the cracker but it also delays pre-crack and post-crack countermeasures. What's the alternative? Open source? Cracking is unnecessary with open source. -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: www.openpbx.org
On Fri, Oct 07, 2005 at 10:51:45AM -0400, Brian C. Fertig wrote: Can they do this? Is this legal? Google fork open source. -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: www.openpbx.org
On Fri, Oct 07, 2005 at 09:45:53PM -0400, Paul wrote: Doug Meredith wrote: Also consider that there are situations where 100% open source is never allowed. Check out visa/mastercard processor certification for a good example. Digium dual licensing availability means I could actually stand a chance of using asterisk as the basis for systems used by military and law enforcement in applications that require extremely high security. There is a popular vendor of closed source products whose security has been compromised often. The security of OpenSSH is well established. Reading this list iwe learn that the open source version of Asterisk is currently being used by military personnel. Asterisk offers ways for users to implement eavesdropping applications which undermines the goal of attaining extremely high security. Open source is for sharing if that's feasible and closed source is not. Dual licensing is for both. -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM
On Sun, Sep 11, 2005 at 11:02:56AM -0700, Derek Whitten wrote: On Sun, 2005-09-11 at 08:44, Mike M wrote: On Sat, Sep 10, 2005 at 04:43:26PM -0700, Chris Travers wrote: Mark Phillips wrote: The suggestion that I have is for various areas to have dedicated civil emergency com units with strategic reserves of fuel (3-4 weeks worth), battery backups, etc. These units would have links (fiber, microwave, and/or satellite, better to pick 2 of 3) to areas outside expected disaster zones. Asterisk could then run across these links. (Sattelite links would best be POTS-type). Great suggestions but these are out of the realm of what a community of individuals can do. I'm thinking about what I as an individual am capable of. These are great suggestions and I believe that it IS in the realm of 'what a community of individuals can do' .. It just depends on the community of individuals involved with the project.. Not to disparage you excellent ideas, but me-myself-and-I cannot marshall a fiber optic link or microwave shot. I'm thinking along the lines of maybe 4 times a year, heading out to a remote area and calling people over a ham radio for practice. I envision this requiring coordination with several other people at most. Being ad hoc, adaptable, quick, mobile, and red-tape-less are my goals. my 0.02 Worth a good bit more than that, indeed. -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM
On Sun, Sep 11, 2005 at 02:45:47PM -0700, Michael D Schelin wrote: BUT, let me tell you about how bad the southern CA. radio site owners are becoming. We had a 4 day outage at a very large site where one of my radios is located. None of them care anymore about backup power. This happened this past week. We took up our own Generator because the site owner (a national site company) won't maintain an old one. My friend (a microwave isp ) fixed the site owners by adding oil and a new battery. That will take us out! Over the weekend I heard an account of the communications breakdown in New Orleans. Ham radio effectiveness was diminished because of a shortage of pre-positioned generators. What I've learned from this thread is that integrating Asterisk and ham radio is feasible and potentially useful. I also learned that essential infrastructure - repeaters - are softening assets. What's more, google hits for asterisk ham have increased significantly and become useful. This thread comes up #3 on Google. Thanks everyone. -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM
On Sat, Sep 10, 2005 at 04:43:26PM -0700, Chris Travers wrote: Mark Phillips wrote: The suggestion that I have is for various areas to have dedicated civil emergency com units with strategic reserves of fuel (3-4 weeks worth), battery backups, etc. These units would have links (fiber, microwave, and/or satellite, better to pick 2 of 3) to areas outside expected disaster zones. Asterisk could then run across these links. (Sattelite links would best be POTS-type). The point is to a disaster-tolerant communications infrastructure which could then be used to to provide additional communications services to the relief workers. With various point to point wireless capabilities, it might be possible to use them to provide cell service to relief workers etc through the installation of GSM microcells (which could be brought in after the fact). See where I am going? Great suggestions but these are out of the realm of what a community of individuals can do. I'm thinking about what I as an individual am capable of. -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [Asterisk-Users] civil emergency comms: Asterisk + HAM
On Sat, Sep 10, 2005 at 09:08:53AM +0200, [EMAIL PROTECTED] wrote: Hello, Asterisk is on the air : http://www.hamwlan.net http://192.168.1.1/HamWlan.htm (see the second drawing) 73 ! F6HQZ, Francois BERGERET, France. Excellent. So you have SIP/IAX clients connecting to a router over HAM radio links, and the router is on a WLAN with an Asterisk box ( 44.151.177.66 : serveur Asterisk (PBX VoIP : SIP/H.323/IAX))? What sort of bandwidth is available on the hamwlan? I tried several different character encoding choices and I just couldn't get the proper representaions for the characters on the web page. Can you recommend an appropriate character set for Firefox for French? Babelfish will probably work better if I used the correct character set. Thanks, -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : RE : [Asterisk-Users] civil emergency comms: Asterisk + HAM
On Sat, Sep 10, 2005 at 11:18:09PM +0200, [EMAIL PROTECTED] wrote: Oops ! Sorry ! It seems that I have forgotten to replace my french characters as ? by the correct sequence eacute; as exemple. I have just modify this page and you can probably read it now (but it is in french only for now, I promess to translate this pages this next cold season). I think I need to research character sets and possible load some that I do not have. I have started some VPN under IPSec to separate public traffic from HAM's traffic as lawyers said in near all the countries. HAM traffic must not be mixed with Internet? As I am self training on Asterisk from monthes and use one at my home for my own private telephone lines, I have think that it could be nice to connect my Asterisk box to my HamWlan network (without any telephone access because it is forbidden in France). Interesting. It's OK to have VoIP over Wi-Fi, and VoIP over HAM, but not PSTN over HAM? Of course, you could enable POTS on your ASterisk box and simulate PSTN calls over HAM. Maybe you could get permission to demonstrate by contacting some disaster relief agencies to show them some possibilities. I am just starting to tell to some HAMs to join me and start some experimentations to see if Asterisk could be interesting for HAM use. HAMs are already using some kind of Internet VoIP as Skype or Echolink are (Echolink is a HAM network connecting people and radio equipments). With Asterisk, we can use conference rooms (mine is [EMAIL PROTECTED]) or to share an UHF repeater linked to a room or a specific number. I have not enougth bandwith as I desire... I have two providers and the best is about 2.6 Mbs download and 650 kbs upload. The ideal way could be to place an asterisk in a ITSP white room with bigger bandwidth, but it is a dream only :-) What is your bandwidth on the HAM radio links? Is that even a good question? For now, it is only the beginning, and I play to see if any HAM's interest. Please keep the list and/or me personally informed. I have a keen interest in what you are doing. -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM
On Thu, Sep 08, 2005 at 07:28:34PM +, Mike Hemstock wrote: On Tuesday 06 September 2005 15:27, Mike M wrote: Imagine what a network of systems composed of Asterisk, ham radio, wifi, generators, batteries, and a reserve of fuel could have done for the Gulf coast. I have all of the components above except the ham radio. That's a very interesting idea. I've initiated a request to join my local amateur radio yahoo group. I'm going to see if I can enlist help to demonstrate this idea. -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM
On Fri, Sep 09, 2005 at 01:46:57PM +0100, Peter Bowyer wrote: On 09/09/05, Mike M [EMAIL PROTECTED] wrote: On Thu, Sep 08, 2005 at 07:28:34PM +, Mike Hemstock wrote: On Tuesday 06 September 2005 15:27, Mike M wrote: Imagine what a network of systems composed of Asterisk, ham radio, wifi, generators, batteries, and a reserve of fuel could have done for the Gulf coast. I have all of the components above except the ham radio. That's a very interesting idea. I've initiated a request to join my local amateur radio yahoo group. I'm going to see if I can enlist help to demonstrate this idea. The concept of combining VoIP and ham radio is by no means new - there are many skype-a-like systems around which are used as links or user access to the existing ham repeater network. I don't know of any using Asterisk, though. I think this architecture has value: PSTN---asterisk---voip---radio===+==radio--voip--asterisk---POTS +==radio--voip--asterisk---POTS +==radio--voip--asterisk---POTS and this too: voip svc prvdrvoip---radio===+==radio--voip--asterisk---POTS +==radio--voip--asterisk---POTS +==radio--voip--asterisk---POTS POTS at the emergency end is good because it's familiar, simple, cheap, and runs on a central power source. I don't know radio equipment so I don't know if the upstream radio can multiplex streams onto different frequencies. -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] civil emergency comms: Asterisk + HAM
On Fri, Sep 09, 2005 at 09:31:06AM -0400, Mark Phillips wrote: Generators require fuel which is always in short supply and batteries die out quickly. Fuel and batteries and power efficient systems need planning and management. Don't overlook solar panels as an energy source. They need to be in place all over the country and tested frequently. Adding Ham Radio to the picture doesn't really add much when you are trying to do something like a * network. The radio gear just isn't designed to integrate with the * server. It's software. It can be changed and added to. These things evolve from ideas in discussions like these. Such technologies, whilst legal here in the US, may not be legal elsewhere. What about authorized looting you mentioned? Sometimes you have to take a risk. Develop and demo where it's legal first. If it's not legal than we should ask why and work for change if we don't like the answer. Without question a phone system would be much better than a radio station. Well said. I guess after all this waffle I'm trying to say that ham radio is not a replacement for the telephone and cannot handle the kinds of load that is required by a phone system. What is the bandwidth potential? There are compression techniques from VoIP that might improve radio bandwidth utilization. New protocols can evolve to conserve bandwidth. Load control is a manageable problem. Radio telephony is not new. Telephony over ham might be new only because Asterisk puts telephonyi/voip into the same price range as ham radio gear. Maybe HAM is not the best technology. Maybe wi-fi is what we need. http://www.oreillynet.com/cs/weblog/view/wlg/448 Grassroots engineering can create an emergency civil communications system thereby creating some stored luck. Lucille Ball said, Luck? I don't know anything about luck. I've never banked on it, and I'm afraid of people who do. Luck to me is something else: Hard work -- and realizing what is opportunity and what isn't. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] civil emergency comms: Asterisk + HAM
The disaster in the Gulf coast and the less than optimal initial response suggests to me that citizens must shoulder more responsibility for emergency management. Communications loss must have played a large role in the failures that occurred. I can't help but wonder if there are fewer ham radio operators today and that if there were more, maybe they could make a difference in future emergency situations. Imagine what a network of systems composed of Asterisk, ham radio, wifi, generators, batteries, and a reserve of fuel could have done for the Gulf coast. I have all of the components above except the ham radio. I suspect there are some folks on this list that have already implemented such a system. If so, I would like to read about what they have done so I can develop a plan to participate in this network if one exists. There's not much on google for asterisk ham. http://sourceforge.net/projects/hamlib/ http://www.radioadv.com/default.htm Thanks, -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Community Participant; Katrina Refugee
On Tue, Sep 06, 2005 at 09:07:36AM -0700, Karl S. Katzke wrote: Just don't do what one of my friends did, and fed-ex his backup CDs to his mom in Biloxi ... who kept them in a box on a shelf out back in the shed. After the storm, they found precisely half of one... embedded in the siding on the house... Got to choose the off-site location such that 99% of disasters will not simultaneously affect all locations. -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Community Participant; Katrina Refugee
On Sun, Sep 04, 2005 at 01:04:41AM -0400, JR Richardson wrote: If anyone has any trade secrets on successfully recovering waterlogged electronic equipment, please let me know. Glad that you're safe. We're still waiting to hear about some of our people. Notes of encouragement: 1) DMS-10s have been through floods, hosed off, and made to operate again. 2) I sent an Audiovox cell phone to the 12ft end of a swimming pool. One of the life guards brought it to me a while later. I washed it and dried it and used it for another year. 3) If you have a really valuable HDD then maybe its worth taking it to a recovery specialist. Note of discouragement: 1) My Motorola flip-phone fell into the toilet. I washed it and dried it and it started working - sort of. It has water tattle-tale stickers inside. Maybe they designed it to stop working when it gets wet :). Note to us all: We should do off-site backups. Encrypt sensitive stuff. Send copies to relatives and friends for safe keeping. I know I've been putting it off for 5 years now. -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma card problem with EWSD Exchange
On Thu, Sep 01, 2005 at 01:24:02AM -0500, Kristian Kielhofner wrote: Nguyen Trung Tin wrote: I'm using sangoma card A-101. tested successful with E10 (ACATEL) Exchange, connection with E1, CAS, (using unicall-0.0.3pre4). my system run success, incoming call and call out are good. when i switch to EWSD (SIEMENS) R-15 . my asterisk faill, cannot connect with EWSD. Have you tried contacting Sangoma? I forwarded this to them. -- Mike ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Validating a phone number
On Sun, Jul 17, 2005 at 12:15:06AM -0700, trixter http://www.0xdecafbad.com wrote: In short you might investigate a phone company service blocker for premium service numbers and try your best to block what you can but it would be impossible for someone without SS7 network access to see what the rate of the call is since these numbers can hide virtually anywhere. The cost of a call is not available from any SS7 service that I know of. NANPA manages all the numbers in the north american numbering plan, if memory serves their page is nanpa.org and they used to have rate center information available on their page for free that you can download (and you would need to parse it and continually get updates as new exchanges are allocated). http://www.nanpa.com/area_codes/index.html You'll have to make a white list. On athe same topic, I'm worried about area codes like 809. Are there any other such area codes that should be avoided? Ahh glad you brought that up, see above. I think there are a couple of them, but I dont know off hand what they are.. try googling 'toll fraud 809' and see if that works. 809 is a valid area code. Read this: http://www.lincmad.com/telesleaze.html ___ 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 A104c vs. A104u
On Thu, Jul 14, 2005 at 10:46:21AM +0100, Gavin Hamill wrote: Hi, Just a quickie - if I want to implement an * solution purely for voice (well, and physical fax machines / dialup modems..) on EuroISDN E1s, is there any benefit to the A104u over the A104c? I'm just trying to decide if the extra ?200 for the A104u is worth it :) Isn't it the other way around? c u? The c version has channelized HDLC which means board does HDLC instead of main CPU. Less bit banging on main CPU is good. 200 of what currency? http://en.wikipedia.org/wiki/ISO_4217 -- Mike ___ 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] HDLC bad FCS
Comments throughout. -- Mike On Mon, Jul 04, 2005 at 08:23:55PM -0400, Carlos Alperin wrote: That has nothing to do with the fact that you 're going to check what is going on on that line. You're going to check BER (Bit Error Rate) to see if you have line problems. FCS is indication of frame checksum error which can be caused by bit errors. BERT is a level 1 test. When I asked you for location, I was asking if your Asterisk box was in your Computer Room, away from your TELCO provider. If you have everything on the same location, then much better. Now you mention a DMS-100, is that the Siemens one? Nortel: DMS-100 Siemens: EWSD If that is the case, better because the line analyzer is a function on that switch. Carlos -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Monday, July 04, 2005 8:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] HDLC bad FCS I think I need some help here. I didn't understand much of what you just said there. I though HDLC was a Layer 2 protocol? How can it have a location? FCS is a frame check sequence.. right? So I'm getting data out of sequence (or frames are missing from the sequence?) My gear is in a data centre, with the DMS-100 switch in the next room. Does this help? It simplifies the troubleshooting scenario. At least you don't have outside plant support ignoring your problem. What is BER? Bit Error Rate: bi-polar violations, jitter, etc.; put circuit into loop-around; tester generates and receives pattern and indicates sent versus received results; try it if you have the equipment and cooperation of the DMS handlers Carlos Alperin wrote: I believe that you need to analyze the packets at your provider site. They should be able to do that. Is your HDLC located on your location or on your provider. This test should be done where Asterisk is running, because is where the problem is reported. Start to look for Line Analyzers for HDLC, in order to check BER. If BER is high enough, then the problem is internal on your server. Regards, Carlos Alperin Senior System Engineer Seneca Communications, LLC [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Monday, July 04, 2005 7:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] HDLC bad FCS I have 2 servers, configured identically. Each has a TE405P and 2 PRIs. One server was experiencing crackly audio on one circuit, accompanied by HDLC bad FCS messages. The telco recabled and moved me to another port on the DMS-100. The audio is better, but there are still bad FCS problems on the span. I have moved the PRI in question to the other server, and the problem does indeed move with the circuit. That's pretty damning in my opinion. Did you do this demonstration for the DMS handlers? There are no zaptel timing/interrupt problems present on either server. The fact that 3 PRIs are error free and that the problem moves with the circuit tells me that there is still a problem on the circuit. The telco believes that there is nothing more that they can do (provision a complete new circuit?). Provisioning a new circuit sounds like a good idea. You might ask to see the DMS config tables for all four circuits. Keep an eye on clocking choices. I don't get HDLC aborts, so the problem may not be _that_ serious. Does anyone have any comments? Would a newer (unstable) version of Zaptel drivers help? Would line-build-out parameter in zaptel.conf make any difference? LBO is probably not going to help. Your 3 other circuits work fine and are probably cabled over the same distance. Use lowest value LBO. I would not settle for a circuit that regulary pops FCS errors. In a close connection like you have FCS should occur rarely. You are at the end of the food chain so you'll have to prove the DMS is screwing up. Trying running your PRI circuits back-to-back to further ensure that your equipment is operating correctly. You'll have to show the DMS handlers your demonstrations and gently ask for help. You'll have to help them establish confidence in your equipment. You should also use the cables from the 3 good circuits to connect to the suspected bad port of the DMS. Use the cable on the bad circuit to connect good DMS ports to *. This might help point the finger at a bad DMS port or at a bad cable. Make a spreadsheet of various combinations of DMS ports, * ports, and cables. Enlist DMS handlers to help cycle through the variations. Good luck, -- Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit:
Re: [Asterisk-Users] HDLC bad FCS
On Wed, Jul 06, 2005 at 07:31:30AM +1000, Rod Bacon wrote: Thankyou for an excellent post. I hope it helps. I put up SS7 circuits and I'm starting to put up PRI circuits so I recognized what you were describing. If you think of it, post your progress or success. -- Mike ___ 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] FXS interfaces
On Wed, Jun 22, 2005 at 02:58:54PM -0400, Jerry wrote: Mike M [EMAIL PROTECTED] wrote: Think opposite. Green modules are fxs and should be handled with the fxo signaling. Red modules are fxo and should be handled with fxs signaling. Note the red and green colors here: http://www.digium.com/index.php?menu=fxsvfxo The opposite thing is hard to catch on to at first, but once you get the hang of it isn't so bad. Plus, the various drivers try to help out with hints if they think the signalling is wrong. commheads should be accustomed to this sort of thing. DTE/DCE is an old one that is similar. Become one with the confusion. _Don't_ plug a phone into a red module jack. _Don't_ plug a PSTN line into a green module jack. Silly question: Can you connect an FXO port to an FXS port? Doing this for inter-server exchange would be ugly, but I'm thinking that it might be useful for testing or experimenting. (Maybe not, because it's not simulating a true CO interface, but I was wondering if it was possible). You try it first :-). Sounds logical. As an aside: Why can't you plug a phone into a red module jack? It won't work, but is there anything harmful there? (Not plugging the PSTN into the green jacks is obvious -- the ringing voltage on the PSTN will probably damage the FXS card). Yeah. I thought about that too. I decided it was just good policy to use the rules above. Here's another question: If you plug the PSTN wire into a green module is there a fire risk? I don't know if the Digium equipment has CE/UL ratings. If they don't then it may be worth elevating caution to concern for fire safety. -- Mike Thanks, J. ___ 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] FXS interfaces
On Wed, Jun 22, 2005 at 06:22:57PM -0400, Jerry wrote: Hi Alessandro, But all ports are green! p1 -green p2 - green p3 - green p4 - green I think he means the daughter card color, not the LED on the card slot. What color are the actual daughter cards? Indeed, I was referring to daughter color and _not_ LED color. But the per-port LED color is a logical mix-up. Sorry for not being more specific. (I hope Digium maintains the color of daughter cards differences. I think it's a brilliant device. It might even be the reason for the price difference between the daughters. From a marketing point of view they should sell both cards for the higher of the two prices :-). -- Mike ___ 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] FXS interfaces
On Wed, Jun 22, 2005 at 10:41:18AM -0300, Alessandro wrote: Does somebody know why no load modules to FXS? I used zaptel-1.0.7 version driver. [EMAIL PROTECTED] zaptel-1.0.7]# modprobe wctdm Here's a clue: ZT_CHANCONFIG failed on channel 3: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? /lib/modules/2.4.21-27.EL/misc/wcfxs.o: post-install wcfxs failed /lib/modules/2.4.21-27.EL/misc/wcfxs.o: insmod wctdm failed You have new mail in /var/spool/mail/root snip It's TDM22B device. It's got 4 modules. What color are the modules in positions 1, 2, 3, 4 on the TDM400P card? Don't be confused by the 0-3 numbering, just add 1. See below zaptel.conf: Just the relevant sections next time :). (The lines not starting with '#'.) Think opposite. Green modules are fxs and should be handled with the fxo signaling. Red modules are fxo and should be handled with fxs signaling. Note the red and green colors here: http://www.digium.com/index.php?menu=fxsvfxo fxsks=1,2 fxoks=3,4 For TDM22B: fxsks=(position of red module)(position of red module) fxoks=(position of green module)(position of green module) _Don't_ plug a phone into a red module jack. _Don't_ plug a PSTN line into a green module jack. -- Mike ___ 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 Asterisk Implementation
On Wed, Jun 22, 2005 at 09:50:42AM -0500, Don Brearley wrote: I've been planning to replace my aging CENTREX switch with a new PBX and am seriously considering Asterisk as my solution. You have a CENTREX switch? I'm confused. Do you have a PBX? Do you pay for Centrex services from your phone company? Do you have a Class 5 swith with Centrex features? Why not take a cautious approach? Leave what you have in place. Install Asterisk and investigate the various line interface options. Enlist early adopters on your campus to participate in the trial. Connect Asterisk to your existing system with PRI. Gradually ramp up the Asterisk system and ramp down the existing system. -- Mike ___ 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] FXS interfaces
On Wed, Jun 22, 2005 at 01:03:33PM -0300, Alessandro wrote: Mike, It's got 4 modules. What color are the modules in positions 1, 2, 3, 4 on the TDM400P card? Don't be confused by the 0-3 numbering, just add 1. The colors in positions 3 and 4 are green, 1 and 2 light is off. You should double check what you are reporting. I deliberately reversed the zaptel settings on my box and got this: [EMAIL PROTECTED] src]# modprobe wctdm ZT_CHANCONFIG failed on channel 25: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? /lib/modules/2.4.21-27.0.1.EL/misc/wctdm.o: post-install wctdm failed /lib/modules/2.4.21-27.0.1.EL/misc/wctdm.o: insmod wctdm failed [EMAIL PROTECTED] src]# Jun 22 15:11:31 b2 kernel: PCI: Found IRQ 11 for device 01:0a.0 Jun 22 15:11:31 b2 kernel: Freshmaker version: 71 Jun 22 15:11:31 b2 kernel: Freshmaker passed register test Jun 22 15:11:31 b2 kernel: Module 0: Installed -- AUTO FXS/DPO Jun 22 15:11:31 b2 kernel: Module 1: Not installed Jun 22 15:11:31 b2 kernel: Module 2: Not installed Jun 22 15:11:31 b2 kernel: Module 3: Not installed Jun 22 15:11:31 b2 kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Look familiar? ZT_CHANCONFIG failed on channel 3: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? /lib/modules/2.4.21-27.EL/misc/wcfxs.o: post-install wcfxs failed /lib/modules/2.4.21-27.EL/misc/wcfxs.o: insmod wctdm failed Then I corrected my zaptel.conf and got this: [EMAIL PROTECTED] src]# modprobe wctdm [EMAIL PROTECTED] src]# Jun 22 15:14:14 b2 kernel: PCI: Found IRQ 11 for device 01:0a.0 Jun 22 15:14:14 b2 kernel: Freshmaker version: 71 Jun 22 15:14:14 b2 kernel: Freshmaker passed register test Jun 22 15:14:14 b2 kernel: Module 0: Installed -- AUTO FXS/DPO Jun 22 15:14:14 b2 kernel: Module 1: Not installed Jun 22 15:14:14 b2 kernel: Module 2: Not installed Jun 22 15:14:14 b2 kernel: Module 3: Not installed Jun 22 15:14:14 b2 kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Jun 22 15:14:14 b2 kernel: Registered tone zone 0 (United States / North America) Everything works. Your system is telling you that you have nothing in position 1 and 2, and red modules in position 3 and 4: Jun 21 19:06:16 darthvaden kernel: PCI: Sharing IRQ 9 with 00:1f.5 Jun 21 19:06:16 darthvaden kernel: Freshmaker version: 71 Jun 21 19:06:16 darthvaden kernel: Freshmaker passed register test Jun 21 19:06:16 darthvaden kernel: ProSLIC 3210 version 2 is too old Jun 21 19:06:16 darthvaden kernel: Module 0: Not installed Jun 21 19:06:16 darthvaden kernel: ProSLIC 3210 version 2 is too old Jun 21 19:06:16 darthvaden kernel: Module 1: Not installed Jun 21 19:06:16 darthvaden kernel: Module 2: Installed -- AUTO FXO (FCC mode) Jun 21 19:06:16 darthvaden kernel: Module 3: Installed -- AUTO FXO (FCC mode) Jun 21 19:06:16 darthvaden kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Try changing your zaptel.conf to: fxoks=1,2 fxsks=3,4 You should install your green modules in position 1 and 2. Position 1 is closest to the connectors. Be sure of the following: p1 - green (closest to the connectors) p2 - green p3 - red p4 - red If you still have problems after double checking everything you may need to get help from the mfr or distributor. I had 2 RMAs on a recent order. -- Mike ___ 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] FXS interfaces
On Wed, Jun 22, 2005 at 05:19:47PM -0300, Alessandro wrote: Mike, I got current stable release in CVS repository, and I think that Ok. See below: /var/log/messages Jun 22 17:04:35 darthvaden kernel: PCI: Found IRQ 9 for device 02:09.0 Jun 22 17:04:35 darthvaden kernel: PCI: Sharing IRQ 9 with 00:1f.5 Jun 22 17:04:35 darthvaden kernel: Freshmaker version: 71 Jun 22 17:04:35 darthvaden kernel: Freshmaker passed register test Jun 22 17:04:35 darthvaden kernel: Module 0: Installed -- AUTO FXS/DPO Jun 22 17:04:35 darthvaden kernel: Module 1: Installed -- AUTO FXS/DPO Jun 22 17:04:35 darthvaden kernel: Module 2: Installed -- AUTO FXO (FCC mode) Jun 22 17:04:35 darthvaden kernel: Module 3: Installed -- AUTO FXO (FCC mode) Jun 22 17:04:35 darthvaden kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Jun 22 17:04:35 darthvaden kernel: Registered tone zone 0 (United States / North America) Congratulations. What were you using prior your pull from CVS? Maybe something old that didn't recognize the the TDM400P and its daughters? But all ports are green! Really? Maybe they aren't making the RED FXO cards anymore. You should look at them carefully for p/n differences and not rely on colors. The zapel driver tells you what you need to know too. p1 - green p2 - green p3 - green p4 - green I wonder if Digium will update their website? It's got a strong commitment to red FXO modules in the graphics. -- Mike ___ 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 can you check that eg TDM04B hardware installed and drivers OK
On Tue, Jun 21, 2005 at 07:09:46AM -0600, Rich Adamson wrote: I am struggling to get my TDM04B working. Just to rule out a hardware problem how can I check that the hardware works? How can I then check that the drivers are loaded correctly? 1. from the linux command line, type 'dmesg' and look for Found a Wildcard TDM: Wildcard TDM400P REV H (4 modules) if you see that, the TDM card is recognized by the OS. Here's what I get on a working system: [EMAIL PROTECTED] src]# modprobe wctdm [EMAIL PROTECTED] src]# Jun 21 10:10:55 b2 kernel: PCI: Found IRQ 11 for device 01:0a.0 Jun 21 10:10:55 b2 kernel: Freshmaker version: 71 Jun 21 10:10:55 b2 kernel: Freshmaker passed register test Jun 21 10:10:55 b2 kernel: Module 0: Installed -- AUTO FXS/DPO Jun 21 10:10:55 b2 kernel: Module 1: Not installed Jun 21 10:10:55 b2 kernel: Module 2: Not installed Jun 21 10:10:55 b2 kernel: Module 3: Not installed Jun 21 10:10:55 b2 kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Jun 21 10:10:55 b2 kernel: Registered tone zone 0 (United States / North America) [EMAIL PROTECTED] src]# cat /proc/interrupts CPU0 0: 17893766 XT-PIC timer 1: 2 XT-PIC keyboard 2: 0 XT-PIC cascade 4: 357411641 XT-PIC eth0, wanpipe1 8: 1 XT-PIC rtc 10:3381408 XT-PIC Intel ICH2 11: 178236906 XT-PIC wctdm 14: 50492 XT-PIC ide0 15: 0 XT-PIC ide1 NMI: 0 ERR: 0 [EMAIL PROTECTED] src]# cat /proc/interrupts CPU0 0: 17894203 XT-PIC timer 1: 2 XT-PIC keyboard 2: 0 XT-PIC cascade 4: 357419974 XT-PIC eth0, wanpipe1 8: 1 XT-PIC rtc 10:3381408 XT-PIC Intel ICH2 11: 178241275 XT-PIC wctdm 14: 50494 XT-PIC ide0 15: 0 XT-PIC ide1 NMI: 0 ERR: 0 -- Mike ___ 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] modprobe wctdm waiting for ever
On Tue, Jun 21, 2005 at 10:47:59AM -0300, Edgardo Lust wrote: Hi, I have a pentium 4 with Intel motherboard and one TDM400P (2 fxs, 2fxo) modprobe zaptel is Ok but When I execute modprobe wctdm never load the module, I can wait for 1 year but never response me (error or OK). I need to do ctrl+c This sounds like a problem I had. Some documentation out there is not up to date with the newer line interface cards. Try starting over: modprobe -r wctdm modprobe -r zaptel Then just do: modprobe wctdm -- Mike ___ 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] FXS
On Tue, Jun 21, 2005 at 08:09:52PM -0300, Alessandro wrote: Does Somebody know why no load modules to FXS? I used zaptel-1.0.7 version. What color are the modules? Post your /etc/zaptel.conf -- Mike ___ 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] app_curl.so: Can't locate module sound-service-0-3
Hi, What does an't locate module sound-service-0-3 mean in the context below? I don't use curl as far as I know. Should I noload it in modules.conf? [app_curl.so]Jun 19 08:41:25 b2 kernel: Registered tone zone 0 (United States / North America) Jun 19 08:41:34 b2 kernel: Intel 810 + AC97 Audio, version 1.01, 02:25:38 Dec 24 2004 Jun 19 08:41:34 b2 kernel: PCI: Found IRQ 10 for device 00:1f.5 Jun 19 08:41:34 b2 kernel: PCI: Sharing IRQ 10 with 00:1f.3 Jun 19 08:41:34 b2 kernel: i810: Intel ICH2 found at IO 0xdc40 and 0xd800, MEM 0x and 0x, IRQ 10 Jun 19 08:41:34 b2 kernel: i810_audio: Audio Controller supports 6 channels. Jun 19 08:41:34 b2 kernel: i810_audio: Defaulting to base 2 channel mode. Jun 19 08:41:34 b2 kernel: i810_audio: Resetting connection 0 Jun 19 08:41:34 b2 kernel: ac97_codec: AC97 Audio codec, id: ADS96 (Analog Devices AD1885) Jun 19 08:41:34 b2 kernel: i810_audio: AC'97 codec 0 Unable to map surround DAC's (or DAC's not present), total channels = 2 Jun 19 08:41:34 b2 kernel: i810_audio: setting clocking to 41349 Jun 19 08:41:34 b2 kernel: i810_audio: ftsodell is now a deprecated option. Jun 19 08:41:34 b2 modprobe: modprobe: Can't locate module sound-service-0-3 = (Load external URL) == Registered application 'Curl' -- Mike ___ 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] Best platform
On Tue, Jun 14, 2005 at 03:06:52PM +0300, Tzafrir Cohen wrote: On Sat, Jun 11, 2005 at 08:19:58AM -0400, Mike M wrote: On Sat, Jun 11, 2005 at 02:03:59PM +0200, Michiel van Baak wrote: On 13:22, Sat 11 Jun 05, Serge Schumacher wrote: What platform should you suggest to use asterisk ? I love the way the Debian updates work. Me too, but has the installation improved with the latest Sarge release? It sure has. The installer generally automates most of the necessary tasks. I'll definitely try it out soon. The announcement claims there are improvements. Debian has been extremely slow to improve it installer. Not improvements. A total rewrite, and a very good one. The new installer is much better. Not to mention much more modular. It does increase the memory requirements to 24MB, but I hope most folks here can live with that. Nuts! My CPU with 16Mb is obsolete now. Memory requrements for an installer are usually for a large enough ramdisk that is extracted before a swap partition on the disk is availble. Naturally you can always manually install a system using a chroot from another system. I used CentOS 3.4 on two recent Asterisk installs with no problems. But why would an asterisk installation require so many manual steps? (if there are manbual steps, you will get some of them wrong). The difficulty is self-inflicted: using GRUB instead of LILO and a desire to have a minimal load. Building Asterisk from a minimal load let me get a first hand look at all the dependencies. It boosted my understanding of the system. I can't say I recommend the procedure to anyone needing to build more than one or two systems. Thanks, -- Mike ___ 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 trouble
On Sun, Jun 12, 2005 at 06:10:53AM -0400, Michael Di Martino wrote: However i still get the same error. Please help we cannot connect call form my norstar to asterisk w/ it dropping in 10 seconds. Jun 12 10:51:51 NOTICE[213005]: PRI got event: 5 on Primary D-channel of span 2 Jun 12 10:51:51 WARNING[213005]: No D-channels available! Using Primary on channel anyway 48!Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 35 Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 36 Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 37 Jun 12 10:51:51 NOTICE[262160]: Alarm cleared on channel 38 Hmmm. Links works and suddenly stops. No changes to either side both of which you control. To test the cable idea someone had, pull the cable and see if messages are the same or different. If the same, then it's likely the cable that's the problem. If different, the it's likely that the cable is not the problem. My vote is for clocking. I've had non-Asterisk T1/E1 circuits act fine for long periods and then flake out. Correcting the clocking relationship fixed the flakyness. One way to investigate clocking is to use the pri commands to watch the D channel activity. If you see lots of Q.921 messages (SABME, RR, etc.) then it might be the result of fouled up messaging from bad CRC and other broken protocol events stemming from bad clocking. I've collected some articles and discussions on T1/E1 clocking here: http://www.voip-info.org/tiki-index.php?page=Asterisk+PRI -- Mike ___ 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] Best platform
On Sat, Jun 11, 2005 at 02:03:59PM +0200, Michiel van Baak wrote: On 13:22, Sat 11 Jun 05, Serge Schumacher wrote: What platform should you suggest to use asterisk ? I love the way the Debian updates work. Me too, but has the installation improved with the latest Sarge release? The announcement claims there are improvements. Debian has been extremely slow to improve it installer. I used CentOS 3.4 on two recent Asterisk installs with no problems. -- Mike ___ 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] Best platform
On Sat, Jun 11, 2005 at 02:30:31PM +0200, Michiel van Baak wrote: I will have a look at it later this week since my workstation is now replaced by a laptop so I have some testing hardware :) You're running from an upgraded Slink? That's the beauty of Debian. You may need to use Sid if you are using a laptop (I've got a well-maintained cheat sheet I'll share if you want it.) I'd try Sarge first. I run Sid on my notebook. I used CentOS 3.4 on two recent Asterisk installs with no problems. Isn't CentOS the free alternative for RHEL ? Yep. I never liked the filesystem layout RH used. I don't care about stuff like that - which probably indicates a lack of sophistication :) But if it works for you, use it :) That's the beauty of freedom :) Yep. I built two Asterisk boxes recently. I started with Debian and got the first working. The second install on an identical machine ran into problems. I probably didn't execute a step properly. I got tired of all that horsing around and decided that it was more important to have an Asterisk box running quickly than to have a well-maintained Linux box. I loaded CentOS 3.4 on both boxes and it just worked. I haven't learned the yum maintenance tool yet. I've been told that apt can be made to work with CentOS. CentOS 3.4 has a older version of Flex. The Asterisk compile suggests that you upgrade to a newer version. I went to the sourceforge Flex site and downloaded the most recent bz2 and followed the instructions for conf/make/install. Asterisk was very happy after that. Several days after building the two Asterisk boxes, Debian releases the looongg awaited 3.1 Sarge. I would have tried it over CentOS if it were available when I needed it. I'm going to give Debian Sarge a try in the near future. If they have a reasonably modern installer then I'll jump back into the Debian camp for my Asterisk work. -- Mike ___ 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] Best platform
On Sat, Jun 11, 2005 at 10:58:13AM -0400, Andrew Kohlsmith wrote: On Saturday 11 June 2005 10:36, Mike M wrote: You're running from an upgraded Slink? That's the beauty of Debian. What distro *doesn't* let you do this? I've been doing it this way with Slackware since the 3.x versions for chrissakes. I think the RH distros are not too good with upgrading. You don't hear many testimonials about running RHEL after having started from RH 5.0. Upgrading from the internet for free is not a good idea if you want to sell a new box 'o software every year to same person. With Debian, upgrading is nearly idiot-proof. I don't know about Slack. I've never used it. -- Mike ___ 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 and SS7
On Thu, Jun 09, 2005 at 12:28:15PM -0400, VOIP Consultant wrote: I have the exact same problem.It would ideal if we could set an astersik box with 2 E1 ports to do an IP-to-SS7 conversion. Anyone has done this before? See http://www.ss7box.com/asterisk.html C. Savinovich At 11:08 AM 6/9/2005, you wrote: The telephone company in Honduras say they will only supply an E1 circuit with SS7 signaling. Has anyone else run into this? Can anyone recommend a work-around for this problem? Thanks ___ 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] E1 and SS7
On Thu, Jun 09, 2005 at 06:11:06PM -0600, Michael Welter wrote: VOIP Consultant wrote: I have the exact same problem.It would ideal if we could set an astersik box with 2 E1 ports to do an IP-to-SS7 conversion. Anyone has done this before? I'm looking a signaling gateways--does anyone have any words of wisdom? http://www.sigtran.org What's your - budget - application - connection count - traffic volume - growth plan - protocol on the IP side - link type: A or F This can go deep and wide and way OT. I'll dispense what I know off-list to anyone interested. -- Mike ___ 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] English vs American voice files
On Wed, Jun 08, 2005 at 02:15:06PM +0100, Paul Redstone wrote: Hi In the end we found it easy to record our own using this section in extensions.conf. This also meant that we could add our own company specific ones in the same voice (not shown here). Basically you get someone to dial the 8NNN1 to record or 8NNN2 to playback. The prompts are shown below and we just printed out this text. It was our intention to use festival to read these, but this was easier. The text has been amended to reflect the UK (e.g. Hash instead of pound). Many sites may not need all of them and if you omit them the US voice will play instead. This looks interesting! That's a lot of work you're sharing. Thanks. I think the same voice is good idea and having a parochial voice is important as well. Has any enterprising person thought of a download service for this stuff? exten = _8015X,1,Macro(record-message,gb/tt-allbusy, All representatives of the household are currently assisting other telemarketers. Please hold and your call will be answered in the order it was received. ) That's funny. The National Do Not Call list in the US seems to be working so there's not as much need for it. Politicians wisely insulated themselves from the effects of the list. During election campaigns, this might be useful: exten = _8015X,1,Macro(record-message,gb/tt-allbusy, All representatives of the household are currently assisting other politicians. Please hold and your call will be answered in the order of what you are willing to do in order to get my vote. ) -- Mike ___ 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] bison/flex version warning
Hi, (I seem to be having some trouble getting messages to post on the list so I may be duplicating an earlier post. Apologies if this is the case.) I am compiling CVS tip Asterisk on a fresh CentOS 3.4 install. I got this warning: make ast_expr.a make[1]: Entering directory `/usr/src/asterisk' bison -v -d --name-prefix=ast_yy ast_expr.y -o ast_expr.c = NOTE: Using older version of expression parser. To use the newer version, NOTE: upgrade to flex 2.5.31 or higher, which can be found at NOTE: http://sourceforge.net/project/showfiles.php?group_id=72099 = My bison is this: bison (GNU Bison) 1.875c (Which appears to be ancient because the s/f link above shows the 2.5.31 flex archive date as 2003-03-31). What's the risk of ignoring this warning? Rhetorical question: why is CentOS using a moldy old flex/bison package? Thanks, -- Mike ___ 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] bison/flex version warning
On Thu, Jun 02, 2005 at 01:16:10PM -0500, Andrew Latham wrote: upgrade just ./conifure and make it... Centos version is older... Try upgrading. I followed the instructions in warning and in the package README and the warning went away on a subsequent make clean; make install make.out 21. Still using the same bison: [EMAIL PROTECTED] asterisk]# bison --version bison (GNU Bison) 1.875c It's the flex update that did it: [EMAIL PROTECTED] asterisk]# flex --version flex 2.5.31 I misinterpreted where the problem was. I thought it was in bison and instead it was in flex. This is way more direct involvement with lex/yacc than I've ever had before, so I pass that off as my excuse. I guess I could have just read the warning message that explicitly says that the expression parser is flex. Here's what the good output looks like. I like to put this stuff in to make for better googles. make ast_expr.a make[1]: Entering directory `/usr/src/asterisk' bison -v -d --name-prefix=ast_yy ast_expr2.y -o ast_expr2.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-HEAD-06/02/05-20:25:56\ -DASTERISK_VERSION_NUM=99 -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fomit-frame-pointer-c -o ast_expr2.o ast_expr2.c ast_expr2.c: In function `ast_yyparse': ast_expr2.c:1124: warning: implicit declaration of function `ast_yylex' flex ast_expr2.fl gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-HEAD-06/02/05-20:25:56\ -DASTERISK_VERSION_NUM=99 -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -fomit-frame-pointer-c -o ast_expr2f.o ast_expr2f.c ast_expr2f.c:1772: warning: no previous prototype for `ast_yyget_column' ast_expr2f.c:1848: warning: no previous prototype for `ast_yyset_column' ar r ast_expr.a ast_expr2.o ast_expr2f.o ranlib ast_expr.a rm ast_expr2.c Thanks, -- Mike On 6/2/05, Mike M [EMAIL PROTECTED] wrote: Hi, (I seem to be having some trouble getting messages to post on the list so I may be duplicating an earlier post. Apologies if this is the case.) I am compiling CVS tip Asterisk on a fresh CentOS 3.4 install. I got this warning: make ast_expr.a make[1]: Entering directory `/usr/src/asterisk' bison -v -d --name-prefix=ast_yy ast_expr.y -o ast_expr.c = NOTE: Using older version of expression parser. To use the newer version, NOTE: upgrade to flex 2.5.31 or higher, which can be found at NOTE: http://sourceforge.net/project/showfiles.php?group_id=72099 = My bison is this: bison (GNU Bison) 1.875c (Which appears to be ancient because the s/f link above shows the 2.5.31 flex archive date as 2003-03-31). What's the risk of ignoring this warning? Rhetorical question: why is CentOS using a moldy old flex/bison package? Thanks, -- Mike ___ 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 setup
On Thu, May 26, 2005 at 01:30:03AM -0500, Jay Milk wrote: http://www.digium.com/downloads/configuring_zaptel.pdf Nothing new here. This is more or less a verbatim copy of what's in /etc/zaptel.conf which is (2) below. http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf http://www.voip-info.org/wiki-Asterisk+config+zapata.conf Nothing new here either. The PRI information in these two links is covered in the (1) below. The example is exactly the same. Google is your friend. You might think that asterisk+pri or asterisk+pri+configuration or asterisk+pri.configuration would turn up more than emails from 2002. I even found the email where libpri was released under GPL: http://www.marko.net/asterisk/archives/0105/0022.html What I haven't found is a guide that gives the following steps: -- 1) compile/install libpri 2) compile/install asterisk 3) here's how to tell if your asterisk load is PRI ready 4) gather this information from your service provider 5) configure the collected information like this (OK, this does exist if you know what to google for) 6) here's how to test and troubleshoot your PRI connection If such a document doesn't exist then I'll start one because I'm getting ready to go through the process. If it does exist, then I'd like to follow it. -- Mike -Original Message- From: Mike M [mailto:[EMAIL PROTECTED] Sent: Wednesday, May 25, 2005 9:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] PRI setup On Wed, May 25, 2005 at 09:41:13PM -0400, Mike M wrote: I can't find documentation for setting up PRI. A push in the right direction would be most appreciated. Sorry for replying to my own post. I found: 1) http://www.digium.com/downloads/hw_article 2) the comments in /etc/zaptel.conf 3) a good discussion on timing choice: http://lists.digium.com/pipermail/asterisk- users/2005-May/107292.html Is there more? Thanks again, -- Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-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] PRI setup
On Thu, May 26, 2005 at 10:11:53AM -0500, Jay Milk wrote: You don't know how to compile asterisk? So why'd you ask about PRI? If you're starting from scratch, google for Getting Started with Asterisk and click on the first link. I've compiled asterisk from CVS head on a Debian Sid system. It works fine with my FXS and FXO interfaces for calls in both directions. I'm ready to move on to the PRI interfaces and found the documentation trail harder to follow. I outlined a document that I'd like to see about PRI installation, configuration, and test. The first steps are kind of obvious to the experienced, nevertheless, they shouldn't be omitted. If I was writing this document, I would create links to other documents that cover these steps. This document: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html/x460.html has you building zaptel and asterisk but not libpri. In an Asterisk/PRI HOWTO it would be important to mention building and installing libpri in the correct order. It would also be good to describe how to verify that it was done correctly so one doesn't waste time with an improperly configured system. Thanks for your help so far. -- Mike ___ 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] PRI setup
I can't find documentation for setting up PRI. A push in the right direction would be most appreciated. Thanks -- Mike ___ 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 setup
On Wed, May 25, 2005 at 09:41:13PM -0400, Mike M wrote: I can't find documentation for setting up PRI. A push in the right direction would be most appreciated. Sorry for replying to my own post. I found: 1) http://www.digium.com/downloads/hw_article 2) the comments in /etc/zaptel.conf 3) a good discussion on timing choice: http://lists.digium.com/pipermail/asterisk-users/2005-May/107292.html Is there more? Thanks again, -- Mike ___ 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] 128 kbs satelite link
Hi, I have not incoming phone number to test, but I think I can call you. If I have termination to your country I'll call you (please give me your stationary phone, not mobile). -- Best regards, Mikemailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How much to use Dialogic?
On Monday 09 June 2003 02:47, Matthew John Darnell wrote: BTW, I think it has been covered here before that the D41 is a half duplex card and wouldn't be good for conferencing. -- Steven Critchfield [EMAIL PROTECTED] Does anyone have an application that will parse the archives so you can search them? I was going to search the archives but it is too tedious to go month by month. http://www.asteriskpbx.org/index.php?menu=support At first I was searching archives manually until I found the Google search tool on the support page. I was using raw Google and was getting too much chaff. Enjoy. -- Mike M. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dinosaur *
Hello, With some trepidation I've come to inquire about platform requirements for * after having spent a couple of hours searching and browsing the archives and skimming the Handbook (very nice). I've found recommendations for 800-1000 Mhz and 128-256 MB RAM machines. My curiosity is not about what machine I need to start using * to support live comm ops. Rather, I want to know if a couple of dinosaurs that I have in the corner can be used to do some learning. I have a Pentium 133Mhz 16MB RAM box with PCI bus, and a Pentium 75Mhz 32MB RAM box with PCI bus. Will these boxes work for back-to-back experimentation with PRI in this configuration?: POTS phone-[ * ]---PRI---[ * ]---POTS phone I assume that I'll need to get a couple line cards and a couple T1 cards. Thx, -- Mike M. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dinosaur *
On Monday 02 June 2003 18:47, Scott Lambert wrote: On Mon, Jun 02, 2003 at 04:46:51PM -0400, Mike M wrote: Hello, With some trepidation I've come to inquire about platform requirements for * after having spent a couple of hours searching and browsing the archives and skimming the Handbook (very nice). I've found recommendations for 800-1000 Mhz and 128-256 MB RAM machines. My curiosity is not about what machine I need to start using * to support live comm ops. Rather, I want to know if a couple of dinosaurs that I have in the corner can be used to do some learning. I have a Pentium 133Mhz 16MB RAM box with PCI bus, and a Pentium 75Mhz 32MB RAM box with PCI bus. Will these boxes work for back-to-back experimentation with PRI in this configuration?: It seems to me that the Digium cards are DSPless. That means the host CPU has to act as the DSP for all these lines. I'm thinking a low end Pentium *might* be able to handle one FXS and one FXO port simultaneously. Hmm. Maybe the dinosaur would make a nice answering machine. I wouldn't try running a PRI card in one. Without compression you might be able to IAX the traffic onto a LAN connected to a WAN of some type. My 500Mhz celeron doesn't like to compile and take calls at the same time. I've got a 266 Celeron in a laptop running W-me. Not much of powerhouse, that one. It was better when running Linux. I like the AMD price-performance points personally. DSPless hardware is cheap but you have to have a serious host CPU to make up for it. OK. That makes me recall some things I read in the archives about DSPs on boards vs not on boards. POTS phone-[ * ]---PRI---[ * ]---POTS phone I assume that I'll need to get a couple line cards and a couple T1 cards. a couple of FXS cards and a couple of routers to handle a data T1/PRI. line side? generates ringing? http://www.marko.net/asterisk/archives/0001/0098.html Instead of routers I was thinking of using a twist cable with RJ45 on each end: 1 RT - TT 4 2 RR - TR 5 4 TT - RT 1 5 TR - RR 2 The PRI link is only one meter in a captive environment. Based on everything written so far, I think I'll build a couple of low-end machines sooner rather than later. -- Mike M. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Who would use Asterisk SS7?
On Thursday 29 May 2003 09:38, Michael Bielicki wrote: We would be a hour 0 user. And probably would also be abel to get some partners to test SS7 interconnect with since it would rid us of a hell of problems :) :-) I've been following the 2-4 port T1 cards thread closely because that's the kind of application that could benefit from having SS7-IMT. -- Mike M. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
