Re: [Asterisk-Users] how does one get the very best quality output?
Clayton Smith wrote: Hi, i'm trying to send some songs over via asterisk, so i'm trying to get the very best quality possible i've been using gsm, using sox with a rate of 8000, single channel, resampled q1, and got some good results, but i'm wondering if there is at all a better way I'm using voicepulse, which supports * GSM * G.711ulaw * G.711alaw * ADPCM * ILBC * SPEEX any of those better to send music through G.711 is a lossless codec, so either G.711 would be better than a lossy codec like GSM for sending music. Nick ___ 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] Soft DSS for Asterisk
Wiley E. Siler wrote: Is there a Software based DSS application available for Asterisk? Yes... look in the wiki. VoIP people just call them GUIs though. :-) Nick ___ 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 not starting - SOLVED!
Andreas Roedl wrote: Am Sonntag, 8. August 2004 19:45 schrieb Dave Cotton: The problem's somewhere else I'm running Asterisk CVS- HEAD-08/07/04-22:38:39 with all the fpm.mp3s as is with no problem. But not on gentoo, what version of mpg123 does gentoo have? 0.59s-r3 There is only one version of mpg123 to run with Asterisk, an that version is 0.59r. Anything else is just asking for trouble. You may still have to strip your ID3s out, but you'll probably find that .59r just works. Nick ___ 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 scalability?
Roy Sigurd Karlsbakk wrote: Hi I plan to setup an asterisk box to function as a SIP gateway forwarding lots of calls to/from a backend of several other asterisk boxes, each with a TE410 card for PSTN connectivity. It will only gateway the calls into the PSTN gateways. No transcoding is planned - only plain ALAW. How many concurrent calls would you think this can handle? I'm asked to plan a system that can handle 1000 concurrent calls... Search the archives and the wiki. Look for a thread a few months ago called Asterisk on 64-bit Nick ___ 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] Successfully Using $135 Avaya sip phone
Brian Elton wrote: I think I am the first to use the $135 Avaya 4602 SIP phone, but I need some support from the community to fix one problem I have with it. The phone stops working after about 20-30mins if I have mailbox=context in Asterisk; when I do have mailbox=contect in asterisk the sip debug returns 481 extension does not exist. Anyone willing to help me figure out why? I have two debugging suggestions for you: 1) Upgrade to latest CVS 2) Try using ethereal to look at the SIP packets going back and forth before the phone stops working Nick ___ 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] VoiceMail Not releasing
Steve Totaro wrote: [I think you'll find that inline-posting makes treads easier to read] - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 30, 2004 9:59 PM Subject: [Asterisk-Users] VoiceMail Not releasing About twice a week we have a caller that comes in and hangs up on voicemail. We have 2 x100ps each with their own irq. When the caller hangs up asterisk does not release the line. The line rings busy, sometimes I can do a soft hangup Zap/1 and release the line sometimes I have stop asterisk and remove and re-insert the modules. I have the same issue with IAX2. I get messages anywhere from 5 min to 45 min of silence. Look in your voicemail.conf for maxsilence and silencethreshold: ; How many seconds of silence before we end the recording maxsilence=10 ; Silence threshold (what we consider silence, the lower, the more sensitive) silencethreshold=128 Nick ___ 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] Source for 9-911 Labels to attach to phones?
John Fraizer wrote: [Please don't top post. Conversations on mailing lists flow more logically when you post inline.] That should be exten = 911.,1,blah and exten = 9911.,1,blah You don't want to not catch a call when the user is scared to death and hits too many 1's. Better yet, with some help from the wiki: exten = 911.,1,ChanIsAvail(Zap/1) exten = 911.,2,Dial(Zap/1/911) exten = 911.,3,Hangup() exten = 911.,102,SoftHangup(Zap/1-1) exten = 911.,103,Wait(1) exten = 911.,104,Goto(1) exten = 9911.,1,Goto(911,1) Nick ___ 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 for a large scale implementation
Harry Schechter wrote: I am looking at Asterisk for a large scale implementation. I was wondering if anyone had any experience (that's code for nice things or not so nice things to say about it) with Asterisk for 50k plus users. I don't have 50k users, but I can say that it's definitely possible with Asterisk; however, you'll need a lot of planning -- it will certainly be a measure-twice-cut-once proposition :-). You're going to need to learn about database-based config files, IAX trunking, affinity-based load balancing, and likely what a good consultant charges :-). Look for the thread Asterisk on 64bit from a while back for some of my ideas for a large-scale implementation and some interesting counterpoints to those ideas. We can probably help you more if you can give us more details. Are you planning on IP phones or analog stations? Will you be doing PSTN termination? With a VoIP or traditional carrier? The more details you can provide, the better. Feel free to reply to me directly or back to this list (if it's appropriate. I'm not sure what the proper netiquette is). Please report to the list or Wiki what your experiences are, whatever they may be. A lot of people have speculated on what it would take to do a massive implementation, but nobody has really proved us right or wrong. Nick ___ 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] Source for 9-911 Labels to attach to phones?
Greg Hill wrote: On Mon, 26 Jul 2004, John Fraizer wrote: That should be exten = 911.,1,blah and exten = 9911.,1,blah You don't want to not catch a call when the user is scared to death and hits too many 1's. won't you need _ in it (_911.) in order to make it do pattern matching? Yes, I think that was just a typo. I did it too :-). Nick ___ 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 G.729 codec and VLANS
Anton Tinchev wrote: The readme says that the license uses all network cards MACS The MAC address is unique a 6 byte address assigned to every 802-family (802.1 Ethernet, 802.11 wireless, etc.) network interface. What happens when VLANS are added or removed? Nothing... VLANs have absolutely no effect of MAC addresses; a VLAN is just a virtual partition within a switch. Is it safe? Completely. Nick ___ 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 Asterisk bounty: SIP simultaneous
Mike Machado wrote: On Sun, 2004-07-11 at 12:31, Paul Mahler wrote: The whole point of a SIP registration is to identify a UNIQUE device. You CAN'T HAVE multiple devices registered as the same SIP device. That's WHY the last device that registers gets the traffic. This doesn't have ANYTHING TO DO WITH ASTERISK. This is a SIP issue, not an Asterisk issue. You should just be happy that Asterisk will do what you want, even if SIP won't. If you really, really want to do this, up the bounty to about $50,000 and get the SIP specification changed. Did you even read the RFC? Section 10.2.1 clearly talks about adding multiple bindings to the same address-of record. Just to quote and save everybody the searching: Once a client has established bindings at a registrar, it MAY send subsequent registrations containing new bindings or modifications to existing bindings as necessary. The 2xx response to the REGISTER request will contain, in a Contact header field, a complete list of bindings that have been registered for this address-of-record at this registrar. Nick ___ 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 Book
Steven Critchfield wrote: On Thu, 2004-07-08 at 09:16, [EMAIL PROTECTED] wrote: I do not recall telling anyone 6 weeks, My book located at www.saww.net/asterisk/ is being shipped to everyone that has not received their orders as of next week. maybe next time you should get your facts straight before lieing in this mailing list I do not have a problem that you are trying to write your own book all the best wishes but lies do not help Be aware that the URL you just posted says it is Backordered, ships in 1-3 weeks. Be careful when you say someone is lieing. It may be true, but unless Or lying, as us spellers* like to call it. :-) you can back it up as absolutely false, it could be called a small exxageration. I hope somebody copy edited his book. His is not the very worst grammar I've ever seen, but I'm fond of (correctly used) punctuation in books that I read. Since there isn't a sample chapter available to assuage my fears of sloppy prose, I'll save $57 and just hope there's nothing good I'm missing. Based on the subtitle and blurb, however, I think I have a good idea of what's inside. I don't mean to personally attack the author, but the dude has scared me away from buying it. Nick *I realize Steven misspelled it on purpose ___ 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] tdm400p static - out of ideas
Ryan Courtnage wrote: Hello, Over the past several weeks, we have been having an intermittant problem with our deployment of a TDM400P card (4 fxo). We have tried many things, and the problem still re-occurs. The Problem: Occasionally (every 48 hours), the TDM400p card will stop answering incoming calls on ALL fxo ports. Attempts to send outbound calls on any Zap channel will result in hearing a loud 'static' noise on the line. Let's look at some possibilities of line problems: What time does it stop answering? Is it ever during ALIT times (usually very early morning)? Have you tried calling the telco to see if it could be their problem? How far away from the CO/mux are you? Have you tried a new/different card? If you didn't want to fork out the cash for a new one, you could try a X100P/knockoff* on one of the lines to see if that fixes the problem... if so you can deduce a bad card. Nick *I usually don't recommend the knockoffs, but for a day of testing $10 sure beats $100... everybody else should support Digium! :-) ___ 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] VoIP hackers gut Caller ID
rich allen wrote: this is really simple, companies like Nortel, Lucent need to change their code for caller id, if the number should be blocked then dont transmit it to the far end switch That's a really bad idea. Even worse than top-posting. My local PSAP should know what number I'm calling from, because I'd like police/fire/EMS units to show up at my house if I can't tell them where I'm calling from. My phone company would also enjoy knowing where the call came from for the sake of preventing toll fraud from any Tom, Dick, and Harry with a SS7 connection. If CLID is blocked (or presentation restricted in SS7 ISUP parlance) only networks should see the Caller*ID, never users. This is a situation where network operators must not abrogate their responsibly to make and enforce policy; software solutions to policy problems are never panacean, just as policy can't fix an unencrypted password file. Nick ___ 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] VoIP hackers gut Caller ID
Chad Whitten wrote: this is true, but Bellsouth (our local RBOC) only allows numbers in our DID range to pass. I can set the outbound caller id to anything, but if its not in our DID range, then the lead number of the DID range is sent out. Are other telco's not doing this? No, not as a rule. And if you complain, the ones that do can make it go away, Nick ___ 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] Re: iax or sip
Randy Bush wrote: 1. Control a call, (maybe you want to do some ACL type filtering, maybe you want to keep track of usage, maybe you just to be in control...) Hmmm. Post setup, which clearly needs to go through all servers (or pbxen) in path, I don't see a win here. Send more clue. More likely is that the phones on the LAN are SIP and the boxes on the WAN are talking IAX, since that seems to make the most sense to me. hide one end from the other. I have a customer and a carrier. I don't want one to know who the other is lest they get together and cut me out of the equation. Yikes! Despite ad homina on this list, even I am not that sneaky. But I can see folk having legitimate needs such as this in an emerging market in desperate times. It's not always so sneaky... imagine that I'm a VoIP provider targeting homes and small business. I'm best to buy minutes in bulk from ATT and/or MCI or any carrier who can offer super-cheap rates in bulk. These carriers don't sell VoIP to home users, they sell it to people like me. I still don't really want my users or competition know where I buy my minutes from, nonetheless. And ATT doesn't really want a direct connection to the home user. So it works out that I am the logical middle-man, especially if I can trunk calls within a few hops of the user (like if I'm also their DSL provider as well) to save everybody bandwidth. Most significantly, this hierarchal paradigm is the most familiar to telcos and telephone people in general. Nick ___ 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] Answering Service Agent Auto Login
Michael Blood, Matraex, Inc. wrote: Hello all, I am building a software based on asterisk to handle incoming answering service calls. I have one problem that I have not been able to figure out a reasonably priced solution to: The goal of this software is to allow the agent to be able to do their entire job from the desktop. The only thing that seems to be a problem is getting the operator (agents) headset logged on to the asterisk system using a computer command. How about AgentCallbackLogin([AgentNo][|[EMAIL PROTECTED]) or AddQueueMember()? Nick ___ 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 Causing Cisco 7200 Router to Crash?
Brian Wilkins wrote: On Yaum al-Arbi'a 12 Jumaada al-Awal 1425 05:42 pm, Harold Workman wrote: As far as loosing the configuration...the only reason I could see that happening is if you either are doing one of the two... not saving the configuration...or you have the configuration register set to something like 0x2142. look on show version for the configuration register. it should be 0x2102. And again, i would look for tracebacks...it could either be a memory issue or a bug in the IOS. But you will know if you get console access to the router as u bring up the asterisk... [I've fixed your top-posting... threads are much easier to read if you reply inline] A traceback is not possible. The best thing I can show everyone is the reboot message. You might try using a packet sniffer like Ethereal on the Asterisk box to see what is happening leading up to the crash. The logs got obliterated when the Asterisk server started up and the best we can imagine, sent an invalid code to the router. This isn't an Asterisk problem; routers should NOT crash no matter what packets are sent -- especially good routers. We are going to set up a small test subnet here and bounce around on the router to see what is the problem. The Cisco 7200 router uses the IOS that is optimized for VoIP. The server caused our fiber card to burn up and now we have to replace it. If your fiber card is no longer functioning, there is more to this issue than some malformed packets... is your router on a UPS? Is the cable run to the router over 100m/300'? Nick ___ 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 64bit ?
Kevin Walsh wrote: Nicholas Bachmann [EMAIL PROTECTED] wrote: Kevin Walsh wrote: Dr. Rich Murphey [EMAIL PROTECTED] wrote: How do you balance the number of active connections per server? In theory, you could use a load balancer. That's as long as you can share the SIP/IAX registrations between the nodes. I'm not sure if that can be done yet - I haven't looked into it. It can. SIP registration info can be stored in a database; see http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers Sorry - I meant the information relating to registrations that have already been made. Like you get when you type sip show users. The database stores everything about a SIP user in the DB: name, secret, IP, etc. Perhaps that's not necessary anyway; The user should attempt to re-register if the connection is broken, and may find itself connecting to a new server automatically. I think you misunderstand; with a LBR and registrations in a database, the user would never know his * box went down unless he was in the middle of a conversation that had the box in the media path. The SIP phone would never have to reregister until the regular registration timeout. Nick ___ 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 64bit ?
Dr. Rich Murphey wrote: How do you balance the number of active connections per server? [Rich, this deep in a tread, it's helpful to everybody if you post inline] TCP load balancers can make affinity tables (or just affinities in Cisco parlance) that map clients outside the LBR to servers. There is some logic to take clients out of the table after a timeout so that they can be evenly rebalanced later, and the result is a near-even distribution of clients between servers. There is no need for the LBR to understand SIP or RTP, it only makes tables to say what clients use what server. If the LBR detects a downed server, its clients are remapped to a new server when they make new requests. On most units, administrators can also gracefully ween servers out of the pool for scheduled maintenance. The downside of course, is cost. ATI's Rapier 24 L3 routers (which I use because I know they're comparatively cheap and have lots of good features) run about $100/port, plus a feature license for load balancing. Soft load balancers also exist, such as the free Linux Virtual Server that somebody else pointed out. LVS is capable of doing affinities, but I'd pay the $3-5k for a decent LBR before I trusted a $400 PC; the LBR does create a single point of failure, unless you've got an LBR that supports a fail-over balancer (which the ATI units do). Nick ___ 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 64bit ?
Kevin Walsh wrote: Dr. Rich Murphey [EMAIL PROTECTED] wrote: How do you balance the number of active connections per server? In theory, you could use a load balancer. That's as long as you can share the SIP/IAX registrations between the nodes. I'm not sure if that can be done yet - I haven't looked into it. It can. SIP registration info can be stored in a database; see http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers Nick ___ 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] Can one send CLID NAME over PRI?
Jason Kawakami wrote: Subject: [Asterisk-Users] Can one send CLID NAME over PRI? Reply-To: [EMAIL PROTECTED] Is it possible to send CLID NAME on a PRI? The numbers we send out are being received by telco and propagated, but the names we send out are not showing up. Is this a feature in PRI? Do we need to set PRI_NET instead of PRI_CPE? Is this just not possible? Is this a telco config issue? my experience with this is the carrier is doing some matching of presented number (what you are sending them) to their records of the 'owner' of that number. You're correct; the telco takes the number you've sent and uses their own database to figure out CLID name. If for example you present your billing telephone number to them they will send out your name with the number but otherwise I have had little success. I have heard of working with a local/regional ps or carrier 911 coordinator to have this fixed but have had no experience with it. If the OP is trying to get correct location information to a PSAP, he might have the telco set up several phone numbers with the physical address of each building, so that when dialing out, they see the correct location of the emergency call. This would be easy enough to set up in the dialplan or even sip.conf. Nick ___ 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] License and Commercial Use
Miroslav Nachev wrote: Hi, I can't find anywhere on the Asterisk web the license terms for commercial use of Asterisk software. Do I have to pay something (and how much) if I want to use the Asterisk in our IP PBX solutions? From a README: * LICENSING Asterisk is distributed under GNU General Public License. The GPL also must apply to all loadable modules as well, except as defined below. Digium, Inc. (formerly Linux Support Services) retains copyright to all of the core Asterisk system, and therefore can grant, at its sole discretion, the ability for companies, individuals, or organizations to create proprietary or Open Source (but non-GPL'd) modules which may be dynamically linked at runtime with the portions of Asterisk which fall under our copyright umbrella, or are distributed under more flexible licenses than GPL. If you wish to use our code in other GPL programs, don't worry -- there is no requirement that you provide the same exemption in your GPL'd products (although if you've written a module for Asterisk we would strongly encourage you to make the same exemption that we do). ___ 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] Voicemail
Gunnar Schaller wrote: J Which voicemail is current and latest? J Voicemail J or J Voicemail2 I didn't want to reply to the original post with the answer, because: * This question has been answered numerous times already. * The poster MESSED UP THREADING by replying, erasing the body, and writing a new message. I think Voicemail ist the latest. No, voicemail and voicemail2 execute the same code; the two names exist for backward compatibility. Nick ___ 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 Sunday News: Off track with 1.0, moving forward
Olle E. Johansson wrote: The decision is to base the future 1.0-release on the CVS head tree. The current stable-1.0 tree will be released as something intermediary, maybe 0.91, and at that point it will be considered end-of-life. At some point when we have cleared the bug tracker from major issues, we will fork a new stable-1.0 tree and start working on that. As a community, we now need to focus on solving all the bugs in the CVS head tree. We need help, Mark Spencer can't handle all bugs by himself. So when reporting bugs, make sure you are available for questions and testing. Any patches in the bug tracker that you can test, test. Report your findings to the bug tracker, both good and bad. Does Asterisk have a test plan for releases? It seems like if there was a plan for testing that people could carry out (in a distributed manner), our releases might not have so many quirky bugs. Most testing can be automated; only some would have to be done in an interactive manner with real people. Nick ___ 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 Sunday News: Off track with 1.0, moving forward
TC wrote: Does Asterisk have a test plan for releases? It seems like if there was a plan for testing that people could carry out (in a distributed manner), our releases might not have so many quirky bugs. Most testing can be automated; only some would have to be done in an interactive manner with real people. Nick Is Nick volunteering to write test cases (for the next 12 mths) Yes, I would happily write some of them. I would love to see how we could write automated tests for all the different config combinations for all the different uses that lead to 'so many of the quirky bugs' Well, I'm not suggesting we immediately write a test case for every possible function. However, the functions of major components like SIP and IAX don't change very much. So, automated test cases to transfer calls, put calls on hold, etc. could be static from version to version. We might had a chance in hell if we wrote a test case before any functions were written Test cases for modules can be written by module authors (who know what to test for anyway). Ideally, having test cases would be requisite for a module's inclusion in the main Asterisk distribution. back porting test cases to what is it 80K lines an growing is a project the size of asterisk What would we be back porting? I'm talking about starting now. please tell talk up if you know some trick here :) Yes, start slowly and move methodically. Software products twice Asterisk's size have test plans. And anything methodical is better than haphazard the GIHTW* method currently in use. Nick *Gee, I hope this works! ___ 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 Sunday News: Off track with 1.0, moving forward
Olle E. Johansson wrote: Maybe Adam or Steve can add some scripting capability to their IAX/SIP clients so we can use them for testing :-) Having a few of these somewhere would be a good start: http://ameritec.com/fcm/index5.html Nick ___ 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] FXO answering quicker
Andrew Yager wrote: Hi, I don't know if this is possible - but can I set up asterisk to answer the FSO line after one or two rings rather than four? I haven't (yet) found a configuration variable to let me do this... Do you have: 1. Caller*ID turned on in zapata.conf? 2. Wait() before your answer()s in extensions.conf? The combination will cause pickup to take 4 rings. Nick ___ 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] FXO answering quicker
Andrew Yager wrote: Hi, Thanks for those tips. I have now removed both of those things - and I'm down to two and a half rings till answer. In my asterisk console I get the following messages: Jun 7 09:04:28 NOTICE[-1244730448]: chan_zap.c:4800 ss_thread: Got event 2 (Ring/Answered)... Jun 7 09:04:31 NOTICE[-1244730448]: chan_zap.c:4800 ss_thread: Got event 2 (Ring/Answered)... Jun 7 09:04:31 NOTICE[-1244730448]: chan_zap.c:4800 ss_thread: Got event 2 (Ring/Answered)... Jun 7 09:04:34 NOTICE[-1244730448]: chan_zap.c:4800 ss_thread: Got event 2 (Ring/Answered)... as it rings - I'd ideally like a single ring and then answer. Is that possible to do? Have you tried updating the zaptel drivers? Nick ___ 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] Feedback needed! FindMe/FollowMe Feature Spec.
Brian D'Arcy wrote: Callee answers, app_findme says: There is a call for you from (CIDNum), to accept this call, press *, otherwise press #, or hangup. If I press *, the caller hears, I have found this person, connecting you now.. Just one suggestion: make it # to accept and * to hang up, in order to keep in consistent with queues, which use that behavior. Nick ___ 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] Beep Sound
Philipp von Klitzing wrote: Hi! Does anyone have a more clear beep tone for the voicemail? Try Playtones(): http://www.voip-info.org/wiki-Asterisk+cmd+Playtones Playing the beep gsm, as far as I can tell, is hardcoded into app_voicemail. So, the options are either to replace the gsm or edit the C code to use ast_playtones_start(). Nick ___ 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] VOIP Spam
Duane wrote: Tom Green wrote: Brian, Encrypted SIP messages can be sent using TLS. However, I don't think it is realistic to expect everyone calling you to have a public/private key pair. I don't quite agree. SMTP servers that support SMTP-TLS and have valid certs + config do exactly that already... But I think Tom's point is that SMTP-TLS is not very common. However, a PKI for VoIP would be much easier, and much more manageable, than PKI for email. Each provider would have to maintain a key server that stored keys for their users. Then, a public, central registry of provider keys would be needed. The main challenge would be getting private keys into phones. Alice --- Alice's Provider (AP Co.) - Bob's Provider (BP Co.) Bob [Signed by Alice] [Alice's Verified Sig][Alice's Verified Sig] [Signed by AP Co.] [AP Co.'s Verified Sig] [Signed by BP Co.] In this system, Alice would sign and send her SIP messages to her provider's SIP proxy. Her provider, AP Co., proxy would verify the signature with its own key server, and, if valid, would sign it with the AP Co, key and pass it on to BP Co.'s proxy server. The BP Co. proxy could then check AP Co.'s signature, sign the message, and pass it to Bob. Bob, then, must only check that the message is signed by the user's provider. There are, of course, weaknesses in this plan. To name a few: 1. It's a chain of trust: it's hard for Bob to verify Alice's signature directly -Not impossible to fix 2. A central registry must be created that's free and open for providers to use but secure enough to verify members. -Think about the global IP address distribution agencies 3. Phones must get private keys securely. Nick ___ 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] VOIP Spam
Duane wrote: Nicholas Bachmann wrote: 1. It's a chain of trust: it's hard for Bob to verify Alice's signature directly -Not impossible to fix CAcert.org's whole purpose is cheap, easily obtainable security... It employs a web of trust in the website frame work to build up and distribute face to face identification checks... A web of trust is different from the chain of trust I'm talking about. In a web of trust, a key is signed by lots of different people; ideally, everybody can trust everybody. In a chain of trust, each member only knows and trusts the adjacent members. 2. A central registry must be created that's free and open for providers to use but secure enough to verify members. Again CAcert.org fulfils this criteria... Sort of... CAcert.org is a Certificate Authority. A CA just signs public keys, while a key server stores a copy of them. What I'm talking about is more like http://pgp.mit.edu/. -Think about the global IP address distribution agencies 3. Phones must get private keys securely. Last one is as much a technical issue as a people issue, although PIX firewalls implement (forget the acronym) where they send a request to a CA and the CA sends back a certificate, I keep meaning to implement it for CAcert but I lack a PIX for dev testing... But we're not looking at certificates; we're looking at public/private keypairs. Phones can generated the keypairs, but how does the phone prove to the key server that it is an authorized phone? With just a simple password? Nick ___ 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] E1 Red Alarm
Konrad Gorski wrote: maybe CRC problem? try: span=1,1,0,ccs,hdb3,crc4 No, the provider told us no CRC (and I checked anyway, they weren't kidding). Nick ___ 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] SIP - Receptionist
[EMAIL PROTECTED] wrote: Hi All! I am thinking about fork-lift-upgrading a Nortel-Meridian key system with a * PBX driving SIP phones in the office. The interface to PSTN would be a fractional T1 PRI (11 lines plus D channel). The GS phones look acceptable for most users. The forthcoming Sayson 480i would work for management types. The receptionist, however, is currently used seeing a backlit display - with buttons - attached to her phone - showing all the extensions in the office, and who's has a conversation going etc. We believe that autoattendant should only be used after hours ;). Question: How do I drive - acquire such panels with asterisk? What are they called? This much I can answer: a Digital Station Selector (DSS) is what you're talking about. who makes em? I have seen Monastery, but that may be too cumbersome an interface for the relatively high call volume. I hope I explained what I am looking for. As far as I know, you're on your own. I've thought about a Java applet that acted as a DSS... it would be really simple w/ the manager interface. Look at astman (and the Wiki) for details. Nick ___ 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] E1 Red Alarm
Anton Tinchev wrote: Check the crossover cable. I did, by making a connection between spans 1 and 2 on the Digium card. That span worked fine. Nick Nicholas Bachmann wrote: Howdy - I'm trying to get a Malaysian PRI E1 up on a TE410P, with no luck. Right now, the setup is Telco - HDSL - WorldDSL UTU801- 2 BNC E1 - balun - crossover - TE410P Right now, the CSU/DSU-ish WorldDSL box has a green light indicating E1 sync, but the TE410P shows a red alarm. I checked the card by plugging the crossover from port 1 to port 2 on the 410 (it worked fine). It I change any of the cabling (i.e. swap things around), the green light goes off. I have my suspicions about the balun (http://www.ctcu.com/catalog/datacom/balun.pdf). Would a DB15F-RJ45 converter be better the the BNC-balun-RJ45 arrangement we have now? Here's my zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 The telco line IS working; it was tested and put in a couple of days ago. Any ideas why this isn't working? Nick ___ 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] newbie
Andrew McRory wrote: I can offer some links that helped me... [...] If anyone has other links I'd appreciate them! Don't forget http://www.asteriskdocs.org/ ! Nick ___ 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] E1 Red Alarm
Howdy - I'm trying to get a Malaysian PRI E1 up on a TE410P, with no luck. Right now, the setup is Telco - HDSL - WorldDSL UTU801- 2 BNC E1 - balun - crossover - TE410P Right now, the CSU/DSU-ish WorldDSL box has a green light indicating E1 sync, but the TE410P shows a red alarm. I checked the card by plugging the crossover from port 1 to port 2 on the 410 (it worked fine). It I change any of the cabling (i.e. swap things around), the green light goes off. I have my suspicions about the balun (http://www.ctcu.com/catalog/datacom/balun.pdf). Would a DB15F-RJ45 converter be better the the BNC-balun-RJ45 arrangement we have now? Here's my zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 The telco line IS working; it was tested and put in a couple of days ago. Any ideas why this isn't working? Nick ___ 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] RE:Asterisk PABX switch
[Please don't top post] Nikolay Koev wrote: Hi, Nick I believe * can connect PABX through VoIP, but my question is Whether it can switch calls between PABXs directly, within the TE405P, without conversion to IP. And on the other hand, all PABX Yes, as long as that's how your dial plan is set up. To be able to make calls to the analogue PBX through VoIP. All E1 lines are distant SDH, and VoIP to cisco is Ethernet distant too. ___ 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] Hotel wake-up
Bob Knight wrote: Nicholas Bachmann wrote: Bill Michaelson wrote: Anybody know how to implement a hotel wake-up call feature with *? It seems like it could be accomplished with an AGI and a script that wrote call files. Have the AGI prompt for the wakeup time (or have a web interface for a front-desk person do it) and write a file to a directory indicating when the wakeup call should occur. Then, have a Perl script that goes through those files and generates a call file in /var/spool/asterisk/outgoing at the right time. Call files make retries simple as well, allowing you to space them and choose how many you want. If you wanted to get fancy, you could use a database (perhaps with triggers?), voice recognition, or mp3s for the user to wake up to. Good old at job may be able to help with this (man at). I thought about cron, but not about at, since I usually turn atd off on servers, but you're right, it would great here: [EMAIL PROTECTED] root]# echo wakeup 1234 | at 6:30 or in Perl open(AT, |at 6:30) or die $!; print AT wakeup 1234; close( AT); Nick ___ 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] Hotel wake-up
Rob Fugina wrote: On Sun, Feb 29, 2004 at 06:35:54PM -0500, Matthew B Marlowe wrote: I haven't figured out yet how to make * wait until the call in answered before playing a recording (without the recipient pressing #). Show application dial. Use option A Unfortunately, there doesn't seem to be anywhere to put options such as that in a call file. I've tried several things, but haven't found the majick yet... How about an EAGI that detects hellos, or background noise? ___ 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 PABX switch
Nikolay Koev wrote: I wonder if the next is possible with *: PABX | E1 | PABX E1- Asterisk E1PABX | \ E1\ | IP PABX \ Cisco 827V Analogue PBX Yes, this is possible to do, assuming your other IP PBX supports on of the VoIP protocols * does. You'll also need a TE405P or a TE410P for the E1 interface. If possible, how much power the CPU must have? Since you'll be doing encoding and decoding on a bunch of channels, you'll want a farly beefy setup. Nick ___ 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] Hotel wake-up
Bill Michaelson wrote: Anybody know how to implement a hotel wake-up call feature with *? It seems like it could be accomplished with an AGI and a script that wrote call files. Have the AGI prompt for the wakeup time (or have a web interface for a front-desk person do it) and write a file to a directory indicating when the wakeup call should occur. Then, have a Perl script that goes through those files and generates a call file in /var/spool/asterisk/outgoing at the right time. Call files make retries simple as well, allowing you to space them and choose how many you want. If you wanted to get fancy, you could use a database (perhaps with triggers?), voice recognition, or mp3s for the user to wake up to. PlugIf this sounds too complicated, email me off list; I could write this very inexpensively for you./Plug Nick ___ 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
