Re: [Asterisk-Users] ACD calls to busy agents
Julian Lyndon-Smith wrote: Tom Rymes wrote: That's a good idea, but it does not help when the agent receives a call from the queue. If an agent has call-waiting enabled (at least on our 7940 Ciscos...) the queue will send another incoming call while the agent is still on the phone withthe last call sent to them from the queue. Is that not the case? Have I misconfigured something? The Queue should not be sending a call to an agent that is marked as paused, that is what the pause was desigined for. Are you using more than 1 queue with the same agent ? When accepting a call from the queue, what mechanism is there to pause the queue member? Yes, it's possible to pause the agent when she places an outbound call or when recieving a direct-dialed or extention-dialed call, but how do you pause the agent when she accepts a call from the queue? To the OP: We too use Cisco 7940s for our office, and what I ended up doing, was turning off call waiting completely, then using the first line appearance for the user's actual extension, and the second line appearance for the call queue. It's just as annoying as call waiting without getting slammed by queue calls. -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 ___ --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
Nice smartass remark... of course anyone can register a domain name. Is forking asterisk legal? Of course it is! Asterisk is under the GPL, which means that anyone can fork it at any time for any reason. Look at this in a positive light... many open source projects have forked, and the branches almost always end up feeding on one another. Look at the competition between various linux distributions. Look at the competition and colaboration between the various *BSD communities. They all give and take from one another, creating a better /family/ of products. Oh, and the idea that these guys are out to get the same benifits that Digium enjoys is insane. I'd imagine that while Digium may make some money from selling alternate licenses, they make most their money from hardware sales and support. IMO, there's absolutely nothing wrong with a fork. In fact, were I someone with some seroius coding skills and/or the resources to make it happen, I'd have forked the damned thing 2 years ago, and likely would have been able to migrate it over to a true OSS license (BSD) by now. I know that the idea of forking asterisk has been tossed around by a LOT of people for a long time now, I'm glad it finally happened. -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 Kevin Walsh wrote: Brian C. Fertig [EMAIL PROTECTED] wrote: Further info. The domain is registered to Marc Olivier Chouinard. He has posted in the dev list. Can they do this? Is this legal? Yes - anyone can register a domain name. ___ --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
Jean-Michel Hiver wrote: IMO, there's absolutely nothing wrong with a fork. In fact, were I someone with some seroius coding skills and/or the resources to make it happen, I'd have forked the damned thing 2 years ago, and likely would have been able to migrate it over to a true OSS license (BSD) by now. Tss, tss. You can't change the GPL license to anything that is 'stricter' or 'freer'. Cheers, Jean-Michel. Licence changes can be made... look at Cistron Radius. They started with Livingston's code, which was under the BSD license. Once their code had been completely rewritten, they did an audit and found that they were no longer using the original code base and made the decision to move to the GPL. Why they wanted to move to a more restrictive license is beyond me (and this thread), but they did it. -- Troy Settle Pulaski Networks 866.477.5638 http://www.psknet.com ___ --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] MAX PRI for single server (was: Not enoughlinesavailable for Asterisk implemetation)
I would be most interested in seeing some TNT/APX configurations and corrosponding SIP configurations for Asterisk. Right now, I'm using call routes and switching off a T1/PRI to my asterisk box, and would love to change that to pure SIP if possible. The only caveat is that my TNT boxes are primarily used for dialup traffic. Also, on the TNT, I see calling name information coming in from the PSTN (Lucent 5E), but the TNT will not pass it through the PRI to my * box. Am I understanding correctly that calling name information also does not work with SIP? Thanks, -- Troy Settle Pulaski Networks 866.477.5638 http://www.psknet.com Damon Estep wrote: If you are looking for real high density VOIP termination I would look at something like a Lucent APX 8000, configure correctly it can pass 2500+ g.729 calls to the PSTN course we paid lots of $ for ours. Chris Chris, My experience has been that the APX and TNT products require a single SIP proxy, how are you load balancing 2500 calls? If all of the traffic is outbound it is fine, but what about origination? Are you using something other than asterisk as a SIP proxy? On a smaller scale the TNT is a good bet since the number of calls it will do (672 with t3) is closer to what an asterisk box can do without trans-coding. You can connect 1 partially populated TNT to one * box and not need another sip proxy, you can also have a failover sip proxy configured but not active unless the primary fails to respond. Both the TNT and APX have issues with calling name delivery over PRI when connected to a Lucent 5ESS configured to do end office LIDB dips, so calling party name on inbound calls can be a bear, look to connect to a Nortel DMS if you have the option -- go figure the LUCENT media gateways work better with Nortel class 5's than then they do with lucent class 5's. Have you learned something I have not about how to get all of the calls a TNT/APX can handle terminated on the SIP side without still having a single point of failure in the SIP proxy? ___ --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 ___ --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: Top posting
This is top posting, but you don't need to worry about it yet. First, you need to learn how to post in plain text and how to properly quote a message. Sorry to continue in HTML format, but switching to plain text really screwed this one up. BTW, as for the top-vs-bottom argument, I have friends who are tops, and friends who are bottoms. Every one of them seem to get extreme satisfaction from their relationships with the other. -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kavit MunshiSent: Tuesday, November 16, 2004 8:46 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Re: Top posting Jon Radon wrote: Worst thread ever. On Tue, 16 Nov 2004 19:02:07 -0600, Michael Greb [EMAIL PROTECTED] wrote: On Tue, 16 Nov 2004 09:53:52 -0600, Jay Milk [EMAIL PROTECTED] wrote: So, that's how my tax dollars are spent? Outrageous, and certainly news-worthy. Good luck fighting off CNN and the like when this leaks out. Not at all, this is one of my favorite policies that has come from the performance improvement department. Yes that is right, it is official policy at my location to not deal with people who top-post. PI decided that with people moved around between positions it is always best for bottom-posting just as if on a mailing list even in two party communications as, if another person comes into the discussion, it is much quicker, and thus cheaper, to have a properly formatted communication to come up to speed. This is the same as the policy that businesses that send ill-formatted bussiness letters will not receive addition business when there is another suplier capable of delivering the product/service. Top-posting is even grounds for being written up if you later need to forward a copy of a message on to another department or person. Michael Greb ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users what is top posting anyway? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Multi companies
Sure, you can assign different contexts to different zap channels, but how does this help? Normally, the telco will send each call on the first available channel in a given trunk group, sometimes, they will come in on random channels. When a call rings in on Zap/1-1, the only way to know what to do with it, is by the DNIS information. exten = 2200,1,NoOp,Company A - main line exten = 2201,1,NoOp,Company A - Fax exten = 2211,1,noOp,Company A - CEO Direct Line exten = 3000,1,NoOp,Company B - main line exten = 3001,1,NoOp,Company B - Fax exten = 3022,1,NoOp,Company B - Sales exten = 3023,1,NoOp,Company B - Customer Service Any of these calls might come in on any of your lines, so how does setting a different context for different zap channels help? -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Shaw Sent: Friday, July 23, 2004 7:15 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Multi companies Can't you just assign a context to a Zap channel or Group of Zap channels in zapata.conf just like you do with SIP? (e.g. Context=company1) If so, you don't need any of that, just create separate IVR contexts for each company and assign those contexts to specific Zap channels or channel groups you want... I might be wrong but that would seem to be the logical way to do it... - Original Message - From: Joshua McClintock [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 23, 2004 3:50 PM Subject: Re: [Asterisk-Users] Multi companies You don't need the _ on the front of those extensions since those particular examples aren't patterns. My mistake. Check out www.voip-info.org for MANY good examples. On Fri, 2004-07-23 at 15:21, Joshua McClintock wrote: Depending on the context that your 'incoming' lines are on, you can do something like this: [incoming-lines] exten = _1235551212,Macro(autoatt-company1) exten = _1235551213,Macro(autoatt-company2) [macro-autoatt-company1] Do some junk, dial some peeps [macro-autoatt-company2] Do some junk, dial some peeps On Fri, 2004-07-23 at 14:57, Martin Keding wrote: I am fairly new to Asterisk and I want to do some testing with multi-companies on the same box. I have two inbound lines and I basically want one to trigger auto-att. for company 1, the other line to trigger auto-attend for company 2. Could somebody point me to a sample conf. or documentation. Thanks Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Call queues
Avizion, you're joking right? -= Info about application 'AddQueueMember' =- [Synopsis]: Dynamically adds queue members [Description]: AddQueueMember(queuename[|interface[|penalty]]): The AddQueueMember function does indeed allow you to set the penalty. Too bad penalties don't work though (or maybe they work too well?) SIP/100, penalty 1 SIP/200, penalty 2 Call comes in, SIP/100 picks up Call comes in, SIP/100 is busy, but SIP/200 NEVER rings... *sigh* -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of avizion Sent: Friday, July 23, 2004 5:54 AM To: [EMAIL PROTECTED]; Jeremy Kenney Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Call queues Quoting Jeremy Kenney [EMAIL PROTECTED]: Hello I am new to asterisk I want to setup the call queues where it will ring multiple devices at the same time and send the call to the first one that is picked up. There doesn't need to be an agent login for this I don't think I just want setup so no login is required. Please help There are several ways to accomplish this. Like the two others posts suggest - you can simply use the Dial() application directly. This will leave you with exactly the functionality you are asking for. What is does not give you is a real queue where members can join / part as they see fit (app. AddQueueMember / RemoveQueueMember). If you want to have your agents logged in from the start, you can simply define these in etc/queues.conf like SIP/phone1 or IAX2/phone1. The last option will even let you define a penalty (in etc/queues.conf). What this lacks is a persistant penalty. I've been using a little time investigating this - and I came to the conclusion that if I want persistant penalties for dynamically added members I would have to write my own wrapper in AGI. While I'm pretty much done with that part - it's not exactly a beautiful hack - but I might publish it if wanted. I will be posting on the asterisk-dev list soon - in order to get second oppinions on this implementation. Several things needs coverage - but all this in due time :) I hope you can use this - and feel free to ask into any of the above... Regards - avizion on irc.freenode.org #asterisk ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Echo on a PRI
On the subject of echo on a PRI, I too get this, but only when calling people in certain rate centers. Calls within my LATA (primarily VZ) are completely free of echo. Calls to a neighboring LATA (all Sprint) have echo on almost every rate center. I wish I knew more about this so I could rip Sprint a new one and tell them to fix their trunking, but... -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Tuesday, July 20, 2004 7:00 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Echo on a PRI On Mon, 2004-07-19 at 19:12, David Goldfein wrote: Hi, I recently set up the following in a production system (2.8 GHZ Xeon, 1 Gig Memory, Dell 2650). Telco - PRI - Asterisk - T1 - PBX I am getting an occasional noticeable echo on some of the phone lines (random inbound and outbound). Everyone I ask keeps telling me that I can't be having echo since I am on a PRI, which is a digital circuit. Ok, so I can't be having echo, but I am! Does anyone have any ideas of what might be causing the echo in this situation? Your PRI and the T1 itself cannot introduce echo on their own. What you may see though is that you are introducing a delay as you traverse the asterisk link. Asterisk will buffer 8 bits per channel from the PRI before it send it down the T1 line to the PBX. This is a new delay that is now added on to the latency your PBX introduces. A guess is that you also get the 2 machines fighting against each other on the echo. I doubt you can turn off echo cancel in the PBX so you should try turning it off in asterisk. It should help reduce some latency in asterisk and let the PBX handle the rest of the echo cancel on it's own. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Echo on a PRI
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Tuesday, July 20, 2004 9:14 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Echo on a PRI On Tuesday 20 July 2004 09:04, Troy Settle wrote: On the subject of echo on a PRI, I too get this, but only when calling people in certain rate centers. Calls within my LATA (primarily VZ) are completely free of echo. Calls to a neighboring LATA (all Sprint) have echo on almost every rate center. I wish I knew more about this so I could rip Sprint a new one and tell them to fix their trunking, but... Are you sure it's Sprint's fault? I mean perhaps calling within your own LATA has less delay than calling neighbour LATAs and, combined with the delay that the T100P/TE405P introduces, presents enough delay to perceive echo... Pretty sure. Severe echo problems are only apparent when calling destinations within certain rate centers in this particular Sprint LATA (956) from my LATA (244). What's weird, is that inbound calls /from/ these same rate centers seem to have much less echo problem. It's possible that there's a something wrong with the trunking between my telco (KMC Telecom), the tandem (Verizon), and my LD carrier (MCI), then going to the destination (Sprint). The reverse call path is Sprint = Sprint = KMC = me. Fortunately, most of our calls are inbound, so it's not a huge issue at this time. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Queue: strategies and penalties
All, For the last 10 months, I've been using strategy=ringall. This has worked fine and did what I wanted, but at this point, I'm needing to implement a 'penalty' or delay for some members of the call queue. 1: remote users(remote flunkies) 2: level-1 support (flunkies) 3: level-2 support (glorified flunkies) 4: level-3 support (super flunkies) When a call comes in, I want it to ring the first group for 30 seconds, and if there's no answer, ring groups 1-2 for 30 seconds. If no answer, ring groups 1-3 for 30 seconds, and if still no answer, ring all 4 groups until the call is answered. What do I need to do to get this behavior? If the answer involves $$, tell me about it, I'm not afraid to spend some cash to help streamline my business. Thanks, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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-Users List Etiquette
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Lee Sent: Tuesday, June 15, 2004 6:34 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk-Users List Etiquette Kevin Walsh wrote: Steven Critchfield [EMAIL PROTECTED] wrote: You forgot to add in how awful it is when people post using HTML and then override font sizes or assume blue is an appropriate font color for their message. While I know some people don't like it when I turn my attention to them, if it takes me even one more button press to be able to read your mail, it isn't likely to be interesting to me to even bother helping you with your problem. Since the majority of unix users understand how each of us tweak our environment to be the most productive for us, we don't like it when you take liberties with our settings. He also forgot to mention how awful it is when people lazily top-post instead of taking the time to format their followups correctly. This is especially true when trying to follow a thread found in the archives. I fully agree with your anti-HTML comments, by the way. I think you will find that about half the people out there disagree with this sentiment (a guess based on the number of top and bottom posters I have seen) so no matter how often you ask it is not likely to change things much. Top posting is what a lot of people are very comfortable with. It also has the advantage in lists that when you step through a thread the answer to the last item is ready for you to read. So If you bottom post you make life harder for the thread reader but if you top post you make life harder for those that get a long mail out of the archives.Who should we favor? Don't ask why I am bottom posting, I have no good reason, it just so happens that I am. I don't like HTML either but a lot of people don't know they can switch it off or that it even exists (its a word processor isn't it?). Getting offended by these personal preferences just leads to that etiquette problem, the god ol flame war. Or at least heated debate that will never be won with so many advocates for each side, that the lists become quite full of top/bottom html/text arguments. Please don't bring these subjects into things it just makes people with other views upset. I'm quite content to post at the top, bottom, or inline. It really just depends on the nature of the message I'm replying to, the subject, context, and format of earlier messages in the thread. However, my preference is for top posting. The reason, is that in order to read my message here, you had to scroll through ~70 lines of previous discussion. Stuff that you've /already/ read since you've been following this thread. Oh! Wait, you found this in an archive, so you /want/ to have the thread fully quoted so you don't have to go hunting down the references. Good, that's why I didn't trim this post. Oh, wait, the guys that are following this thread as it's being discussed would prefer that I trim out the stuff up there, in which case, I would be neither top posting, nor bottom posting. This message would be a post unto itself that wouldn't have any quoted material at all. Afterall, you've already read the referenced material. So, the bottom line is that top-posters are lazy? I say yes, we are. We don't want to have to scroll through pages of quoted material just to get to the new stuff. I say that the bottom posters are lazy. They want a bottom post so that they enter into a thread 12 messages later, and not have to read the thread 'backwards.' Read your mail to begin with, and you wouldn't have this problem, and you would actually start to appreciate the top posters, because they're making it so you don't have to scroll through ~70 lines of quoted material to get to the new stuff. -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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-Users List Etiquette
A: Because we read the question in the previous message. Q: Why should I post my reply above the quoted text? -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hermann Wecke Sent: Wednesday, June 16, 2004 2:39 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk-Users List Etiquette On Wed, 16 Jun 2004, Nicholas Bachmann wrote: You might try reading http://www.caliburn.nl/topposting.html -- it explains why people don't like top posting. Or read this quote: A: Because we read from top to bottom, left to right. Q: Why should i start my reply below the quoted text? - -- http://www.i-hate-computers.demon.co.uk/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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-Users List Etiquette
-Original Message- From: Gonzalo Servat Sent: Thursday, June 17, 2004 9:34 AM Sorry to butt into this thread, but I think this is where you went wrong. There was absolutely no need to quote 70+ lines of text to say what you had to say. You're supposed to quote the relevant bits (as I did with this email), not the entire thread. It's an open mailing list, you're not butting in at all. I agree with you completely, however, there is this great tool called 'exageration' that is sometimes used to make a point when a real-world example would be too small to be perceived as signifigant. For those nay-sayers, please look at my post carefully. I bottom posted, keeping the existing style, and while I left the quoted material untrimmed, I also mentioned the other extreme, which is to completely exclude any quoted material at all. The bottom line of this issue is that everyone has their preferences, and no amount of crying and whining will cause the other side to comply with your wishes. There are valid reasons for both posting styles, live with it. Those who continue to whine and cry about top posting need to be larted with a vengence. It's like the last cry of those who lost the vi-vs-emacs debate. Just because you prefer one over the other doesn't make everyone else 'wrong.' IMO, the top-vs-bottom topic really needs to be classified right along side with the RH-vs-Debian, red-vs-blue, unix-vs-windows, ford-vs-chevy, linux-vs-bsd, and other similar cases of personal preferences. The is no winner, there never will be a winner. BTW, for those of you who are curious, I too dispise HTML formatted email in a mailing list environment. I also dislike those who flagrantly disregard existing styles within a thread (but, it's ok if different threads have different styles). I also have very low regard for those among us who would hijack a thread. I don't use a threaded mail reader myself (sucks to be me), but when browsing archives by thread, it's really annoying to find questions about personal lubricant in the middle of a heated debate about top-vs-bottom. Of course, sometimes a thread will mutate naturally, at which point, it may be appropriate to change the subject (which I'm not going to do, since I'm too damned lazy. Oh, for those curious, my single, biggest beef with mailing lists, is the inclusion of a list tag in the Subject: line. I know it's Asterisk-Users, because it says so in the To: line. It also says so in the List-ID: and Sender: lines. -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] X100P to hardware PBX
Can you plug a regular telephone into the same port on your 'hardware' pbx
and use it?
--
Troy Settle
Pulaski Networks
http://www.psknet.com
866.477.5638
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Boot
Sent: Wednesday, June 02, 2004 8:24 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] X100P to hardware PBX
I have asterisk successfully dialing out using a X100P over a normal
analogue PSTN line. But when I try to dial out over an
analogue line that
goes via a hardware PBX the call asterisk does not dial. Is there a
configuration change I should make ? I am thinking of
something like not
wating for dial tone.
From my extensions.conf
[outgoing]
exten = _X.,1,Dial,Zap/1/${EXTEN}
---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.693 / Virus Database: 454 - Release Date: 5/31/2004
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RE: [Asterisk-Users] Hyperthreading?
I'm running asterisk on a 2.8Ghz w/HT and 2.4.25 kernel. I wasn't aware that I needed to disable HT, but all seems to be running ok for now. The 2.4.x kernel seems to be completely ignorant of hyper threading, which IMO, is quite frustrating. HTT has been around for years now, and 2.4 kernels still can't use it. I've been trying, unsucessfully, to get a 2.6 kernel built and running, but it doesn't like my ethernet card. The eepro100 and e100 drivers both (at separate times) load fine, detect the nic, but the nic can't function. FWIW, the hardware reported in dmesg/lspci is 82562EZ. It works fine with linux 2.4, and FreeBSD 4.8 and 5.x. -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bond Sent: Tuesday, June 01, 2004 4:35 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Hyperthreading? Are they any issues still with hyperthreading processors, I've read and been told by a few people to make sure its disabled in bios if I want to use * on a hyperthreading machine. Kind Regards, Chris Bond ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Hyperthreading?
# uname -a Linux roanoke-voip01 2.4.25-gentoo-r2 #6 SMP Mon May 31 07:08:41 EDT 2004 i686 Intel(R) Pentium(R) 4 CPU 2.80GHz GenuineIntel GNU/Linux # cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 2 model name : Intel(R) Pentium(R) 4 CPU 2.80GHz stepping: 9 cpu MHz : 2793.042 cache size : 512 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe cid bogomips: 5570.56 # dmesg | grep -i cpu Initializing CPU#0 CPU: Trace cache: 12K uops, L1 D cache: 8K CPU: L2 cache: 512K CPU: Physical Processor ID: 0 Intel machine check reporting enabled on CPU#0. CPU: After generic, caps: bfebfbff CPU: Common caps: bfebfbff CPU: Trace cache: 12K uops, L1 D cache: 8K CPU: L2 cache: 512K CPU: Physical Processor ID: 0 Intel machine check reporting enabled on CPU#0. CPU: After generic, caps: bfebfbff CPU: Common caps: bfebfbff CPU0: Intel(R) Pentium(R) 4 CPU 2.80GHz stepping 09 per-CPU timeslice cutoff: 1462.56 usecs. enabled ExtINT on CPU#0 WARNING: No sibling found for CPU 0. . CPU clock speed is 2793.1604 MHz. cpu: 0, clocks: 1995112, slice: 997556 CPU0T0:1995104,T1:997536,D:12,S:997556,C:1995112 -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Tuesday, June 01, 2004 9:16 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Hyperthreading? I'm running asterisk on a 2.8Ghz w/HT and 2.4.25 kernel. I wasn't aware that I needed to disable HT, but all seems to be running ok for now. The 2.4.x kernel seems to be completely ignorant of hyper threading, which IMO, is quite frustrating. HTT has been around for years now, and 2.4 kernels still can't use it. They can't? HT is detected in /proc/cpuinfo (flags) and I see two processors with 2.4.25 SMP kernels... What exactly isn't it using? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Updated Zaptel this morning and *BOOM* *CRASH*
First time around, I just unloaded/reloaded the modules. The box locked up tight. On reboot, I get this: general protection fault: CPU:0 EIP:0010:[c01defb3]Not tainted EFLAGS: 00010097 eax: f61d4260 ebx: f61d4260 ecx: edx: f61d425f esi: f61d4264 edi: f61d4260 ebp: f4de7f14 esp: f4de7ef4 ds: 0018 es: 0018 ss: 0018 Process sh (pid: 387, stackpage=f4de7000) Stack: 0297 0001 0001 0086 0001 f61d411c 0202 f61d4008 0004 f897ebaf 0001 0001 f897eccf f61d411c 0004 0008 f61d4008 f6551680 f61d4008 f61d4000 Call Trace:[f897ebaf] [f897eccf] [f89c7852] [c01d0349] [c01d0568] [c01d2fa8] Code: 8b 01 85 45 f0 75 1c 8b 02 89 d3 89 c2 0f 18 00 39 f3 75 e9 0Kernel panic: Aiee, killing interrupt handler! BTW, this is kernel 2.4.25-gentoo-r2 -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Updated Zaptel this morning and *BOOM* *CRASH*
-Original Message- From: Troy Settle Sent: Monday, May 31, 2004 6:49 AM First time around, I just unloaded/reloaded the modules. The box locked up tight. On reboot, I get this: general protection fault: CPU:0 EIP:0010:[c01defb3]Not tainted EFLAGS: 00010097 eax: f61d4260 ebx: f61d4260 ecx: edx: f61d425f esi: f61d4264 edi: f61d4260 ebp: f4de7f14 esp: f4de7ef4 ds: 0018 es: 0018 ss: 0018 Process sh (pid: 387, stackpage=f4de7000) Stack: 0297 0001 0001 0086 0001 f61d411c 0202 f61d4008 0004 f897ebaf 0001 0001 f897eccf f61d411c 0004 0008 f61d4008 f6551680 f61d4008 f61d4000 Call Trace:[f897ebaf] [f897eccf] [f89c7852] [c01d0349] [c01d0568] [c01d2fa8] Code: 8b 01 85 45 f0 75 1c 8b 02 89 d3 89 c2 0f 18 00 39 f3 75 e9 0Kernel panic: Aiee, killing interrupt handler! BTW, this is kernel 2.4.25-gentoo-r2 Oops... Missed one step. It's not locking up until a few moments (less than a second?) after running ztcfg. -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] RxFAX generates no tiff file
Running spandsp 0.0.1k, tiff 3.5.7. I put some audio log files in the same directory. Thanks, -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Saturday, May 22, 2004 10:27 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RxFAX generates no tiff file Hi Troy, People had a lot of problems like this with earlier versions of spandsp. However, the latest version is pretty solid, and people are using it in high volume production applications. If you are getting these bad results with the latest version I would be interested to see the audio log file, so I can investigate the reason. Regards, Steve Troy Settle wrote: Dunno about not being able to generate a tiff, I got rxfax to do that, but they're badly malformed. http://roanoke-voip01.psknet.com/fax/ -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] app_queue and app_groupcount
I just disable call waiting on all my sip phones and on all zap interfaces. No problem. -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julien Levi Sent: Saturday, May 22, 2004 6:43 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] app_queue and app_groupcount The new app_groupcount looks great for most applications but it a is a step back for call queueing... since app_queue calls physical interfaces and not extensions, app_groupcont can't be used to limit the calls passed to a dynamically added agent. I presently use the broken sip incominglimit feature (even though it's less than ideal as it also limits outgoing calls preventing consultative transfer using sip refer commands) I could start to use the agents app with agentcallbacklogin to (almost) emulate the current behaviour and use app_groupcount - I can automate the login using agentcallback login, but not the logoff, it prompts for an extension to forward to requireing # to pressed to log off - is there any way round this? I'd prefer to keep the simplicity of simply dialing one number to log on in or out of the queue from any phone, without having to define agentids, passwords, etc which we don't need. I hope incominglimit and outgoing limit aren't going to be removed entirely... -- Julien ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] RxFAX generates no tiff file
Dunno about not being able to generate a tiff, I got rxfax to do that, but they're badly malformed. http://roanoke-voip01.psknet.com/fax/ -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike Heininger Sent: Saturday, May 22, 2004 12:52 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] RxFAX generates no tiff file Hi, I am trying to receive a fax with the spandsp library. The sending fax says success but there is no tiff file generated. I use exten = 7000,1,rxfax(/tmp/testfax.tif) in my extensions.conf. The connection is via SIP/G.711 as I have read on the list that this can sometimes work (I know Fax over IP is troublesome without T.38). I think the transmission should not be the problem because of the success on the sending fax. This is the debug output. Am I missing something? TIA, Mike *CLI-- Executing RxFAX(SIP/uid-c5b6, /tmp/testfax.tif) in new stack Changed from phase 0 to 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up Slow carrier down T4 timeout in state 9 Changed from phase 3 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 T2 timeout Start receiving document Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Lucent Phones
-Original Message- From: Gregory Junker On Mon, 2004-04-12 at 11:28 -0400, Troy Settle wrote: At this point, I'm using straight Asterisk, with a a PSTN gateway at a data POP passing calls via IAX to my PBX here in the office. Who is the PSTN gateway provider? The only CLEC around here that is seriously considering any sort of VoIP commercial service is Time Warner Telecom (TWTC), our current telecom provider, and I have no details on what they are considering. If VoicePulse had a reason to offer PSTN local exchange service in this area I'd drop TWTC like a bad habit...I'd then settle for the Cincinnati Bell DSL or some other form of lower-cost business-class broadband for IP data access. KMC Telecom is my CLEC. I'm colocated with them at their central office. I have a DS3 for bringing PRI into my Lucent TNT. The TNT can function as a rudimentary switch and has the ability to generate T1/PRI that plug right into my * box. So, in essense, I'm my own PSTN gateway provider. FWIW, you should be able to completely eliminate the Connectreach and bring your T1 directly into *. You just need to find out what channels on the T1 are used for voice, and which are used for data. Using a T400 or TE405, you can cross connect the data channels out to another T1 to go into your router. TWTC has examined the T100P and informed me that it's impossible, since their IBL uses proprietary formatting and signalling. Also, I ought to be able to use the data channels directly, according to Digium, since my proposed Asterisk box is also our router. If they are lying to me (which I doubt...they have a vested interest in using a proprietary method), then as for finding out which channels are used for what is as simple as trial-and-error and a cell phone. ;) (and of course, the $400 or so to pick up a T100P to try it out...) I am guessing first four are voice and next 12 are data. I'm rather confused by this. It was my understanding that the connectreach was nothing more than a glorified channel bank with IP routing capabilities (I have 2 customers with 6 voice lines, and 384k of data that's handed off as ethernet by the connectreach. The voice lines come off a 50pin telco connector. If TWTC, like my CLEC, offers the connectreach at no additional cost, then I seriously doubt that they would lie to you. Returning the connectreach would save them some small amount of money at the end of the day. If their solution is propriatary, that's fine, but I don't see how/why they wouldn't be able to reprovision the T1 as a normal circuit. If you can get Digium to give you a 30 day refund window, then I'd say that it's well worth it to give this a try. -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Config docu for SIP-PSTN gw ?
Andreas, The documentation you seek is on the Asterisk website and the Tiki. If you jump on IRC, I'm sure there will be plenty of people around that can answer questions, or for a small fee, perform your initial configuration for you. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas Czerniak Sent: Sunday, April 11, 2004 8:45 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Config docu for SIP-PSTN gw ? Hi all ! Have anyone a resource / link for documentation to configure Asterisk to act as a SIP 2 PSTN gateway (ISDN PRI) ? Thx. Regads, Andreas. -- If you want to pray. Go to the sea. Andreas Czerniak [EMAIL PROTECTED] PGPkey http://pgp5.ai.mit.edu:11371/pks/lookup?op=getsearch=0xEDB224EC ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Lucent Phones
Greg, You're going through what I went through last year, and I feel your pain. I started by mixing * with Lucent, but that turned into a nightmare. It was a twisted and convoluted setup to transfer calls to more than a couple VOIP extensions, or for callers to dial anyone's extension directly. At one time, I think I had a call pass through * three times and the lucent twice. What a mess. At this point, I'm using straight Asterisk, with a a PSTN gateway at a data POP passing calls via IAX to my PBX here in the office. From there, I have a mix of SIP and POTS (cordless) extensions. FWIW, you should be able to completely eliminate the Connectreach and bring your T1 directly into *. You just need to find out what channels on the T1 are used for voice, and which are used for data. Using a T400 or TE405, you can cross connect the data channels out to another T1 to go into your router. I agree that it would be cool as hell to reverse engineer Lucent's phones. Having an 18D on everyone's desk would be the coolest damned thing ever. The problem, of course, is not only reversing the protocols, but also developing the hardware interface (a regular channel bank will not do the trick). -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Junker Sent: Wednesday, April 07, 2004 7:28 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Lucent Phones Right, I know that the voice part is POTS because I have a standard cordless phone plugged into our Partner system. Hmm, wouldn't ETR be covered under a patent and not a copyright? And has 17 years been up yet? And if someone is selling devices that convert to/from ETR, then the protocol spec is available in some form (even if it's some draconian Avaya licensing scheme). I agree that Avaya has a vested interest in keeping the spec out of the public eye (sell phone upgrades, sell Merlin adapter modules), but this technology is definitely getting long in the toothwhich doesn't mean that my users exactly want to give up the familiarity of the Partner phones just yet. ;) And since I already have Partner phones, and don't really care to spend $200-$300 a pop to replace them with Snom or Cisco phones (good as they may be)... My goal is to get rid of that box on my wall. I already got rid of one (Cisco 1720 that was our router, replaced by a Linux server/router), now I have two to go (Lucent ConnectReach for our Time Warner Telecom IBL, and the Partner ACS phone system). Hell, Lucent Technologies ought to pay me rent for the amount of space they occupy on my walls. [rant=on] It is completely obnoxious to me that I have to take an incoming channelized T1 and have it broken out into physical copper wire so that I can insert it into my Partner system for voice. If I had then to take that copper, spend beaucoup more bucks to be able to put it back INTO digital form so that it can work with an Asterisk PBX...that's borderline surreal to me. Everyone is so vested in making sure that none of their damned equipment interoperates with anyone else's (yet all the while paying serious lip service to the holy grail of standards) that I am to the point where DCMA be damned, if I can measure it I can figure it out. It pisses me off no end that TWTC can't simply send a normal T1 into my business (and therefore allow me to use a simple T100P), and I'll bet that when they start offering VoIP in this area (SW Ohio) it'll also involve some absurd piece of proprietary equipment further to clutter up my wall or rack. [rant=off] At any rate, yes, I could pick up a TDM400 and have Asterisk act like Partner ACS analog extensions, or pick up 3 X100's and use it directly for the incoming lines (and then deal with the user fallout regarding adaptation to X-Lite or something similar), but I just can't bring myself to do it, honestly. Ultimately, I want those boxes off my wall because technologically, they do not need to be there. Guess I'm stuck with finding 7960's on eBay as cheap as I can. *sigh* Anyone want an outmoded Partner ACS R1.0 analog phone system? ;) Greg On Wed, 2004-04-07 at 17:46 -0500, Steven Sokol wrote: On Wed, 2004-04-07 at 15:44 -0500, Eric Wieling wrote: Don't expect the fancy function buttons to work, however. That's specifically what I was asking about... Has anyone tried to decipher the ETR signaling protocol? Or is it such a closely guarded Lucent/Avaya secret as to make the formula for Coca-Cola look like an open-source recipe? ETR (Enhanced Tip/Ring) supposedly uses some variety of serial protocol over two lines to provide the screen functionality. The voice channel is still POTS. These phones are sold with the Partner system and can be added to the Magix systems
[Asterisk-Users] Cisco QoS Howto
Can anyone point me to some sample Cisco QoS configurations suitable for IAX2? I've looked through Cisco's site, and get overwhelmed with the level of documentation (too much of a good thing). My PSTN gateway and PBX (both *) are connected via 2xT1 (per-packet load balancing) between a Cisco 7206 and a 3640. When the total bandwidth pushes much past 50%, I start getting some crazy distrotion (jitter?), making it impossible for one or both parties to understand the other. TIA, -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] freeBSD zaptel driver
There have been several bites on the bounty, but nobody's hooked yet, so I don't think a duplication of effort is an issue at this point. I do wish that someone would get this done though, as I don't exactly get thrilled over maintaining linux boxes. *sigh* -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Arnold Sent: Monday, March 01, 2004 7:16 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] freeBSD zaptel driver On Sun, 29 Feb 2004, Michael Rowley wrote: Does anyone have any information on the zaptel driver under freeBSD? I know that there has been a 1200$ bounty posted, but wasn't sure if anyone with any talent has taken up the project. (I don't really have any talent... :| ) We have people looking into zaptel support under FreeBSD. But are there other people out there working with this issue? It time to speak up now so we don't duplicate the effort. And i don't really care who gets the bounty, as long as it gets done. /Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 sounds also now in CVS
Perhaps someone is writing, or has written, an AGI script to fetch current weather conditions and spit it out to callers? -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken Alker Sent: Tuesday, January 20, 2004 3:26 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New sounds also now in CVS I keep noticing the references to words related to weather in this thread and I am getting more and more curious; why the weather related words for a PBX? What other broad topics for words exist right now besides those that are PBX specific and weather-related? ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 sounds also now in CVS
How about a hashed directory structure? Something like this would be easily human and machine readable. This can also be an opportunity to lay the groundwork for internationalization. Numbers and digits would have their own directories, as would the demo phrases, agent and voicemail sounds. .../sounds/en/o/n/on.gsm .../sounds/en/days/1.gsm .../sounds/en/months/0.gsm .../sounds/en/numbers/h-19.gsm .../sounds/en/numbers/2.gsm .../sounds/en/t/h/thousand.gsm .../sounds/en/numbers/4.gsm .../sounds/en/a/t/at.gsm .../sounds/en/numbers/7.gsm .../sounds/en/numbers/40.gsm .../sounds/en/numbers/6.gsm .../sounds/en/letters/a.gsm .../sounds/en/letters/m.gsm .../sounds/en/t/r/troy.gsm .../sounds/en/w/r/wrote.gsm (sorry if I'm a little off from when I actually press the send button :D) -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mindworks Wireless Sent: Monday, January 19, 2004 1:53 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New sounds also now in CVS Is probably not the best way to handle it, but you could store all sounds in one directory and then create another directory that has subdirectories like weather. The items that are most frequently used would then be symlinked to the original sound directory. Just another way of organizing it. If someone wanted a sound to be included in the sub-directory, they could easily do a ln -s... Brent On Sun, 18 Jan 2004 [EMAIL PROTECTED] wrote: ... Quoting John Todd [EMAIL PROTECTED]: As to the specifications of directories for sounds in certain groups: yes, I think that is a good idea, but I am unsure how to implement it. Mark and I touched on that last night while adding the sounds to the CVS server, but I told him not to create a separate directory for weather terms because it would be difficult splitting the sounds into categories, which is perhaps only laziness on my part. I thought that in the future it would be difficult to determine what words should be put in their own directories versus what words should be moved to the main directory. As an example, if one were to do a network monitoring list of sounds (which, actually, I have had Allison already do now that I look at my archives) then a partial list of sounds would be up, host, down, dns and ping. The terms host and ping clearly should be in a directory with monitoring sounds. But... where would down go? It's generic enough that it should really go in the main directory. However, it's specifically a part of the monitoring sound set. So where does it go? I couldn't come up with an answer on this, so I just ignored the question for now. :-) Opinions welcome. It will probably be impossible to divide audio clips into different directories without duplication of clips or massive headaches determining direcories. My suggested method of handling this is to have all of the sounds in one directory and create multiple indexes. Each index would have listed all words/phrases for the topic. For example all weather terms would be placed in a weather index. Any phases needed for weather would be in here even if it appears in other indexes such as a time index or a monitoring index. The index would point to the actual audio clip of this common directory. The index for each topic could be a text file with a list of phrases with their corresponding file name. So there would be as many files (indexes) as catogories (ie Weather, monitoring, etc). When an audio clip was added it woud be added to one or more of these index files. We use a similar method on our intranet for indexing pdf files of USDS sheets. :| Upon further thought, perhaps an index could include another index. The index for numbers comes to mind since almost all of the others may include this one. Would we really want to repeat all of the numbers in an index? Anyhow, maybe I am just talking myself into a corner. Take the suggestion just as a point for further discussion. JT -- Don Pobanz - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo
RE: [Asterisk-Users] wav49 voicemail problem with Windows Media Player
I can't reproduce this either, but I do have the gsm codec installed (though WMP won't play a .gsm file). I play the wav49 files in Winamp with no issue. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Warwick Ward-Cox Sent: Thursday, January 15, 2004 10:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] wav49 voicemail problem with Windows Media Player I'm having the same problem. Warwick - Original Message - From: Jim Flagg [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, January 15, 2004 5:39 PM Subject: [Asterisk-Users] wav49 voicemail problem with Windows Media Player Someone submitted a bug about wav49 voicemail problems with the Windows Media Player here http://bugs.digium.com/bug_view_page.php?bug_id=254 bkw918 changed the status of the bug to resolved because he could not reproduce the error with his version of Windows Media Player. I am having the same problem as the original bug poster. I am using WMP 9.00.00.3075 running on Windows XP and using Asterisk CVS-01/13/04-00:08:32. Is anyone else having this problem? For a quick check click on the bug link above and then try to play the attached wav file with your Windows Media Player. It would be great if you could also verify if wav49 files recorded on your Asterisk machine give the error. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 on FreeBSD 4.9?
I'm about to post on bugs to offer a bounty for work on FreeBSD. I'm fairly certain that others will come along to increase that bounty. Before I do post it, I would like some input on what the requirements should be. Here's what I have so far: - Must be completed before 6/30/04 - Support for all Zaptel hardware - Commitment of the drivers to both 4-STABLE and 5-CURRENT/STABLE I'm not completely conversant on how GPL software can be committed to the kernel, but I believe it can be done under the contrib/ directory. I do not want this work to exist as a series of downloads/checkouts/patches/modules if it can be avoided. I don't want to patch my kernel or load modules. I want to be able to do a cvsup on /usr/src, add necessary device entries to my kernel config file and build it. I'd like to see astersk and libpri installs follow the reccomendations and requirements found in the FreeBSD hier(1) man page. Specifically, it should install completely to /usr/local/. Preferrably, I'd like to see a port created for both asterisk and libpri, even just a metaport that uses CVS to fetch the source and any OS-specific patches. Any comments before I post the bounty? I will recommend that those with suggestions on the requirements and those that offer additional bounties for this will sit in committee to determine when the requirements of the bounty have been met. -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, January 13, 2004 8:27 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk on FreeBSD 4.9? On Tue, Jan 13, 2004 at 12:24:20PM -0500, Jason T. Nelson wrote: love to be able to use Asterisk under FreeBSD. I've browsed the archives and perceived what appears to be a slightly hostile attitude towards those who ask about Asterisk support of other free operating systems even without using Digium hardware. Is this Linux-specific bias intentional or accidental? I would call it historical. Asterisk was first developed on Linux, and little attention was paid to portability. This is changing, though there are still Linuxisms in the code. I would hesitate to consider it stable yet on anything other than Linux, but YMMV. I personally would like to see Asterisk portable to any *nix with pthreads, and am working to make this happen. As always help in the form of patches, testing or accounts for building and testing on less common types of systems are appreciated. -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 on FreeBSD 4.9?
John, I thought you might be interested. I don't know the particulars about driver portability between the BSD's, but it seems that at least on x86 hardware, it should be fairly easy. I'll include those 2 in the bounty. I'm not sure what hier(1) has on the other BSDs, but in FreeBSD it is completely acceptable and desirable to have /usr/local/etc/ for local configurations. /, /usr are only for the base OS. Of course, these are simple build-time configuration options to have. Each OS (even each linux distro) has it's own heir(1) scheme, perhaps the work to get a clean and proper installation of asterisk on FreeBSD will prompt the developers to also have asterisk install itself properly on other platforms obeying their respective hierarchies. John, Do you think you could talk Mark into making some hardware available for test/development platforms if we end up with a non-digium person attacking this? -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Wednesday, January 14, 2004 9:22 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk on FreeBSD 4.9? I'm about to post on bugs to offer a bounty for work on FreeBSD. I'm fairly certain that others will come along to increase that bounty. Before I do post it, I would like some input on what the requirements should be. Here's what I have so far: - Must be completed before 6/30/04 - Support for all Zaptel hardware - Commitment of the drivers to both 4-STABLE and 5-CURRENT/STABLE I'm not completely conversant on how GPL software can be committed to the kernel, but I believe it can be done under the contrib/ directory. I do not want this work to exist as a series of downloads/checkouts/patches/modules if it can be avoided. I don't want to patch my kernel or load modules. I want to be able to do a cvsup on /usr/src, add necessary device entries to my kernel config file and build it. I'd like to see astersk and libpri installs follow the reccomendations and requirements found in the FreeBSD hier(1) man page. Specifically, it should install completely to /usr/local/. Preferrably, I'd like to see a port created for both asterisk and libpri, even just a metaport that uses CVS to fetch the source and any OS-specific patches. Any comments before I post the bounty? I will recommend that those with suggestions on the requirements and those that offer additional bounties for this will sit in committee to determine when the requirements of the bounty have been met. -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638 [snip] Troy - While it is not 100% relevant to your requests, I'd like to see continued support of NetBSD/OpenBSD in this same vein and added to the bounty, since the additional work to get things correctly functioning on those two systems seems to be fairly minor while the hood is open. MacOS is a different animal, and (IMHO) lower on the must-have list when it comes to Zap device support, though it would still be cool. If OpenBSD (1st choice) and NetBSD (2nd choice) can be added for Zap device support, count me in on the bounty. Talk to me privately if you want to get a dollar figure. I've had * running on OpenBSD, but of course no Zap hardware. I'd move everything over to OpenBSD if it supported Zap, since that's my primary OS for all the platforms in my network. While Linux in it's various flavors is great, it's simply not what my network runs, and so my * boxes are the odd man out systems, which makes me somewhat uncomfortable from a security and management perspective. Additionally, if files are to be installed in /usr/local, then I'd like to see the configs remain in /etc/asterisk since on my systems (and many other people's) the /usr/ directories are for binaries only; no configurations or moving parts so those directories can be mounted read-only or mounted from a common server if necessary. I'm sure this is what you meant, but I've seen config directories unwisely located in /usr/local before, and I wanted to make sure everyone is of the same mind where that is concerned. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BOUNTY POSTED - Zaptel drivers for *BSD
http://bugs.digium.com/bug_view_page.php?bug_id=847 I'm placing a $250 bounty on getting Zaptel drivers working under BSD. While the 'hood' is open on this, we'd like to have drivers completed for FreeBSD, NetBSD, and OpenBSD on the x86 platform (other platforms optional). This needs to be done in a timely fashion. I'd like it completed by 6/30/2004, but once the project is started, we can adjust the deadline if it's not reasonable. In addition to the Zap drivers, we will require that patches be submitted so that libpri, asterisk, and other parts of the project have clean and proper installations on a per-os basis, as determined by each system's hier(1) man page and ports/package system. In addition, I spoke with Chris Coleman from Daemon News/BSD Mall today, and he's willing to lend the hardware and place an additional $250 bounty on the project. I would like [EMAIL PROTECTED] to donate/lend the hardware instead, as I believe he can do so more easily than BSD Mall can. I have not yet spoken to Mark about this yet. If I understood Chris correctly, he's going to work on finding a developer to write the drivers and get them committed to FreeBSD. I'm not up on Net/Open, so don't ask me about those, but I do want them included in this bounty. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] after hours
They told me not to feed the trolls, but here goes anyways. The difference between 9AM and 9PM is 12. 12 + 9 = 21 So, if you want 9PM, use 21:00 (which you cleverly did include in the example you asked about). Now, as for different schedules on different days, here's what I have: ; First, let's do the holidays include = holiday|*|*|1|jan include = holiday|*|*|31|may include = holiday|*|*|4|jul include = holiday|*|*|6|sep include = holiday|17:00-23:59|*|24|nov include = holiday|*|*|25|nov include = holiday|17:00-23:59|*|24|dec include = holiday|*|*|25|dec include = holiday|17:00-23:59|*|31|dec ; these are the days we're open include = day|09:00-19:59|mon-fri|*|* include = day|10:00-14:59|sat|*|* ; if we're not open, we're closed (duh!) include = night -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, December 18, 2003 3:27 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] after hours When setting include = daytime|9:00-21:00|mo-fri|*|* How does this determine what is different between 9 AM and 9 PM And after hours ??? I want different hours on Saturday and Sunday And a different welcome message after hours Any help appreciated Regards Mick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hybrid T1 Service (WAS: Channelbank Recomendation and GS102 question)
-Original Message- From: Walker Haddock Sent: Thursday, December 04, 2003 7:54 PM To: [EMAIL PROTECTED] We have an installation with 9 inbound voice channels (one is the fax) and 768K data. It is a Hybrid PRI. It terminates into a T100P. It is working great! The cost was better than the POTS plus data. This is a service that I'm interested in selling. Would you be willing to share with me (the list) exactly how you have this set up (read: your configuration files)? I've never used linux as a router, and am a bit leary of doing this and selling it as a supported service. I've got the voice stuff down I think, my primary interest is in how you accomplished the data portion. Thanks, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Channelbank Recomendation and GS102 question
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Thursday, December 04, 2003 8:48 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Channelbank Recomendation and GS102 question At 8:15 PM -0500 12/4/03, Jim Flagg wrote: - Original Message - From: Walker Haddock [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, December 04, 2003 7:54 PM Subject: Re: [Asterisk-Users] Channelbank Recomendation and GS102 question We have an installation with 9 inbound voice channels (one is the fax) and 768K data. It is a Hybrid PRI. It terminates into a T100P. It is working great! The cost was better than the POTS plus data. Can I ask what Telephone/Internet service provider you are getting this from? Does anybody else have a setup like this? Very interesting. I've had now two fights with providers (Verizon and SBC) who would not offer such a service, claiming that it was impossible to hybridize a PRI. I think that's a great offering, and of course, it is possible, and especially appealing for Asterisk users. I, too, would be interested in hearing from what vendor you are getting such a service. John, Check the front of your local phonebook for CLEC listings. In your area, I'd expect to find at a bunch listed, and at least two or three that are facilities based, capable of serving most areas in the Willamette Valley (Vancouver down to Eugene). If not, perhaps there's a good business for you to investigate. =D Our CLEC here, KMC Telecom, does the hybrid T1 thing as a matter of course. I can have a 6x6 system delivered to my customers for less than $400/month (.09/local connect), or unlimited local outbound for less than $500/month. KMC even provides the customer with a Lucent Connectreach, which breaks out the POTS lines and can hand off the data either as a FT1 or Ethernet. I'd like to play with using * to do it all, but need to find a qualified guinea pig first. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Web Admin - was:Re: [Asterisk-Users] CDR Web Search Frontend
Again, we need to seriously consider moving this to a separate mailing list and getting a 'Features' thread started, as well as a 'Mission' thread. These should get everyone's feet on the same path. I agree that the web administration application needs to be be something different than simply displaying the configuration file. By the time we're done, I think it would be ideal to have abstracted the entire * configuration and store it in some sort of organized fashion (flat-file, RDBMS, XML, whatever). -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tjardick van der Kraan Sent: Thursday, October 02, 2003 7:33 AM To: [EMAIL PROTECTED] Subject: Re: Web Admin - was:Re: [Asterisk-Users] CDR Web Search Frontend - Original Message - From: WipeOut [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, October 01, 2003 12:09 PM Subject: Re: Web Admin - was:Re: [Asterisk-Users] CDR Web Search Frontend I think there is room for everyones ideas, the more the better.. The biggest problem I see with these things is that many people seem to end up developing in parallel streams and the result is 5 seperate projects all half baked and incomplete.. What is needed is for everyone to pool their efforts and come up with a definitave web application to run on top of Asterisk.. That's why i sent out this mail as the last thing i want to do is start on something where 5 others are starting on on their own too. We just need someone to take on the project and if someone is ready to do so then fine if not i'll be happy to keep track of features etc and people that are willing to put in their time and effore on this. But again i don't want to step on anyone's feet in case they are allready doing this. Maybe a php-dev mailinglist might be a good help here too ? Anyway I am rambling.. So I will stop now.. No you wheren't ;) Greetings, Tj ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Any way to get out of a remote console without stopping *
-Original Message- From: Martin Pycko Sent: Thursday, October 02, 2003 4:13 PM use quit or ctrl-D Martin From what I can tell, * doesn't honor EOF, at least I've had no luck with it. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Posting Styles (WAS: Google newsgroup or Forum setup.)
-Original Message- From: Roderick Montgomery Sent: Tuesday, September 30, 2003 8:24 AM According to Troy Settle: Why do they do that? Quite possibly because they, like myself, hate having to scroll through pages and pages of quotes to get to the reply, which isn't always clear where it might start. Troy, you're not complaining about bottom-posting; you're complaining about folks that don't trim their quotes down to context. See how I left out all of your previous post except the relevant question above? You only need to quote enough of the previous message to gain context for the reply -- only lazy folks quote the entire message, reposting the entire thread with every new reply. Your replies go below, so reading the message from top to bottom is in chronological order. My posting style changes from thread to thread and even message to message. It depends entirely on the material I'm responding to. If I'm the first respondant, I'll top-post or go inline. In this case, we're discussing a general idea so it doesn't make a huge difference IMO. In some cases, there are several individual ideas and/or points that need responded to, so I break up the original and respond inline with the quoted material. In other cases, like this one, the (sub)thread is going with bottom posting, so I follow suit. FWIW, if you search for my name, you'll see thousands of posts over the last decade, and while I could be mistaken, I don't believe I've ever been flamed for my posting style. I also believe that this is the first time that I've ever even responded to a complaint about top/bottom posting. The one thing that does drive me crazy, is when people reply to a thread, and neglect to quote anything of what they're responding to, and you can never tell who or what their remarks are regarding. BTW, your line wrap seems to be a little too long, but I wouldn't swear to it, as Outlook sometimes does funny things when quoting. Ciao, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Google newsgroup or Forum setup.
-Original Message- From: Steven Critchfield Sent: Tuesday, September 30, 2003 9:23 AM On Tue, 2003-09-30 at 07:53, costas wrote: See my Mon, 29 Sep 2003 10:30:23 -0400 email (Sorry emails have no message #s to refer to :) ) This is why top posting bites. What the hell are you talking about? He's referring to a message previously posted to the list, which is in bad form. He should have included his comments with proper quotation and citation. I no longer have the message he's referring to in my folder, so I'll have to go hunt through the archives to see what he's talking about. Or not. It's not important enough for me to spend the time on. FWIW, referring to message #'s in web forums is /worse/ than quoting in a mailing list. It will often force you to click back to a previous page, which forces you to click over to another page to read another reference, etc... Of course, you can also use the BBCode [quote], but that's also a pretty ugly solution as well. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CDR Web Search Frontend
-Original Message- From: Jamie Carl Sent: Monday, September 29, 2003 7:44 PM Guys! I'm putting the source up on SourceForge on my existing account. Questions is this tho: Suggestions please! I would like to get this on SF by the end of the day. (it's 9:33am here). Someone I know once said: I'm not a coder, I'm just an idea man. Well, I'm not a marketing man, just a coder. With all the discussion about licensing issues and the sort, I think it's time for a full blown 3rd party application to work with Asterisk while at the same time not causing Asterisk to become encumbered. For such a project, I'm license neutral. While I prefer the BSD license, the GPL would work just as well for such a project. I'd say the first order of business, is to move this discussion to a separate list so as not to annoy the purists. Perhaps Digium would be willing to host it? Call it Asterisk-Addons and let us go have some fun? Here's some general thoughts on the project. For management, the interface and API is already defined. The only way I can think to improve this, is with the addition of SNMP read/write as well as traps, but I'm sure that would create additional licensing issues for Digium. For general configuration, we can write text configuration files. We could also add a hook in Asterisk to tell it to obtain it's configuration from a different fd (one that opens a socket to our stand-alone system). Doing this would allow suckers (like me) to run a live configuration from a MySQL database, and would allow Digium to say too bad it crashed, but we can't help you if you're not using the default configuration schema. For accounting, asterisk can write to a named pipe instead of or in addition to the default csv file. The only other things I can think to add, would be an option to send data to syslog as well. Additional hooks can be defined if needed, as long as we used named pipes or sockets or whatever to maintain a seperation. Doing it like this would keep Asterisk at arms length from the encumbered code, allowing Digium to keep the dual license option while also allowing the rest of the world to explore the possibilities. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Google newsgroup or Forum setup.
Don't tell me, but your next request is going to be to move the IRC discussion to AIM? Look, for all that a web forum does, a mailing list is /so/ much quicker. Messages to the list are pre-sorted into folders, which I access via IMAP from Outlook. Outlook sorts them by subject/date, I pull up the first message with my mouse, and from there on out, it's down arrow or delete. I can read through hundreds of messages without ever touching my mouse (yes, I use windows, but hate the mouse). I haven't played with it much, but from what I remember, Evolution is almost as easy to use with/without the mouse. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of costas Sent: Sunday, September 28, 2003 5:12 PM To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: [Asterisk-Users] Google newsgroup or Forum setup. I am sure this has been asked before, but why not use Google newsgroup or at least some forum BBS software instead of this cumbersome mailing list process? -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with GPL license of Asterisk
A few answers: 1) if your application is not released to a 3rd party, you do not have to make the source available 2) if you build your application as a module that loads into a stock asterisk server, you do not have to disclose your source 3) if you need to make changes to the core in order for your application to work, you'll need to disclose source for your changes to the core, but not for your application. This sounds horrid, but it's not too bad, as your simply augmenting the core API and keeping your goodies in the binary only portion of the release. With that said, if you're writing an application that you would like to sell, your IP lawyer should be able to easily decipher the GPL and advise you as to which parts of your code need to be made public. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of costas Sent: Monday, September 29, 2003 8:38 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Help with GPL license of Asterisk I would appreciate some help with this. I read the GPL license and basically it says you can do whatever you want with the software (sell, modify) as long as you include the source code, the License and make any changes you make available in the same manner to all others. My questions is this: If I develop an external application (say a Call Center application or a GUI management application) that uses Asterisk data is that also GPLd? I understand if I add code to Asterisk, but what about external interfaces? Where is the seperation here of the Cathedral and the Bazaar? Thanks -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CDR Web Search Frontend
After reading this thread, I wonder if we couldn't create an abstract RDBMS interface in the core, then in the addons repository, create modules for MySQL, MSQL, MSSQL, Oracle, Sybase, PGSQL, or whatever. The abstraction layer would simply use whichever addon module it was configured to use. This would eliminate the licensing issues Digium may have when granting a non-GPL license to a 3rd party, while allowing creating a very flexible product. Additional functionality could be added with some simple hooks in the core. For example, if you want your configurations to come directly from a SQL database in real time, add hooks into the code to cause * to read from the abstraction layer instead of it's internal database. It would be simple enough to do, and as a build-time and/or run-time configuration option, I believe it can be done without causing issues to the stability of the core. Another example is CDR, which would do what it does now by writing a csv, but could have a hook added to it to also send accounting data to a SQL server or even a Radius accounting server or something else entirely. To do this right would probably force a major architectural change to *, which I'm sure that Mark would be reluctant to do, but if someone could build the framework, get some folks to test it (in small, medium, and large deployments), I'll bet Mark will accept it. Also, with the advent of the addons repository, has anyone asked Mark if he would be willing to grant commit privileges to more people? It doesn't make much sense to create a new repository for this when Digium already has the facilities available and a new repository for non-core items. BTW, I'm not a coder, I'm just an idea man. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Evans Sent: Monday, September 29, 2003 5:37 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] CDR Web Search Frontend Or do something really smart like the Perl guys and have a backend-mostly-independent DB infrastructure. Hell I think that PHP finally smartened up and went this way, too. Hi Guys I am happy to do this and send the code back. Database independence isn't to hard to achieve. It would be nice if a group of us could get together and discuss how we can make this great app even better and possibly look at getting a small team together to merge this and phpconfig into a single application. Will possible access to cvs for the developers. Thoughts? Mark Evans SiteTel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Google newsgroup or Forum setup.
Actually, top posting, and yes, people do that. Why do they do that? Quite possibly because they, like myself, hate having to scroll through pages and pages of quotes to get to the reply, which isn't always clear where it might start. With top posting, you know the reply starts at the top, and stops at the signature and/or citation. PS, this is /way/ off topic for this thread and this mailing list, and is best dropped. In fact, I feel really bad about hitting the send buttin in about 2 seconds... -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Monday, September 29, 2003 1:37 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Google newsgroup or Forum setup. Top-quoting. Argh. On Monday 29 September 2003 12:16 pm, Keith O'Brien wrote: I'll offer one better. Why don't we mirror all of the maillist posts to a forum. That way both parties are happy. Those that want a forum can use a forum interface and still post to the maillist and those that like the maillist can stay as is. Because the point was that forums, while their proponents feel is the next best thing since sliced bread, don't actually get very much traffic. There's far too many projects out there (Sourceforge, anyone?) which have died due to the dearth of people checking the forum for posts. Note that the mailing list is archived in several different places, and everything is indexed by Google. If the one provider hosting a forum has a catastrophic failure, there isn't much in the way of backups. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Interface with PBX
I'm doing the following to integrate * and a Partner ACS using an 8x16 Zhone channelbank. Channel 1-4 = FXS (extensions) on Partner Chanenl 5-8 = POTS/PSTN Channel 9-16 = FXO (CO lines) on Partner This setup is working pretty well, except for a few issues with call supervision on the Zhone. Incoming calls are answered by *, then placed into a call queue that will ring into the pooled lines on the Partner system. If the caller dials an extension, * dials via one of the extensions (channel 1-4). This works well, except that it sees the line as answered immediately. If I turn on callprogress, it never sees the line answered, even when it is. For outbound, calls are routed to a 2nd * server in another location. Eventually, my inbound calls will come from the second server as well. Eventually, I'll likely drop the partner system and wire everyone directly to the Zhone (almost everyone uses cordless phones anyways). -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Crick Sent: Friday, September 19, 2003 2:37 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Interface with PBX I'm trying to interface * with a PBX, but seems that his ring cadence is somewhat different, and my T100 doesn't show any call coming in. Yeah, I had a similar problem - I was trying to connect an X100P to a small 3x8 analog PBX for testing and it wouldn't grab the call. Thinking about it now, maybe I should have turned caller ID off? Hmm.. Is your T100 connected to a channel bank with FXO ports connected to PBX FXS ports? Or are you using PRI connections? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail notification email with no attachment despite attach=yes
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Marc V. Liotier Sent: Wednesday, September 10, 2003 1:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Voicemail notification email with no attachment despite attach=yes The demo 1235 extension that Asterisk ships with works fine and the messages are sent to the address I set in voicemail.conf. I guess that means that my configuration is working perfectly so far. But when I set up another extension with a voicemailbox, no mail is sent when a message is left, although I can dial voicemail and listen to the message just fine which I guess rules out voicemailbox misconfiguration. The strange thing is that both extensions and mailboxes are configured exactly the same : in extensions.conf : exten = 1235,1,Voicemail(u1234); Right to voicemail exten = 6004,1,Voicemail(u6004) in voicemail.conf : 1234 = 4242,Test mailbox,[EMAIL PROTECTED] 6004 = 4242;Other test mailbox,[EMAIL PROTECTED] ^ | Could this have anything to do with it? You're effectively commenting out the rest of the line (if I have a grasp on *'s config parser. In effect, you have a VM box w/password, but no name or email address. I don't understand why these two seemingly identical configuration yield different results. I guess that I must have missed something that was included in the example and not in my new mailbox. Could somebody give me a hint about what it could be ? -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Unexpected Call Termination!
I'm assuming that both circuits to the * box are E1/PRI, so those settings wouldn't make a difference. To the OP, you may want to run pri (intense) debug on the spans to see what's going on. If you are running RBS to the Nortel box, then the busydetect and callprogress may be the ticket. You may also want to adjust signaling to groundstart, which some PBX's seem to use (check w/ nortel on this). -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: Wednesday, September 10, 2003 12:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Unexpected Call Termination! In /etc/asterisk/zapata.conf: busydetect=no callprogress=no On Wed, 2003-09-10 at 02:44, Surajee Ratnayake wrote: hi, I hav a softPBX setup. Our set up has 2 servers, one is connected to an ISDN PRI E1 coming from PSTN central office and the other server is connected to another E1 which is coming from a Nortel PBX. and 2 servers are connected to a LAN. So when a Nortel PBX users want to get an out side call they go though our servers. But there are some complains coming to us saying that most of the calls do get cut after several time. that is when some body is engaged in a call with an outside number, suddenly call terminates unexpectedly. This is very disturbing for us. Can anybody pls help us with this situation. Surajee -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Request for comments on queue statistics
-Original Message- From: Dave Weis On Tue, 9 Sep 2003, Paul Crick wrote: I have done some trival work with matrix orbital lcd to show some stats counts, calls parked etc Just find lcd a bit small do you have lead on bigger LED signs that you have used b4 ?? I've used a Beta-Brite sign which is pretty similar to a ProLite in functionality, just made by a different company. They're on eBay all the time, search for LED sign as well as the two brand names and you're bound to find something. If you need something bigger try www.translux.com I'll stick to my Bright Light(tm), thank you. Would kill for a linux driver for it though! -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ISDN TA
-Original Message- From: Howard White Sent: Wednesday, September 10, 2003 2:35 PM bottom response = on On Tue, 2003-09-09 at 12:41, Robert Boardman wrote: I have an ISDN TA that has 2 POTS interfases (FXS), can these be used with asterisk? Thanks in advance Robb Yes, I have two such installations. Be advised there are some gotchas. My TAs are older Ascend/Lucent/??? Pipeline 75s which have different tones for off-hook and error conditions that * is not always prepared to listen to. I get voicemail messages with four minutes of P75 off-hook every now and then. I am sure that patches could be applied to solve these issues but I choose to live with it as is. No show stoppers, you understand, just oddities every so often. Howard White Howard, I think the OP wanted to go in the other direction... Which I would assume is possible, as long as you also have an ISDN/CAPI card for linux that can be put into NT mode rather than TE mode. I don't know how this would be wired up, but I would imagine something special may need to be done with regards to U vs. ST interfaces, or it may be as simple as crossing the T/R pair and letting them go. Take this all with a grain of salt though, I'm far from an expert on BRI. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fax
David, Could you elaborate on what hardware you're using for this? Also, speaking of faxes and *, I know that you can't fax over IAX(2), but it seems that faxing over SIP (ala ATA-186) works fine? Would it be possible to set up the fax extension on one * box that can then use SIP to get the call to a second * box that's sitting ~10ms away? -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Carr Sent: Monday, September 08, 2003 4:57 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Fax The way we do it is our T1 comes in with unlimited DIDs. In our case we just order more toll-free numbers, each with its own DNIS. Then we have four FXS ports (Zap/g2) connected to hylafax modems. When the call comes in using DNIS 1234, asterisk sets the callerID name to 1234, sends the call to Zap/g2, and our hylafax config routes the fax to email [EMAIL PROTECTED] Then in our mail table we forward each mail alias to where we really want the fax to go. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ernest W. Lessenger Sent: Monday, September 08, 2003 12:21 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Fax At 07:52 PM 9/8/2003 +0200, you wrote: Is there a way to configure Hylafax or sth one modem behind an ATA-186 to email faxes to different adresses depending on the called number ? I've looked into this myself, and I think the answer is yes, with some minor code changes. My thought is that you would use a separate HylaFax server with six modems in it, and add two Digium FXS cards to the * server. Configure * to send the faxes out the correct FXS port for each company, and configure hylafax to queue the faxes to a different folder for each line. User interface and notification are left as an exercise to the reader, as is the actual hylafax configuration :) The major downside to the above is all the POTS lines you have to run, and the waste of ports. An alternative would be to use only one or two POTS, and have * set the CallerID for each company. Then, have Hylafax queue the incoming faxes based on CallerID. The disadvantage to this is, of course, that you lose any real CallerID information. --Ernest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] freebsd and asterisk ?? anyone yet
I know a few people (myself included) are willing to help provide incentive funds to get this going. The big quesiton for Digium: What will it take to get * up to speed and the drivers ported to *BSD? What will it take to keep it there? -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom (UnitedLayer) Sent: Monday, September 08, 2003 4:56 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] freebsd and asterisk ?? anyone yet On Sun, 7 Sep 2003, John Brown wrote: so has anyone gotten * ported to freeBSD yet ?? Everything I've seen points to it being more an issue of Telco HW support, rather than SW support from asterisk. The Digium HW has yet to be supported in FreeBSD/NetBSD, and Asterisk doesn't support the VoiceTronix cards. I've seen a couple posts on the list about VoiceTronix cards, but seen no news of their support. I think the only cards that works with FreeBSD+Asterisk are the Quicknet Internet Line Jack and Internet Phone Jack. Less than optimal... When either one of those happens, I think we'll see more FreeBSD users. Incentive: If anyone is interested in trading coding time towards getting one of these goals accomplished, for colo+BW services, lemme know. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channelbanks
Ok, the Zhone sucks and the Adtran 750/850 seems to be a little too expensive. Can anyone recommend a decent channelbank that won't break the bank? TIA, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] app_queue input needed...
that when/if Mark tackles app_queue, he'll incorporate that as well. Seems like there are several people working on this family of problems (pos/holdtime/fail out) and it might make sense for us to work together some and standardize our approach. Otherwise we're going to end up with 5 versions of app_queue. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Asterisk stops responding
I'm seeing the same thing as well on current CVS code: Asterisk CVS-08/31/03-01:58:51 At this point, * no longer accepts calls on Zap, Sip, or IAX. On an incoming call from the PSTN, * shows starting simple switch on the appropriate channel, but never answers. Both myself and bkw ([EMAIL PROTECTED]) have been playing with this to no avail. Nothing weird shows up in a process list. Nothing else seems lagged out. At first, we were thinking it might be an interupt issue with a serial console (previously discussed on this list), but that wouldn't explain why all 3 channel drivers were bombing out. * was still getting interupt from Zap (single-span T1, Zhone), but was not answering. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jared Smith Sent: Friday, August 29, 2003 11:33 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] RE: Asterisk stops responding On Fri, 2003-08-29 at 08:25, David Harris wrote: This problem is different from mine. I can still reconnect to asterisk with asterisk -r and still issue some commands. But I cannot issue either reload or stop now they return immediately and do nothing. /davidh I'm seeing the same thing as David... Jared ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Twisted Idea
Ok, this is probably an incomplete thought, but over the last few weeks of reading this list, I think I'm ready to start designing my system, and would like to solicit input. Currently, I have 2 POPs, one each in 2 different states. Each has a Lucent TNT w/~400 trunks. Each TNT has the capability to provision a PRI that can be piped into an * system. So, my interface to the PSTN is complete. Now, in my office, I'd like a 3rd * system that's tied into my existing key systme (Lucent Partner). What I'm thinking, is that the * system would handle AA, IVR, and VM applications, transfering calls either into the general pool (via FXS ports to the CO ports on the Partner), or directly dialing an extension via an FXO port. Of course, I'd also end up with a few VOIP extensions as well. Does this sound like a feasable plan? Aside from eliminating the partner system (which isn't really an option), is there anything I'm missing? -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PHP API for Manager - Plaintext auth needed?
I also dislike plaintext, but the vast majority of users will probably run the PHP script on the * system itself, so plaintext won't really hurt. Hell, I doubt that most will even bother to run the scripts on a secure server. I'd say set the default to md5, but leave plaintext as an option. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven J. Sobol Sent: Friday, August 01, 2003 12:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PHP API for Manager - Plaintext auth needed? Quick question: My PHP script is now able to connect to the manager port and successfully authenticate using MD5. I would strongly prefer not to do plaintext authentication at all. Would anyone object to plaintext authentication being left out? -- JustThe.net Internet Multimedia Svcs. [The Fusion of Content Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve Sobol, Proprietor 888.480.4NET (4638) * 248.724.4NET * [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Linux flavor?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Low, Adam Sent: Tuesday, July 29, 2003 9:15 AM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Linux flavor? Personally, I've compiled Asterisk on Redhat and Debian without any problems on either, I think generally Asterisk compiles very easily no matter what the distro but I would recommend that you use the one you are most comfortable/experienced with. Good advice. Unfortunately, nobody with requisite skill has stepped up to get this thing ported over to FreeBSD (though I recently heard a rumor that someone finally got it built). Nothing personal against Linux, in fact I was highly impressed with the RH9 install last week and will likely end up with it on my laptop if not my desktop, but for servers there are a great many people, including myself, who much prefer BSD over Linux. I can't speak for all BSD users, but for myself, the issue is not so much one of trust, it's a matter of (re)learning the linux way. I originally gave up Linux in favor of FreeBSD because in 1996, FreeBSD was better in many ways, most of which no longer apply (just the asthetic ones). -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BSD (WAS: Linux flavor?)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Tuesday, July 29, 2003 12:40 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Linux flavor? Actually, it's been ported over to OpenBSD. It shouldn't be too much of a struggle to adapt Asterisk to FreeBSD, although I haven't tried yet. The real difficulty will be in porting the zaptel drivers over to FreeBSD. BTW, gastman currently compiles and runs on FreeBSD. I'd recommend GTK-2.0, however, over GTK-1.2, as the 1.2 stuff has some critical problems. -Tilghman Right now, you can do a CVS checkout make install on linux. This is awesome stuff. However, for the non-programmer to try to build * on BSD, it is a struggle. As a VOIP only solution, perhaps just a linux binary installation would be adequate to run under emulation. For the development team to get * (and the zaptel cards) running on BSD shouldn't take too much effort. Perhaps it's just a matter of finding the right incentive? My only request would be that it be installed to match BSD filesytem standards (everything in /usr/local). -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk as a stand alone voice mail server
Funny. I just subscribed to this list to ask the exact same question. The application I have in mind though, would be a little more intense. What I would like to create, is a unified messaging center for voice, fax, and follow-me service (home, office, cell, pager). -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronnie Earle Sent: Wednesday, July 23, 2003 12:35 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk as a stand alone voice mail server I'm sure asterisk would make a great stand alone voice mail server. Basically I want to get rid of our voice mail system and replace it with *, but the problem is we use a cisco cluster with skinny clients. So I was thinking the way to contact a * server, would be through our 3640. But so far any attempt has failed. I am wondering if anyone has done something similar. Just want to verify the idea is sound. Please keep in mind I just heard of * a few days ago and don't know much about it. Though it seems pretty easy to use. At least configuring a couple clients was not that tough. Thanks to John Todd for his easy to follow guide at www.onlamp.com. Anyone with something similar? if so some info on what you did would help a lot. Thanks all, Ron E. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
