RE: [Asterisk-Users] Re: How far is IAX to be a Standard
[snip] > A few weeks ago there was a statement from an open source > guru at an event making the mainstream news all over the > world and the statement was this: "Watch out for Asterisk, it will be > bigger than Linux". > > This wasn't coming from an Asterisk zealot driven by wishful > thinking. It was a guy with enough clout to make top > headlines in the mainstream media. You should not quote "an open-source guru" or "a guy". It undermines your credibility. Name the person you are quoting (Jon 'Maddog' Hall, President, Linux International), and provide links to prove your claim. Here, let me help. http://news.zdnet.co.uk/communications/networks/0,39020345,39169076,00.h tm ___ 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: How far is IAX to be a Standard
[EMAIL PROTECTED] wrote: >> -Original Message- >> From: [EMAIL PROTECTED] [mailto:asterisk-users- >> [EMAIL PROTECTED] On > Behalf Of Steve Totaro >> Sent: Monday, November 01, 2004 12:53 PM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: Re: [Asterisk-Users] Re: How far is IAX to be a Standard >> >> No worries here. What works best will win out eventually. > > Not old enough to remember this, but wasn't Beta better than VHS? Interesting point, but then we'd need to define "better". Technically, Beta is considered to be superior. So much so that it became very successful in the professional media industry. VHS went consumer, Beta went pro. The failure of Betamax in the consumer market is probably due to the VHS consortium offering more choice to the consumer than Sony did with Beta. So VHS was "better" by virtue of 1) lower price; 2) wider selection of For IAX, success will be based on several factors: Technical superiority Price (i.e. how much does it cost to configure and maintain) Compatibility (how many products support it). ___ 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] Linux and Windows
> Jay Milk wrote: > > I'm not sure why I'm even discussing the benefits of one operating > > system over another with someone who can't even spell its name. > > > > Um, because he's an intelligent and insightful commentatator on the > various aspects of VoIP in general and Asterisk in particular? Possibly. But it doesn't matter how much knowledge or insight someone has, when, with a few targeted words, he insults 1/3 of the mailing list by implying that they're lazy gamers who never work. I was dealing with such a person a few months ago -- brilliant programmer, but a complete primadonna. While he made great progress on the work assigned him, he disrupted the team and undermined our process. Took me two sr. programmers to replace him, but the productivity went through the roof once he was permanently reassigned to "anywhere-but-here". > People are still reading this thread because beneath the > smoke there is > some technical content worth reading. I haven't seen much technical content here lately. Mostly insults. > Stow the ad hominem stuff and keep your reputation as a good guy. Agreed... I don't want to step on anyone's toes. Just don't like having mine stomped on and tend to kick back. ___ 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] queue_log analyzer
Hello, I'm glad you use XC-AST. What you say might be interesting, something like an hourly call center capacity usage that might as well tell the agent usage per penalty level. In the next versions, we will anyway add a realtime queue monitoring mechanism, so you can leave a browser open and see what happens in realtime. l. In data Mon, 1 Nov 2004 20:08:08 -0800, Jeremy Rusnak <[EMAIL PROTECTED]> ha scritto: Hi, Thanks for the great software. I use this almost daily to monitor our call queues. We have a relatively small queue, with three support agents. One thing that I would love to be able to see in the reports (if it is possible) is to tell when more than one agent is active. We have three agents, as I said, but it is rare that all three of them are on the phone at once. It would be nice to see that x% of calls overlap, for example. This would let me schedule my staff better. Jeremy On Fri, 29 Oct 2004 21:23:54 +0200, lenz <[EMAIL PROTECTED]> wrote: Hello list, I'd like you to know that version 0.3.5 of XC-AST is out - now it is all translated into English and has a 20 page user manual, so I guess it's a bit more user friendly. See http://demo.xcept.it/xc-ast Plans for the future include a real time queue monitoring feature; I was wondering whether to use the Asterisk administrative interface or what, because a number of people seem to notice that using the administrative interface makes * a little bit more crash prone than otherwise, at least in busy environments. Anybody has experience with this sort of things? Also, I was looking at this message: http://lists.digium.com/pipermail/asterisk-users/2003-July/014965.html Xc-Ast now implements most of this stuff - apart from single agente based reports, but they will be implemented soon; I was wondering if there is some way in app_queue to signal that an agent is in wrap-up mode apart from setting a fixed wrap-up time in queues.conf. Thanks for any help and idea. l. -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ 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 -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ 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] Linux and Windows
I never claimed to be unbiased. I said there are many valid reasons for choosing either Windows or Linux as an operating system. For me, one of the valid reasons to have Windows machines around is to pay the bills. When consulting, I still recommend Windows for most clients, because the software they need is available. I also routinely recommend Linux servers for those shops which have the required expertise, and pass those clients on to my "Linux guys". I strive to make the best recommendations based on the clients needs and abilities, and not based upon a personal bias, or a fanatical hate for one operating system (as displayed by some here). I hope this makes my position clearer. > -Original Message- > From: Matt Riddell [mailto:[EMAIL PROTECTED] > Sent: Monday, November 01, 2004 5:51 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Linux and Windows > > > Jay Milk wrote: > > > I make six figures from Windows. I have yet to earn a penny with > > Linux. > > That hardly makes your opinion unbiased. You are obviously a windows > shop or you suggest windows. There are many of us who make our money > mainly from Linux. ___ 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] Linux and Windows
Jay Milk wrote: I'm not sure why I'm even discussing the benefits of one operating system over another with someone who can't even spell its name. Um, because he's an intelligent and insightful commentatator on the various aspects of VoIP in general and Asterisk in particular? People are still reading this thread because beneath the smoke there is some technical content worth reading. Stow the ad hominem stuff and keep your reputation as a good guy. B. ___ 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] Linux and Windows
I can't speak for the US Govt, but I speak for a local government. We use open source everywhere. My departments PBX is Asterisk, Fileservers, Webservers, we use Linux everywhere. In my dealings with the State of TX they are adopting open source for some very mission critical applications. If you wonder about opensource and Govt go read GCN. They talk all about it. There's a place for both Windows and opensource. If you can't do both, or work around either one on either platform you are too narrow minded. I sit here sending you this e-mail on my laptop running Windows XP, through my Exchange server running Windows 2000, that goes to my Linux mail gateway running Postfix to relay the mail outbound. Be more open minded, Windows isn't going away, and neither is open source software. Learn to deal with it. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Karl J. Vesterling Sent: Monday, November 01, 2004 7:46 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Linux and Windows At 06:51 PM 11/1/2004, you wrote: [snip for brevity[ So the U.S. Govt has never used linux anywhere? Wow. Not in most installations, and definitely not in DoD facilities. The "Office of Inspector General" has deemed open source to be "Verboten". That's going to become an interesting situation when Solaris goes open source... http://www.eweek.com/article2/0,1759,1647198,00.asp Question: Why isn't there a commercial solution available in some cases? Answer: What company in their right mind would engineer a competing product to a solution that costs $0.00 ??? Again making the mistake that open source equates non-commercial. Once again... The Office of Inspector General has deemed (any and all) open-source to be forbidden. Whether it be commercial of non-commercial open-source software it's forbidden. Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Yahoo Messenger: karl_vesterling ICQ: 1548052 AOL Instant Messenger: n2vqm Telephone: Washington DC: (202) 448-3009 Extension 0 Annapolis MD: (240) 524-6706 Extension 0 Seattle WA: (360) 516-1822 Extension 0 Niagara Falls NY: (716) 286-9175 Extension 0 Buffalo NY: (716) 608-1121 Extension 0 United Kingdom: 0870 3403428 Extension 0 ___ 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] Linux and Windows
I'm not sure why I'm even discussing the benefits of one operating system over another with someone who can't even spell its name. > -Original Message- > From: Benjamin on Asterisk Mailing Lists > [mailto:[EMAIL PROTECTED] > Sent: Monday, November 01, 2004 5:29 PM > To: Jay Milk > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Linux and Windows > > This has nothing to do with anger, but it has all to do with > truthfulness. > > How many times have you had discussions with Windoze folks > trying to give you good reasongs why Windoze should have its > place when in reality it only came down to the one thing that > they would not admit: all they really cared about is the > ability to play all those games. Interesting... In hundreds of consulting gigs, gaming never once has been a factor. How about that truthfulness? No really... Is gaming the reason why YOU keep a windows machine around (?), because not one of my machines has seen a game in at least five years nor do I know anyone who uses their computer for gaming. The handful of friends who like gaming use their xboxes and nintendos and segas (are those still around even?) > So, I say, if you want to play games and that is what drives > your choice, then say so and don't come up with all these > pretense arguments. But I also say that playing games is not > an activity that one should expect to get paid for. See what I was saying about those insults? Maybe you don't mean to be insulting, but maybe there's a language barrier or some general inability to understand the written word, and that's why you don't grasp the concept of need-driven choices? Many users have a valid NEED for Windows servers and clients... Some applications are only available on certain platforms, so that is what drives your choice. Just as graphic designers chose the Mac platform for a long time because certain applications were only available on the Mac. > > don't like it, don't use it, > > In fact I don't. > > > but don't insult the people who do. > > Who is the one making is the insult here? The one with the > pretense argument or the one who says one shouldn't expect to > get paid for playing games? > > > accept that there are valid uses > > Vaild uses, sure. Question is whether that includes the > workplace. I believe it doesn't. Persistent little troll, aren't you? If you are saying that nobody who's using Windows in the workplace is actually "working", then I'd have to conclude that anyone who defends an operating system as ruthlessly and impolitely as you do, does probably not have much of a life outside of computers. But maybe we both should get back to WORK now. Oh no wait, YOU have to go back to work, I have to go back to playing games on my windows computers, once I finish the work on my linux machines. There are names for people like you, but I don't think they're appropriate for this list, and we'd probably have to explain their etymology to some of our international visitors. ___ 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] Hold music while ringing
What im after is a dial plan, so when a user calls into a 'specific' number, instead of hte meharing hte ringing until I pickup the call on my SIP phone. Tried looking thru voip-info but the clsoest i could find was a WaitUserOnHold in teh dialplan, not sure if this is what im after nor how to implement it Thanks in Advanced Matthew * > Me ___ 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] Eezee phone?
Hi This phone is based on the PA1688 chip. The rumour is that IAX2 support will be available by December...we will have to wait and see. Regards Clive On 1 Nov 2004 at 13:16, Kanuri, Seshu (Company IT) wrote: > > The link refers to an expired auction. It is no longer listed as having > IAX2. That claim was withdrawn till IAX2 on it is stabilized by the Chip > manufacturer. > > > NOTICE: If received in error, please destroy and notify sender. Sender does not > waive confidentiality or privilege, and use is prohibited. > > ___ > 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] soxmix?
Good day all
I cant get asterisk to join 2 recorded files with Monitor() and sox
I have a asterisk version 0.9.1 and a zaptel card
I installed sox 12.17.6 and copyed soxmix to /bin
I added these in my extensions.conf
exten => _0.,1,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten => _0.,2,Monitor(wav,${CALLFILENAME},m)
exten => _0.,3,Dial(Zap/g2/${EXTEN:1})
exten => _0.,4,Congestion
Thanks Seth
but after a call ther in extension-time|m-in.wav and
extension-time|m-out.wav
Please help
Altus
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Re: [Asterisk-Users] Re: How far is IAX to be a Standard
Benjamin on Asterisk Mailing Lists wrote: If you refer to the urban legend that IAX always needs a server to stay in the media path, then you would be wrong. IAX has a mechanism that for all practical purposes is equivalent to a SIP reinvite through which the end points then transition to a mode by which they communicate directly peer to peer. With SIP and reinvites the server still stays in the signaling path (but not the audio path). This makes CDRs accurate. With IAX there is no way for the server to have correct CDRs because it never knows when the call has ended. As far as an IAX server is concerned the call ended when the transfer happened. ___ 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] Amount of time asterisk take to pickup incoming call on ZAP interface
On Tue, 2 Nov 2004 14:25:49 +1100, Sophus <[EMAIL PROTECTED]> wrote: > I set callerid=no in zapata.conf > > and > >exten => s,1,Wait(0) >exten => s,2,Answer > > in extensions.conf > > but there is still a delay of about 5-8 seconds before asterisk picks up. > any ideas? changes in zapata.conf may require you to stop start Asterisk or even unload and reload the Zaptel driver. Also, I don't think you need the "Wait(0)" line in extensions.conf. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] queue_log analyzer
Hi, Thanks for the great software. I use this almost daily to monitor our call queues. We have a relatively small queue, with three support agents. One thing that I would love to be able to see in the reports (if it is possible) is to tell when more than one agent is active. We have three agents, as I said, but it is rare that all three of them are on the phone at once. It would be nice to see that x% of calls overlap, for example. This would let me schedule my staff better. Jeremy On Fri, 29 Oct 2004 21:23:54 +0200, lenz <[EMAIL PROTECTED]> wrote: > > Hello list, > I'd like you to know that version 0.3.5 of XC-AST is out - now it is all > translated into English and has a 20 page user manual, so I guess it's a > bit more user friendly. See http://demo.xcept.it/xc-ast > > Plans for the future include a real time queue monitoring feature; I was > wondering whether to use the Asterisk administrative interface or what, > because a number of people seem to notice that using the administrative > interface makes * a little bit more crash prone than otherwise, at least > in busy environments. Anybody has experience with this sort of things? > > Also, I was looking at this message: > http://lists.digium.com/pipermail/asterisk-users/2003-July/014965.html > Xc-Ast now implements most of this stuff - apart from single agente based > reports, but they will be implemented soon; I was wondering if there is > some way in app_queue to signal that an agent is in wrap-up mode apart > from setting a fixed wrap-up time in queues.conf. > > Thanks for any help and idea. > l. > > -- > Creato con M2, il rivoluzionario client e-mail di Opera: > http://www.opera.com/m2/ > ___ > 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] Queue
Mark Halverson wrote: Regarding a call queue - I was told that in order to have a queue users would have to login to the asterisk queue on their extension. What you probably mean is that the agents would log in (via AgentLogin or AgentCallbackLogin application). If using AgentCallbackLogin, * would prompt for the extension # to ring them back when there's a caller. They would hear music on hold and then the music would stop and a caller is instantly there without any warning. Not true. in queues.conf there is a setting called "announce" where you can define which audio file to play. If you don't define anything then, yes the two legs would be bridged without "any warning". This setup doesn't work for me. We have multiple employees that telecommute and I need a queue or setup to do the following: Call comes in and rings all the extensions. specify "strategy = ringall" in queues.conf Once all extensions are in use the incoming caller(s) are placed on hold and hear music on hold. enabled by default application behavior. When an extension becomes available, the phone rings. Sounds simple to me It sure is. You can look at http://www.voip-info.org/wiki-Asterisk+call+queues for more info and as a starting point. more links on that page to info about agents and queues.conf settings. Flynn ___ 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] Queue
Mark, This is all very doable. Check out the following page as a starting point: http://www.voip-info.org/tiki-index.php?page=Asterisk%20call%20queues Specifically, you want to look at AgentCallbackLogin, which calls Agents back instead of having them wait on hold. We use roundrobin distribution for our staff instead of Ringall...The issue is that with ringall, it just gets confusing with three people trying to pick up the call "first." Round robin will try each person in a rotating fashion until someone picks up or the queue times out. Jeremy On Mon, 1 Nov 2004 19:44:59 -0800, Mark Halverson <[EMAIL PROTECTED]> wrote: > Regarding a call queue - > > I was told that in order to have a queue users would have to login to the > asterisk queue on their extension. > > They would hear music on hold and then the music would stop and a caller is > instantly there without any warning. > > This setup doesn't work for me. > > We have multiple employees that telecommute and I need a queue or setup to > do the following: > > Call comes in and rings all the extensions. > > Once all extensions are in use the incoming caller(s) are placed on hold and > hear music on hold. > > When an extension becomes available, the phone rings. Sounds simple to > me > > -Mark > > ___ > 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] Queue
Regarding a call queue - I was told that in order to have a queue users would have to login to the asterisk queue on their extension. They would hear music on hold and then the music would stop and a caller is instantly there without any warning. This setup doesn't work for me. We have multiple employees that telecommute and I need a queue or setup to do the following: Call comes in and rings all the extensions. Once all extensions are in use the incoming caller(s) are placed on hold and hear music on hold. When an extension becomes available, the phone rings. Sounds simple to me -Mark ___ 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] List Issue
As I see it we are all a bit better and a bit worse on various subjects. But one thing most of us have in common is being basically decent people trying to help. Very seldom do we get some fool who's trying to undermine or sabotage our efforts. One thing that helps keeping a list friendly is to keep demeaning comments off-list, and only after several failures do so publicly. Even then it's seldom of any value, as few others usually even want to read it. 97.5% of the time it's only interesting to two people. Being that the size of our list has so many different people with various backgrounds and levels of understanding, not only of Asterisk but also of English, we are bound to run into things that looks, well... bad. It's very easy to read into things the wrong meaning when you don't know the spirit it said in. When I see various things like html or mispostings I always try to handle it offline. (Not that I have not made mistakes.) But it takes the edge of the "blade" and keeps it a more friendly place. If you feel a guy is full of it tell him offline. And if you ask what he meant in a way that is non confrontational he's more likely to answer properly and realizing the errors of his way, apologize. When a service company makes a mistake with my account, I call them up and make them my ally against the problem that occurred. Makes them willing to solve it. I might even say (in a calm manner) that I'm very unhappy, and add that I know You did not do it, but I want you to know that I'm unhappy. Amazing how far you can get with some elbow grease. (No crude comments here : ) On the all, I think this is a great list with some very competent and willing people. People who's making the growth of Asterisk possible through their valiant efforts. Keep up the good work! -- Steve Szmidt "There's always two sides to any dispute, your job is to fully understand them." ___ 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] Centrex
> Em Seg 01 Nov 2004 19:18, Tim Sailer escreveu: > > On Mon, Nov 01, 2004 at 07:13:07PM -0200, Marcelo Pacheco wrote: > > > If you plug a plain analog phone to the line, then you need an FXO card > > > in it's place. > > > > Yup. It works with a plain ol' phone, Fax, and modem, but you need to > > dial 9 to get out. Just a generic install of Asterisk and having the > > FXO card configured, asterisk never sees the line ring. > > > > > Centrex usually means that you have hired a virtual PBX system. That in > > > itself doesn't mean FXS/FXO/T1/E1, a Centrex could be shipped to you in > > > many diferent ways. > > > > sigh. I hate phone companies. > > I have the same issue you have, except that the issue is with a real > (Brazilian) PBX. I have 2 american phones (I live in Brazil) that won't > detect a ring from that PBX, but if I put another phone in parallel, so I can > hear the ring, I can pick up that phone and answer, everything else works ok. > However both a TDM400P FXO module and an MD3200 FXO work ok with that PBX. It use to be very common for US phone ringers to be tuned to some specific frequency (eg, 20hz, 30hz,... 60hz) for use in party lines. Ringing voltage going to party 1 might use 30hz while ringing voltage to party 2 might be 50 hz. Might you have a phone that has one of those ringers? ___ 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] MOH whilst waiting for Conference attendees
Is this possible? I'd like for the members of a conference to hear the hold music until such times as the host of the conference has joined. -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ ___ 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] Amount of time asterisk take to pickup incoming call on ZAP interface
Hi, I set callerid=no in zapata.conf and exten => s,1,Wait(0) exten => s,2,Answer in extensions.conf but there is still a delay of about 5-8 seconds before asterisk picks up. any ideas? cheers adam ___ 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] tdm410 driver prevents files being played
I have a brand new HP Proliant DL380 with dual Xeon 3.06 processors, 2gb RAM running RHEL. It has a TDM410P installed. Asterisk works fine without the zaptel drivers loaded. In this state the red lights on the card flash quite happily. There is no T1 installed into the card yet. When I run the zaptel driver for the tdm4xx the lights go out and then any audio file (eg VM) that * wants to play stalls. The CLI says that it is playing the file but nothing is heard and the file never terminates. Hanging up seems to do what is it meant to do. Any ideas? -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ ___ 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: How far is IAX to be a Standard
On Mon, 01 Nov 2004 20:55:40 -0500, Karl Brose <[EMAIL PROTECTED]> wrote: > And what to you base such an assertion on? > Would you care to elaborate on the technical justifications? In a nutshell: The future of voice will be peer-to-peer and that's where IAX has a clear edge. As for a technical discussion of the benefits of IAX, there is quite a bit of material available in the list archives and on the Wiki (search for SIP versus IAX). > standards are not declared by publication or committee votes, but by > the numbers of implementations. precisely my point. > The activity surrounding IAX and > Asterisk at present is encouraging and stimulating, no doubt. indeed. > But there is other competition as well, BT is implementing not SIP, > but MGCP in their transition to an all voip-based network. Is this an argument that bolsters SIP? I would think not. I would think it is a sign that SIP isn't all that omni-potent as SIP lobbyists would have us believe. > Currently, I don't see how IAX can achieve the kind of flexibility > and versatility of SIP, but let the market decide. That may well be the difference between us, you seem to look at things as they are *curently*, I look at the *potential* of things and how they might be in the changed landscape of the future. > Your zealous attitude does little to promote qualified technological > exchange on this list or elsewhere, it rather reminds us of ideological > battles between close-minded parties engaged in power struggles. It's all about balance. While some play the part of a zealous nay-sayer, I may be forgiven to play the role of the zealous supporter. What you call zealous here is of course subject to whether or not you want to take things out of context and how you interpret them: You may think that horse carriages are automatically a bad thing and motorcars are automatically a good thing. The truth is though that this depends on viewpoint. When I say SIP is like a horse carriage, then this reflects the fact that it is a design for the world we have come from, not necessarily the world we are going to. It doesn't mean that horse carriages were a bad thing. After all, motorcars have also brought us noise and pollution. When I say IAX is like a motorcar, then this reflects the fact that it is a design for a modern world. Mind you, driving a motorcar in the days of a non-motorised world was a rather impractical activity. No gas station network, no garages to get things fixed, bad roads. A horse carriage was far more suited to that world. However, that world did change, for good or for worse. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] Directory app and extension
Message: 5 Date: Mon, 1 Nov 2004 16:09:09 -0600 From: Brian Roy <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Directory app and extension To: Asterisk Users Mailing List - Non-Commercial Discussion <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=US-ASCII On Mon, 01 Nov 2004 16:21:46 -0500, David Filion <[EMAIL PROTECTED]> wrote: So, the question is does anyone know of a way to get the extension number when the dial plan context is entered via Directory(), and if so how? David Filion David, You could always right your own directory application (AGI if you want) that reads from Postgres. It would be pretty easy. -Chuji Hi, The problem isn't reading from postgresql. In fact there is app_postgres which work pretty well. The problem is the Directory app doesn't provide appear to provice access to the extension that it has "found", so when it enters the specified context in extensions.conf, we can't refer to the "incoming" extension. We use the extension to do our db look ups, so with out it we're stuck. :( David ___ 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: How far is IAX to be a Standard
On Nov 1, 2004, at 9:37 PM, Benjamin on Asterisk Mailing Lists wrote: [EMAIL PROTECTED] wrote: IAX really isn't the 'one and only' perfect signaling protocol IAX is *not* a signalling protocol. It is a VoIP protocol. And that's the whole point. H.323, SIP, et al those are all signalling protocols, half protocols so to speak. IAX is a self-contained, true internet protocol. In a world where everything is moving towards peer to peer that is an advantage. Dedicated protocols for signalling-only have been designed for large telcos and yesterday's telco infrastructure, not for the future of voice on the net peer to peer style. Think back, The telco started with inline signaling. Pulsing digits, cross bar switching, DTMF, CAS 'Yesterdays telco' took signaling out of band for a reason. I'm not sure putting the signaling back into the bearer channel is a 'good thing'. -- Matthew S. Crocker Crocker Telecommunications, LLC Vice President PO BOX 730 Greenfield, MA 01302-0710 P: 413-746-2760 F: 413-746-3704 W: http://www.crockertelecom.com E: [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
Re: [Asterisk-Users] Re: How far is IAX to be a Standard
This thread was started by Randy Bush, thought that name rang a bell. Good conversation nonetheless. http://lists.digium.com/pipermail/asterisk-users/2004-July/053278.html - Original Message - From: "Benjamin on Asterisk Mailing Lists" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Monday, November 01, 2004 9:17 PM Subject: Re: [Asterisk-Users] Re: How far is IAX to be a Standard >Try travelling in Africa, the Middle East and South Asia, where >everything is still mostly dialup and many of the phone wires >installed go back before the time when plastic was invented. So SIP doesn't work on dialup? That's funny 'cause I'm using it like that... That's not what I said. I said "Try travelling in Africa, the Middle East and South Asia" and that is precisely what I meant. I have been doing a lot of travelling in those places and in many situations and places the only thing that worked was IAX and ILBC. You may want to search the archives of this list because this has been discussed many times before and I have explained my observations in quite some detail. You will also see that it won't make any sense for you to come back saying that I am too stupid to set up a SIP connection and that this would have to be the only reason why SIP didn't work while IAX did, because I had already explained in those earlier discussions that we had been competing against the big names in the industry, the cream of the cream, who came with their SIP gear and their SIP specialists, and they couldn't get VoIP to work where we could with IAX. If you don't agree that having IAX ratified into a standard would be a help, well, that's where we differ. I have been nagging Mark for some time about getting an IAX RFC initiative under way, and I have offered my help to Frank Miller drafting call flow charts and whatever other limited assistance I could be of in his aim to eventually evolve his IAX specification document to the point where it would be suitable for submission to the IETF. So, we are certainly not in disagreement about the benefit of such an initiative. However, I disagree with the notion that only big industry player backed proposals pushed by plenty of lobbying have a chance of succeeding as standards. Many of the most important internet protocols have all been single inventor designs which became standard through grassroots adoption. I certainly believe that IAX has gained enough momentum to become a major standard without the marketing and lobbying dollars that have by now been thrown behind SIP. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] Re: How far is IAX to be a Standard
[EMAIL PROTECTED] wrote: > IAX really isn't the 'one and only' perfect signaling protocol IAX is *not* a signalling protocol. It is a VoIP protocol. And that's the whole point. H.323, SIP, et al those are all signalling protocols, half protocols so to speak. IAX is a self-contained, true internet protocol. In a world where everything is moving towards peer to peer that is an advantage. Dedicated protocols for signalling-only have been designed for large telcos and yesterday's telco infrastructure, not for the future of voice on the net peer to peer style. > signaling (asterisk in this case) asterisk would have to 'tell' the DSP > chip the signaling packets to embed into the IAX/RTP channel. Do you actually know how IAX works? There is no such thing as IAX/RTP, none whatsoever. Again, that's the whole point of IAX. If you refer to the urban legend that IAX always needs a server to stay in the media path, then you would be wrong. IAX has a mechanism that for all practical purposes is equivalent to a SIP reinvite through which the end points then transition to a mode by which they communicate directly peer to peer. > Will any chipmaker (besides digium) > ever see the need to design such a chip? There is already one chipmaker who thought that IAX was important or competitive advantage enough to embed it into their chip. So, precedence has been set already. This trend can be expected to continue. And, no, Digium had no involvement in this (Mark please correct me if I am wrong). rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] Linux and Windows
There are exceptions to the rule... For instance, if an RFP contained requirements that only an open-source solution, or a solution written from scratch would provide the open source solution may be accepted on the grounds that it is the least cost solution. HOWEVER! One may have to file protest after protest after protest in order to win the award of the contract. For instance... Where I work there is absolutely no open-source allowed on production or development systems. However for security purposes certain open-source solutions are permitted. This is because of the fact that where security is concerned open-source tools are often used to find the vulnerabilities. Hence one must use those very same tools in order to identify the vulnerabilities that a would-be hacker would find. On some production systems Perl is used for various things. This is justified by the fact that it is bundled into Solaris by Sun Microsystems. However in Solaris 8 at least it looks as if the version bundled was configured and built by vandals. All in all, it comes down to the evaluation team for the award of the contract. However the evaluation team will have to justify to the OIG (Office of Inspector General) exactly why open source is justified. That can happen, especially if you force the evaluation team to come to the conclusion that it's easier to justify open source to the OIG than to perpetually refute the protests being filed of their award. Germany on the other hand is a different animal entirely. They've embraced open source, but I think that the U.S. will take a "Let's wait and see." approach. The U.S. Govt is also much larger and tends to move at the speed of a glacier when it comes to adopting "new things" where a risk is involved. They tend to err on the side of caution. At 09:10 PM 11/1/2004, you wrote: Its hard to get to the bottom of this. I've seen things on the internet saying open source in all forms is banned. I've also seem lots of things about deployments in the US government in general and the DoD in particular. I guess like most things the left hand never knows what the right hand is doing. I think the main thing holding back government adoption of free things in most markets is the rather small size of a 5% back-hander on a free solution. Steve Karl J. Vesterling wrote: At 06:51 PM 11/1/2004, you wrote: [snip for brevity[ So the U.S. Govt has never used linux anywhere? Wow. Not in most installations, and definitely not in DoD facilities. The "Office of Inspector General" has deemed open source to be "Verboten". That's going to become an interesting situation when Solaris goes open source... http://www.eweek.com/article2/0,1759,1647198,00.asp *Question:* Why isn't there a commercial solution available in some cases? *Answer:* What company in their right mind would engineer a competing product to a solution that costs $0.00 ??? Again making the mistake that open source equates non-commercial. Once again... The Office of Inspector General has deemed (any and all) open-source to be forbidden. Whether it be commercial of non-commercial open-source software it's forbidden. ___ 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 Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Yahoo Messenger: karl_vesterling ICQ: 1548052 AOL Instant Messenger: n2vqm Telephone: Washington DC: (202) 448-3009 Extension 0 Annapolis MD: (240) 524-6706 Extension 0 Seattle WA: (360) 516-1822 Extension 0 Niagara Falls NY: (716) 286-9175 Extension 0 Buffalo NY: (716) 608-1121 Extension 0 United Kingdom: 0870 3403428 Extension 0 ___ 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] Linux and Windows
At 08:23 PM 11/1/2004, you wrote: Try reading what people say before ranting. He has a *CDMA* modem. Can you offer him good modem support for a CDMA modem? If not, try to avoid making yourself look so foolish. Oh good gawd... I thought he was rambling some obscure sort of part number from some fly by night electronics company. Yeah, you're right... That would cause significant impediments. I had mistaken his claim for one of those WinMoDumbs that use the CPU to do the work of what would ordinarily be handled by the hardware package (Rockwell comes to mind.) Actually, most CDMA modems use AT commands, just like most GSM modems do. I wasn't speaking of the Hayes AT command set... Actually, I expect and demand that if something is called a modem nowadays that it understand the hayes AT command set. There were a few manufactured way back when that had their own command sets. Heck, my ancient TRS-80 Model 100 had a built in modem that knew not of AT-Anything. They both an extended set of the usual AT commands, with special commands to access cellular related features. With the difficulties of not having broadband I can understand why he's stuck with what he's got... Personally, I like my hardware as adaptable as possible. Suggestion: http://www.arcelect.com/StarPoint_Digital_CDMA-1xRTT_Cellular_Modem.htm (or something similar... If your OS doesn't speak RS232c you're probably running Minix) The GSM stuff is openly documented. I think the CDMA guys are a bit more annal retentive, but they also have a spec for the standard CDMA modem interface. All Hail Phil Karn (KA9Q - Pioneer of CDMA) GSM is popular in Europe, but CDMA has better coverage area, hence it's much easier to deploy since you don't need to have as many sites to cover a particular area. Not as many sites = less cost to build the plant. Regards, Steve ___ 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 Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Yahoo Messenger: karl_vesterling ICQ: 1548052 AOL Instant Messenger: n2vqm Telephone: Washington DC: (202) 448-3009 Extension 0 Annapolis MD: (240) 524-6706 Extension 0 Seattle WA: (360) 516-1822 Extension 0 Niagara Falls NY: (716) 286-9175 Extension 0 Buffalo NY: (716) 608-1121 Extension 0 United Kingdom: 0870 3403428 Extension 0 ___ 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] Is there a way to disable call wating?
I think that has to be done on a device by device basis. In come cases the configuration change is done on the client (software/device). If it's a card that goes in the linux box that is running asterisk you can set that in the config file for the device. zapata.conf for Zaptel cards. phondev.conf for QuickNet (PhoneJack / Linejack cards.) If it's an external device, or a PC running a VoIP client, there is usually a configuration option for that. At 08:56 PM 11/1/2004, you wrote: I would like to completely disable call waiting. Does Asterisk have an option for that? Thanks, -- John Lange ___ 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 Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Yahoo Messenger: karl_vesterling ICQ: 1548052 AOL Instant Messenger: n2vqm Telephone: Washington DC: (202) 448-3009 Extension 0 Annapolis MD: (240) 524-6706 Extension 0 Seattle WA: (360) 516-1822 Extension 0 Niagara Falls NY: (716) 286-9175 Extension 0 Buffalo NY: (716) 608-1121 Extension 0 United Kingdom: 0870 3403428 Extension 0 ___ 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: How far is IAX to be a Standard
> >Try travelling in Africa, the Middle East and South Asia, where > >everything is still mostly dialup and many of the phone wires > >installed go back before the time when plastic was invented. > > So SIP doesn't work on dialup? That's funny 'cause I'm using it like that... That's not what I said. I said "Try travelling in Africa, the Middle East and South Asia" and that is precisely what I meant. I have been doing a lot of travelling in those places and in many situations and places the only thing that worked was IAX and ILBC. You may want to search the archives of this list because this has been discussed many times before and I have explained my observations in quite some detail. You will also see that it won't make any sense for you to come back saying that I am too stupid to set up a SIP connection and that this would have to be the only reason why SIP didn't work while IAX did, because I had already explained in those earlier discussions that we had been competing against the big names in the industry, the cream of the cream, who came with their SIP gear and their SIP specialists, and they couldn't get VoIP to work where we could with IAX. > If you don't agree that having IAX ratified into a standard would be a > help, well, that's where we differ. I have been nagging Mark for some time about getting an IAX RFC initiative under way, and I have offered my help to Frank Miller drafting call flow charts and whatever other limited assistance I could be of in his aim to eventually evolve his IAX specification document to the point where it would be suitable for submission to the IETF. So, we are certainly not in disagreement about the benefit of such an initiative. However, I disagree with the notion that only big industry player backed proposals pushed by plenty of lobbying have a chance of succeeding as standards. Many of the most important internet protocols have all been single inventor designs which became standard through grassroots adoption. I certainly believe that IAX has gained enough momentum to become a major standard without the marketing and lobbying dollars that have by now been thrown behind SIP. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] Linux and Windows
Its hard to get to the bottom of this. I've seen things on the internet saying open source in all forms is banned. I've also seem lots of things about deployments in the US government in general and the DoD in particular. I guess like most things the left hand never knows what the right hand is doing. I think the main thing holding back government adoption of free things in most markets is the rather small size of a 5% back-hander on a free solution. Steve Karl J. Vesterling wrote: At 06:51 PM 11/1/2004, you wrote: [snip for brevity[ So the U.S. Govt has never used linux anywhere? Wow. Not in most installations, and definitely not in DoD facilities. The "Office of Inspector General" has deemed open source to be "Verboten". That's going to become an interesting situation when Solaris goes open source... http://www.eweek.com/article2/0,1759,1647198,00.asp *Question:* Why isn't there a commercial solution available in some cases? *Answer:* What company in their right mind would engineer a competing product to a solution that costs $0.00 ??? Again making the mistake that open source equates non-commercial. Once again... The Office of Inspector General has deemed (any and all) open-source to be forbidden. Whether it be commercial of non-commercial open-source software it's forbidden. ___ 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] Is there a way to disable call wating?
On Mon, 2004-11-01 at 19:56 -0600, John Lange wrote: > I would like to completely disable call waiting. > > Does Asterisk have an option for that? While there are many very smart individuals on this list, I don't think there is many mind readers nor hackers who will log into your system to explain to us what interfaces you are using. Next time you ask a question, please provide more details on the outset so you are likely to get a direct and appropriate answer. As for the question, it depends on the interface. For Zap, you have the zapata.conf file. For SIP, you will have to either turn it off on the phone itself or check on call numbers to limit the number of attempts to one at a time. Similar for other VoIP protocols. -- 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
Re: [Asterisk-Users] Queue Prioritization
Jonathan Moore wrote: I have a client that uses two queues for a customer support application. One queue is for english speaking customers. The other is for spanish. This is a small company and they have one dedicated call agent for answering spanish calls. They don't get many spanish calls, so this agent is also part of the english queue. They wish to prioritize spanish queues for this agent over english. Without any manipulation the opposite is happening. They english queue seems to swamp the spanish queue. IE no mater how long a spanish queue member waits if there is an english queue call it will go to the spanish operator. Try this in queues.conf: [english] : : member => Agent/1000 member => Agent/1001 member => Agent/1002,5 ; this is the agent also on the spanish queue [spanish] : : member => Agent/1002 I'm thinking that since agent #1002 has a penalty, * would select him/her last for calls in the english queue, leaving him free for the spanish queue. Check the queues.conf file for more stuff on penalties assigned to agents. Flynn ___ 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: How far is IAX to be a Standard
[EMAIL PROTECTED] wrote: Hello IAX really isn't the 'one and only' perfect signaling protocol because many people forget one thing IAX has one technical issue (by design) which makes it difficult to ever get accepted by the big boys, a real big problem for carriers who have big loads on their systems like we do. With IAX the audio (RTP) and signaling goes embedded over one port. We IAX doesn't use RTP. all know that the big advantage ofcourse is that this makes it an excellent performer behind a NAT, but the big disadvantage is that there is not any DSP chip available in the market which is able to get the codecs encoded and put into this embedded rtp+signaling channel, and I wander if there ever will be because another piece of software does the signaling (asterisk in this case) asterisk would have to 'tell' the DSP chip the signaling packets to embed into the IAX/RTP channel.. That would be a whole new DSP standard, Will any chipmaker (besides digium) ever see the need to design such a chip? This paragraph makes no sense whatsoever. Anyhow, the situation now, is that there is no DSP chip, that means .. Your main processor has to encode the channel in total (3 to 4 E1's absolute is the max possible with dual xeon 3 ghz I read somewhere in this case) Most * implementations today do not use custom DSP chips. They often don't need to. They can if the need arises. Echo cancellation is an area where custom DSP would be a big help, and has seriously been considered. This has nothing to do with IAX, though. Another method is to send the incoming IAX on asterisk out again with SIP to a gateway with hardware DSP's.. (like we do).. This needs less performance ofcourse because asterisk doesn't have to do codec encoding, but nevertheless will still have to transcode to get the signaling and RTP merged and submerged from/to this one IAX port to separate Signaling/RTP ports.. Our setup now is the second scenario.. And my first (rough) calculations are that a dual xeon 3.0 ghz can handle about 500 concurrent channels in this scenario... You are just trying to make use of existing hardware. This says nothing about the merits of the protocols involved. Do you trunk your RTP, or accept a huge data flow? RTP trunking isn't properly standardised, but it is on the way, because RTP overheads make a joke of high performance low bit rate codecs. Ever wandered why there isn't any codec (DSP) hardware availiable for asterisk?? I think here is the answer, because it is very hard to make, Digium should then be able to design a totally new DSP chip design .. And that's much more difficult than to design an E1 board. It isn't hard to make. It is very straightforward. Getting enough volume to justify the development has been the issue. Lots of commercial DSP cards to do the job exist, but they are priced high for a low volume market. Its a chicken and egg problem - expensive cards have a small market, cheap cards would have a possibly large but somewhat unproven market. There are no real technicals issues here, just commercial ones. Our case is that we have about 200 E1's of voip (h323 and sip) traffic and are still expanding. If we would have this all on IAX this would be unmanageable, we would need 50 linux boxes. For that much traffic you obviously have quite a bit of some kind of hardware. What would be the problem if that were 50 cheap, almost disposable, Linux boxes? If you bring those E1s in bundled on a couple of OC-somethings you can't bring them into a pure * system today, but just wait a while and see. This has nothing to do with the IAX protocol, though. Conclusion.. IAX Is a good performer behind NAT and perfect for small setups but to work in an enterprise, Much work has to be done. True. Much work has to be done. However, the real barriers are all commercial. The technical ones are quite small, and most of what you said it not really relevant to the barriers at all. Steve ___ 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] Is there a way to disable call wating?
I would like to completely disable call waiting. Does Asterisk have an option for that? Thanks, -- John Lange ___ 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: How far is IAX to be a Standard
Benjamin on Asterisk Mailing Lists wrote: IAX is so vastly superior to SIP, that the comparison shouldn't be things like VHS versus Betamax, but it should be more like horse carriages versus motorcars. And what to you base such an assertion on? Would you care to elaborate on the technical justifications? It appears you are in the habit of making similar sweeping and unqualified statements regularly. In time you should learn to be more objective. The reality about IAX vs SIP looks much different. SIP quickly replaced H323 in the market of voip endpoint devices and call termination/origination because of its design simplicity, transparency and easy debugging potential because it's text-based and built on existing HTTP protocols. H323 is a complex binary protocol, albeit very powerful, but difficult to implement. Today it is still used in carrier interconnects and transmissions, where it is dominant. SIP, in its foundation, is a state-less protocol alleviating network elements from maintaining state information. It is very light-weight and facilitates call control independent of media transmissions. It's unmatched so far in flexibility and expandability. That is a strength. Because of its simplicity, it has been quickly adopted by device manufacturers who found it easy to expand and adapt to a changing market place. The latter has added a host of new features and added quite a bit of complexity to todays SIP implementations and protocol. The rapid growth of consumer market of high-speed Internet access and the simultaneous lag of (complacency in) implementation of new addressing protocols (IPv6) created the unfortunate proliferation of NAT devices, which truly is the worst thing that happened to voip. The reality today is that SIP has its problems in this environment, but so does H323. IAX and Skype are different. They are peer-to-peer, state-full protocols that integrate control and media transmission in the same data stream and communicate on a single, well-known UDP port. They are therefore more easily routable through NATs. This is not a negligible advantage, and can be a powerful technology enabler at the present time. Other advantages of IAX include increased efficiency in trunk (multiple data stream/call) transmissions, because of its compact design, and perhaps some additional peer-to-peer information exchange. The trunking bandwidth efficiency is only a marginal advantage in situation where consumer NAT devices are encountered, since most such installations don't require the bandwidth on one hand, and the current limitations of consumer cable/DSL bandwidth are no doubt temporary. If bandwidth were such a concern, we wouldn't be running TCP/IP anymore, but had switched to ISO protocols 10 years ago. The binary nature of IAX protocol, while no doubt yielding efficiency, is not a recipe for rapid expansion, it will be harder to match SIP's open architecture and ease of debugging. IAX appears to be a nice complementary protocol in the overall voip world. Because of its integration of call control and media it still has to prove its scalability to large voip systems. Because of its current tight association with Asterisk and its capability to exchange dial plan information between Asterisk systems, it has a clear advantage in the interconnection of Asterisk servers. IAX's potential for being a standard is still to be seen, a first step is a clear documentation and publication of specification that vendors can rely upon to achieve interoperability. In the end, however, standards are not declared by publication or committee votes, but by the numbers of implementations. The activity surrounding IAX and Asterisk at present is encouraging and stimulating, no doubt. But there is other competition as well, BT is implementing not SIP, but MGCP in their transition to an all voip-based network. Such potential momentum in deployment cannot be ignored. Currently, I don't see how IAX can achieve the kind of flexibility and versatility of SIP, but let the market decide. I like using IAX for certain tasks and will continue to explore its uses. Your zealous attitude does little to promote qualified technological exchange on this list or elsewhere, it rather reminds us of ideological battles between close-minded parties engaged in power struggles. Besides, only five years ago, SIP was the underdog and H323 was all the rage. At that point people were very sceptical if anybody would even implement SIP because it was yet another standard created by a committee, many of which had failed to catch on in the past. In this respect, IAX has a clear advantage in that it isn't designed by a committee and that it has already got a significant installed base. At VON, only a few people even understood what Asterisk was, let alone even had heard of IAX. Well, you are of course entitled to your opinion, but some folks with far more clout and credibility that you and I and pretty much everybody on this list seem to think diff
Re: [Asterisk-Users] ZTdummy
just look at zaprtc from www.junghanns.net On Mon, 1 Nov 2004 12:35:24 -0900, rich allen <[EMAIL PROTECTED]> wrote: > centrex can be digital or analog. if you can plug a standard 2500 set > into the line and get dial tone, then it is analog and will work just > like any other POTS line (* would need an FXO card). you may or may not > need to dial an access digit. 9 is the normal access digit but it may > be anything or none at all (assume dial 9) > > - hcir > > > > On Nov 1, 2004, at 12:20 PM, Tomas Carnecky wrote: > > > Paul Rodan wrote: > >> It's picky about what USB controller you have. It refused to work on > >> my > >> server because it had the wrong kind of USB controller, go figure. > >> So I used zaprtc and it worked fine. If you have problems with > >> zrdummy, let > >> me know and I'll see if I can help with zaptelrtc. The trickiest part > >> is to > >> make sure you don't have the Real Time Clock (rtc) compiled in the > >> kernel. > > > > ___ > 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 > -- Michael Bielicki ___ 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] Linux and Windows
Agreed; This conversation does not belong on this list. My reason and primary concern however is that Asterisk be hijacked by demand and deployed on an unreliable system and therefore gain the stigma of being "unreliable" since the chain is only as strong as the weakest link. At 06:41 PM 11/1/2004, you wrote: Not to say that all Govt's are like this, but we employ a LOT of open source. It comes down to a money issue. But besides that, we still use Windows, and YES it does have its place. I don't think that this thread belongs on this list. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Karl J. Vesterling Sent: Monday, November 01, 2004 5:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Linux and Windows As far as I understand, corporation, and Govt's like commercial products because of the issue of liability. That is why the Commercial market and Govt market don't accept open source solutions. What is perplexing about the whole situation is that the licensing agreements with commercial software 99.9% of the time indemnify the vendor of any and all liability. So, what we have is the "feeling" of security... Perhaps Linus should convince the various entities that distribute Linux to include a nice fluffy security blanket with the licensing agreement embroidered on it? That way the attorneys can get that "warm fuzzy feeling" they so desire. I speak from much experience regarding this matter... The U.S. Govt won't accept an open-source solution even if it is the only option to cover their ass. They'd rather leave their cheese out in the wind than cover it with an open source solution. Question: Why isn't there a commercial solution available in some cases? Answer: What company in their right mind would engineer a competing product to a solution that costs $0.00 ??? At 05:59 PM 11/1/2004, you wrote: Jay Milk wrote: Why are you so angry? At the risk of throwing oil on the fire, I would submit that Benjamin was *kidding* at the beginning of that mail, and trolling at the end. I agree with him about the "quit anytime I want," but I digress. . . It does appear his flamebait was eagerly pounced upon. Why would the Window$ user$ mind the nose-tweaking? They've got 90%+ of the market. B. ___ 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 Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Yahoo Messenger: karl_vesterling ICQ: 1548052 AOL Instant Messenger: n2vqm Telephone: Washington DC: (202) 448-3009 Extension 0 Annapolis MD: (240) 524-6706 Extension 0 Seattle WA: (360) 516-1822 Extension 0 Niagara Falls NY: (716) 286-9175 Extension 0 Buffalo NY: (716) 608-1121 Extension 0 United Kingdom: 0870 3403428 Extension 0 ___ 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 Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Yahoo Messenger: karl_vesterling ICQ: 1548052 AOL Instant Messenger: n2vqm Telephone: Washington DC: (202) 448-3009 Extension 0 Annapolis MD: (240) 524-6706 Extension 0 Seattle WA: (360) 516-1822 Extension 0 Niagara Falls NY: (716) 286-9175 Extension 0 Buffalo NY: (716) 608-1121 Extension 0 United Kingdom: 0870 3403428 Extension 0 ___ 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] Linux and Windows
At 06:51 PM 11/1/2004, you wrote: [snip for brevity[ So the U.S. Govt has never used linux anywhere? Wow. Not in most installations, and definitely not in DoD facilities. The "Office of Inspector General" has deemed open source to be "Verboten". That's going to become an interesting situation when Solaris goes open source... http://www.eweek.com/article2/0,1759,1647198,00.asp Question: Why isn't there a commercial solution available in some cases? Answer: What company in their right mind would engineer a competing product to a solution that costs $0.00 ??? Again making the mistake that open source equates non-commercial. Once again... The Office of Inspector General has deemed (any and all) open-source to be forbidden. Whether it be commercial of non-commercial open-source software it's forbidden. Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Yahoo Messenger: karl_vesterling ICQ: 1548052 AOL Instant Messenger: n2vqm Telephone: Washington DC: (202) 448-3009 Extension 0 Annapolis MD: (240) 524-6706 Extension 0 Seattle WA: (360) 516-1822 Extension 0 Niagara Falls NY: (716) 286-9175 Extension 0 Buffalo NY: (716) 608-1121 Extension 0 United Kingdom: 0870 3403428 Extension 0 ___ 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] Linux and Windows
Karl J. Vesterling wrote: At 06:41 PM 11/1/2004, you wrote: [ snip for brevity ] If I only ran linux, I'd not even be able to connect to the Internet, since all I have is an EVDO/CDMA PCMCIA modem that barely has working drivers for Windows. I wouldn't attempt to get on the highway riding a rusty old tricycle, nor would I entrust my penguin to such a device. Perhaps you should spend the $30.00 (US) and get yourself a Modem instead of a MoDumb. Try reading what people say before ranting. He has a *CDMA* modem. Can you offer him good modem support for a CDMA modem? If not, try to avoid making yourself look so foolish. Actually, most CDMA modems use AT commands, just like most GSM modems do. They both an extended set of the usual AT commands, with special commands to access cellular related features. The GSM stuff is openly documented. I think the CDMA guys are a bit more annal retentive, but they also have a spec for the standard CDMA modem interface. Regards, Steve ___ 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: OT -- RE: [Asterisk-Users] Linux and Windows
On Mon, 1 Nov 2004 13:32:03 -0600, Jay Milk <[EMAIL PROTECTED]> wrote: > Maybe we could keep the religious Windows vs. Linux discussions out of > here? I think they're hardly productive, nor do they make people who > argue for one or the other look very intelligent... Go back and read the entire thread and you will find that this was about Astwind, that I said Astwind was a good thing because it lowers the entry barrier to people wanting to do Asterisk but having fear of Linux. I still stand by that. In other discussions before I was amongst those who identified Astwind as a tool to run Asterisk on a notebook as a SIP/IAX gateway for a SIP softphone on the same notebook. A use of Asterisk that I promote and even set up a Wiki page for (search for "localhost gateway"). > posts... With the exception of the windows vs. linux discussion. To me, the platform discussion is *not* about Windoze versus Linux at all. To me the discussion is about commoditised operating systems versus Windoze. I don't care what operating system is used, as long as it recognises the fact that operating systems have become commodities and the commodity standard is Unix, whichever flavour thereof. I am even prepared to make allowances for non-Unix legacy systems, that live out the remainder of their lives in niche areas, but that would be systems like VMS, MPE and OS/400 or TPF because none of them are being abused to hold an entire industry hostage. Trying to withstand the trend of commoditisation is a folly thing to do. As an interesting read on this subject, I'd recommend the history of rubber, in particular the attempts to stop caoutchouc seeds from leaving Brazil in order to monopolise rubber. It couldn't succeed forever. Even under the threat of capital punishment, the seeds did get smuggled out of Brazil and rubber became a commodity. Later still, synthetic rubber was invented and rubber still remains a commodity. > Windows is easy to install, configure and very > intuitive to maintain. This user-friendliness is bought at a cost. You can have the same or even better user-friendliness on Unix. Look at those Mac rebranded NeXT stations and servers. That's Unix, too. It also is a good example for the commoditisation I spoke about. Mankind has bigger problems than getting bogged down in countless silly IT issues that waste an incredible amount of time and money for the sake of a monopoly. > Calling one group "lusers" is out of line and insulting, It's an old MIT tradition. You will find this to be present in many Unix commands, even making it into recent arrivals such as SSH where the username is specified with a switch -l What do you think -l stands for? it's -l as in luser. If you don't believe me, read it up somewhere. It might well be in the jargon file. > Someone here brought up a car-analogy. Consider windows a > top-of-the-line luxury sedan and linux a bare-bones SUV. Both will > essentially do the same job, getting you from point A to point B, but if > you're willing to put in the effort, your SUV will do more for you and > be tougher. You can retrofit your SUV with a fancy stereo and a > navigation system, if you're willing to do the wiring. Your luxury > sedan will already have these options ready (and EASY) to use. > > And for what it's worth, I'm sick of people arguing how linux is free > because it's open source, and Windows is oh-so-expensive. I did not make any such statement, nor did I even argue *Linux* versus Windoze as mentioned above. Linux is just one of multiple choices of Unix systems. I personally am not even all that fond of Linux. As an old DEC/VAX head, I prefer BSD over Linux. However, as far as the cost of running Windoze is concerned, even The Economist newspaper openly states for quite a number of years now that it is an open secret in accountant and investor circles that Windoze PCs are far more expensive than the mainframes they replaced, even though the whole point of replacement was the promised lower TCO in the first place. If a paper with the authority of The Economist doesn't even make an attempt to present a counter argument, we can be pretty certain that indeed, the proliferation of Windoze PCs in businesses was financially a folly. > Rather than proclaiming Linux and Windows "camps", Well, you know by now that I don't. > that there are different indications for choosing different OSs, and one > size does not fit all. Precisely my point. And the choices are the commodity systems with names ending in X and derivatives thereof. Windoze is an abnomaly, it's raison d'etre is to keep an illegal monopoly in place. In any event, this whole discussion is far too US centric. European governments are moving off Windoze and far more important, what is going to be the world's number one economic powerhouse in the not so distant future, China, has already decided decisively against Windoze. I wonder what will happen to the 1-million-fruit-flies argument when 1 billion Chinese
Re: [Asterisk-Users] Centrex
Em Seg 01 Nov 2004 19:18, Tim Sailer escreveu: > On Mon, Nov 01, 2004 at 07:13:07PM -0200, Marcelo Pacheco wrote: > > If you plug a plain analog phone to the line, then you need an FXO card > > in it's place. > > Yup. It works with a plain ol' phone, Fax, and modem, but you need to > dial 9 to get out. Just a generic install of Asterisk and having the > FXO card configured, asterisk never sees the line ring. > > > Centrex usually means that you have hired a virtual PBX system. That in > > itself doesn't mean FXS/FXO/T1/E1, a Centrex could be shipped to you in > > many diferent ways. > > sigh. I hate phone companies. I have the same issue you have, except that the issue is with a real (Brazilian) PBX. I have 2 american phones (I live in Brazil) that won't detect a ring from that PBX, but if I put another phone in parallel, so I can hear the ring, I can pick up that phone and answer, everything else works ok. However both a TDM400P FXO module and an MD3200 FXO work ok with that PBX. This issue is something this Centrex line does that is non-standard, although I'm not sure it will help, go ahead and play with /etc/zaptel.conf changing the defaultzone. Try all of them, one by one. Try putting a multimeter on the line and see what kind of ring voltage do you get. Usually the Telco put out 96Vdc, some PBXes I've seen use 48Vdc. Maybe the ring voltage is a little too low ? The ring cadence, is it non-standard ? Try usecallerid=yes and usedistinctiveringdetection=yes Run asterisk on a highly verbose setting (-vv) and see if it can detect anything at all when the line rings. Marcelo Pacheco M2J Comunicacoes e Informatica Vitoria-ES-Brazil ___ 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] Linux and Windows
>>If I only ran linux, I'd not even be able to connect to the Internet, since >>all I have is an EVDO/CDMA PCMCIA modem that barely has working drivers for >>Windows. > >I wouldn't attempt to get on the highway riding a rusty old tricycle, nor would I entrust my >penguin to such a device. Perhaps you should spend the $30.00 (US) and get yourself a >Modem instead of a MoDumb. Sure, just point me to where I can get an EVDO/CDMA modem that works with my ISP down here (Guatemala BellSouth, now purchased Telefonica), and is supported, and I’d be happy to switch. I had to get an extra laptop setup to act as gateway because their desktop drivers crash my machine. I’m just happy the latency is somewhat better than satellite. -Michael ___ 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] User problem
Clive Carter wrote: Gives following errors ERROR[16384]: chan_zap.c: 6181 mkintf : Unable to get parameters ERROR[16384]: chan_zap.c :9109 setup_zap : Unable to register channel '1' WARNING[16384] : loader.c:334 ast_load_resource : Chan_zap.so : load_module failed, returning -1 Unregistered channel type 'Tor' Unregistered channel type 'Zap' It looks like your box can't load the appropriate drivers. How are you loading the drivers? you need to have both the zaptel and wcfxo drivers loaded in order to use the card. make sure both are loaded before starting up asterisk. you can verify it by running the "dmesg" command and looking in the output for stuff related to Zaptel or wcfxo. also there might be IRQ issues - is the card sharing IRQs with anything else? flynn ___ 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] astcc configure
/bin/mail Darren Wiebe oi geli wrote: What should be the EMail Program in astcc-admin's Configure page? I tried sendmail, it did not send any mail to the Admin Email address. Thanks __ Do you Yahoo!? Check out the new Yahoo! Front Page. www.yahoo.com ___ 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] Linux and Windows
At 06:41 PM 11/1/2004, you wrote: [ snip for brevity ] If I only ran linux, I'd not even be able to connect to the Internet, since all I have is an EVDO/CDMA PCMCIA modem that barely has working drivers for Windows. I wouldn't attempt to get on the highway riding a rusty old tricycle, nor would I entrust my penguin to such a device. Perhaps you should spend the $30.00 (US) and get yourself a Modem instead of a MoDumb. Note that I'm not trying to push any agenda (although I am quite pro-MSFT). I just think that people need to take some levelheaded positions, and while ranting about how Windows has no place in the workplace might make you feel good and gain respect from leet kids, it makes it seem like you're rather out of touch with business. That's a short sighted viewpoint... I submit that business is short sighted about computing. Then again, show me a management type that has a full understanding, much less makes good decisions. From a computing standpoint Linux/Unix is a much better computing environment than windows will ever be. When deployed properly it's exceptionally easy to manage from an administration standpoint. It's proven to be more stable time and time again. However, if you want to bring emotions into the equation I can see how a management type would decide that Windows is the way to go. They tend not to stray far from the rest of the bovine herd and dare not make a decision that they can be held directly accountable for. Scenario... Bob Winders is a manager in the company that has several different computing departments. All the rest of the management types have gone with Windows because PC Ragazine has every other page advertising some sort of MicroShaft produckt. Bob Winders knows that if he goes with a Unix solution he will have to defend that position and therefore will stray from the herd. Bob makes the decision to go with a Microshaft solution knowing that he will share the rest of the blame when the network comes crashing down due to the latest bug. Since has made the same decision, therefore everyone is to blame and there is no direct accountability. THAT is but one main reason why an inferior product maintains it's market share. Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Yahoo Messenger: karl_vesterling ICQ: 1548052 AOL Instant Messenger: n2vqm Telephone: Washington DC: (202) 448-3009 Extension 0 Annapolis MD: (240) 524-6706 Extension 0 Seattle WA: (360) 516-1822 Extension 0 Niagara Falls NY: (716) 286-9175 Extension 0 Buffalo NY: (716) 608-1121 Extension 0 United Kingdom: 0870 3403428 Extension 0 ___ 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] astcc configure
What should be the EMail Program in astcc-admin's Configure page? I tried sendmail, it did not send any mail to the Admin Email address. Thanks __ Do you Yahoo!? Check out the new Yahoo! Front Page. www.yahoo.com ___ 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] Speech to Text Conversion
Cirelle Enterprises wrote: has anybody found anything which works for speech to text translation? Implementation being instead of (or as well as) vm wav file being sent in email, a text translation would accompany the wav file Speech to text is just around the corner. It has been for 30 years. :-) Steve ___ 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: How far is IAX to be a Standard
> I'd say some learning on high > availability Linux/clustering etc > is in order. I know all about it, but 50 boxes is just too much. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Tuesday, November 02, 2004 1:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: How far is IAX to be a Standard [EMAIL PROTECTED] wrote: --SNIPSTA-- > Anyhow, the situation now, is that there is no DSP chip, that means .. > Your main processor has to encode the channel in total (3 to 4 E1's > absolute is the max possible with dual xeon 3 ghz I read somewhere in > this case) This has a problem. PC's are so cheap now that you'd be silly to have more than 4 E1s on one PC. The idea is not to build one huge PC, but multiple small ones. This helps no end when trying to add redundancy etc. I.E. if you have say 5000 lines on one PC (to take it to the extreme) and that PC dies, you have just lost 5000 lines. If you have 100 lines per sever and 50 servers, you only lose 100 lines if the PC dies, and you can probably reroute to other servers. > Another method is to send the incoming IAX on asterisk out again with > SIP to a gateway with hardware DSP's.. (like we do).. This needs less > performance ofcourse because asterisk doesn't have to do codec > encoding, but nevertheless will still have to transcode to get the > signaling and RTP merged and submerged from/to this one IAX port to > separate Signaling/RTP ports.. Our setup now is the second scenario.. > And my first (rough) calculations are that a dual xeon 3.0 ghz can > handle about 500 concurrent channels in this scenario... As above. > Ever wandered why there isn't any codec (DSP) hardware availiable for > asterisk?? I think here is the answer, because it is very hard to > make, Digium should then be able to design a totally new DSP chip design .. > And that's much more difficult than to design an E1 board. No. The reason that there is none is that Asterisk is designed to use cheap off the shelf PC hardware. > Our case is that we have about 200 E1's of voip (h323 and sip) traffic > and are still expanding. If we would have this all on IAX this would > be unmanageable, we would need 50 linux boxes. As above. > Conclusion.. IAX Is a good performer behind NAT and perfect for small > setups but to work in an enterprise, Much work has to be done. I'd say some learning on high availability Linux/clustering etc is in order. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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 and GnuGK on the same box?
just curios, John, have you had any success on communication between gnugk and iMerge before without asterisk? I guess check lists would be turning off your test box firewall ( iptables etc ) and make sure all the necessary port are available for h323 call. In my case, I fowarded all calls to monkey player example and see anything plays. H323 channel overall in asterisk is not first citizen. If I can avoid, I would not use it but well, in reality, all this avaya and lucent was built around h323. good luck, On Mon, 01 Nov 2004 17:43:11 -0500, John Gray <[EMAIL PROTECTED]> wrote: > And on further inspection. The iMerge never sends anything to asterisk > after the incoming call was terminated. > > John > > > > John Gray wrote: > > > We have asterisk and the Lucent iMerge working together now. > > > > http://www.voip-info.org/tiki-index.php?page=Asterisk+Lucent+iMerge+Configuration > > > > > > On thing that isn't working well yet. If you call in from the > > outside, wait for the phone to ring on the inside, then hang up. > > Asterisk doens't seem to realize that the call has been terminated. > > It keeps ringing the phone. > > > > John > > > > John Gray wrote: > > > >> Hello Gang, > >> > >> I'm trying to get asterisk to play with a Lucent iMerge. It seems to > >> that GnuGK talks to it a bit better. So I'm trying to get this: > >> > >> PSTN->iMerge->GnuGK->Asterisk. > >> > >> I'd like to get GnuGK and Asterisk running on the same box. Do they > >> get in each others way? > >> > >> Any tricks to getting them both going and talking to one another on > >> the same box? > >> > >> If they conflicting on ports, I suppse an option is to assign the box > >> two IP and have them listening on two different IPs? > >> > >> Thanks, > >> > >> John > >> > > > > > > -- > John Gray [EMAIL PROTECTED] > AgoraNet, Inc. (302) 224-2475 > 102 E. Main Street, Suite 303 (302) 224-2552 (fax) > Newark, De 19711http://www.agora-net.com > > ___ > 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] chan_sccp
I know this came up on the list in the last few weeks, but exactly what is the location for the most recent cvs of chan_sccp [chan_sccp2]? I've looked at the sourceforge cvs, but the most recent change there is like 8 or 9 weeks ago, is that correct? Thanks, Chris -- Christopher L. Wade Unistar-Sparco Computers, Inc. Senior Systems Administratordba Sparco.com Email: [EMAIL PROTECTED] 7089 Ryburn Drive Phone: (901) 872 2272 / (800) 840 8400Millington, TN 38053 Fax: (901) 872 8482 USA ___ 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: How far is IAX to be a Standard
[EMAIL PROTECTED] wrote: --SNIPSTA-- Anyhow, the situation now, is that there is no DSP chip, that means .. Your main processor has to encode the channel in total (3 to 4 E1's absolute is the max possible with dual xeon 3 ghz I read somewhere in this case) This has a problem. PC's are so cheap now that you'd be silly to have more than 4 E1s on one PC. The idea is not to build one huge PC, but multiple small ones. This helps no end when trying to add redundancy etc. I.E. if you have say 5000 lines on one PC (to take it to the extreme) and that PC dies, you have just lost 5000 lines. If you have 100 lines per sever and 50 servers, you only lose 100 lines if the PC dies, and you can probably reroute to other servers. Another method is to send the incoming IAX on asterisk out again with SIP to a gateway with hardware DSP's.. (like we do).. This needs less performance ofcourse because asterisk doesn't have to do codec encoding, but nevertheless will still have to transcode to get the signaling and RTP merged and submerged from/to this one IAX port to separate Signaling/RTP ports.. Our setup now is the second scenario.. And my first (rough) calculations are that a dual xeon 3.0 ghz can handle about 500 concurrent channels in this scenario... As above. Ever wandered why there isn't any codec (DSP) hardware availiable for asterisk?? I think here is the answer, because it is very hard to make, Digium should then be able to design a totally new DSP chip design .. And that's much more difficult than to design an E1 board. No. The reason that there is none is that Asterisk is designed to use cheap off the shelf PC hardware. Our case is that we have about 200 E1's of voip (h323 and sip) traffic and are still expanding. If we would have this all on IAX this would be unmanageable, we would need 50 linux boxes. As above. Conclusion.. IAX Is a good performer behind NAT and perfect for small setups but to work in an enterprise, Much work has to be done. I'd say some learning on high availability Linux/clustering etc is in order. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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: How far is IAX to be a Standard
>Precisely. First world environment! >Try travelling in Africa, the Middle East and South Asia, where >everything is still mostly dialup and many of the phone wires >installed go back before the time when plastic was invented. So SIP doesn't work on dialup? That's funny 'cause I'm using it like that... >> Simply writing a better spec and then hoping >> that magically the entire world is going to support it > >Come on. You know as well as I do that this is not representing the >situation at all. In fact it fails to give credit to Mark and others >who have been working hard to make IAX what it is today. It's not fair >to say that all they did was "write a better spec and hope". >... I didn't mean to infer that that's all that IAX was or that was all the community had accomplished. If you read back in the thread, I'm responding to someone who said something to effect of "we don't need to be a standard 'cause we're just that good". I think that's a shortsighted approach. That's all. If you don't agree that having IAX ratified into a standard would be a help, well, that's where we differ. -Michael ___ 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] anyuser i-3100
Has anyone had any experience with the Anyuser IP Phones (I-3100) They are reported to use H.323 and are real cheap $25. I figured this might be a quick way to dive into asterisk? is this a case of you get what you pay for? is there a better really cheap solution for IP hard phones? thanks, Justin ___ 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: How far is IAX to be a Standard
[EMAIL PROTECTED] wrote: Hello IAX really isn't the 'one and only' perfect signaling protocol because many people forget one thing IAX has one technical issue (by design) which makes it difficult to ever get accepted by the big boys, a real big problem for carriers who have big loads on their systems like we do. With IAX the audio (RTP) and signaling goes embedded over one port. We all know that the big advantage ofcourse is that this makes it an excellent performer behind a NAT, but the big disadvantage is that there is not any DSP chip available in the market which is able to get the codecs encoded and put into this embedded rtp+signaling channel, and I wander if there ever will be because another piece of software does the signaling (asterisk in this case) asterisk would have to 'tell' the DSP chip the signaling packets to embed into the IAX/RTP channel.. That would be a whole new DSP standard, Will any chipmaker (besides digium) ever see the need to design such a chip? I think this is a non-issue. splitting up the data from the signaling is very easy to do, and in a two-chip solution (general purpose + DSP), the GP CPU will handle the IAX (and UDP, and IP, etc) protocols, and hand only the codec payload to the DSP for processing. With RTP, it is still handled the same way, because the GP CPU will still be handling the UDP and IP layers, and I would imagine unwrapping/wrapping up RTP as well. RTP makes some kinds of QoS simpler to do, because the signaling and media are separated, but there really doesn't seem to me to be any kind of major loss from having signalling in the same QoS realm, since it is so small and insignificant compared to the media itself. IAX' biggest problem is that the market momentum is behind RTP-based protocols, and the market will probably (eventually) build it's advantages (NAT-transparency, trunking) into SIP or something, before they switch to IAX. (I.e. you could enhance SIP to have an option of presenting data and control on the same port, or do perform some kind of trunking, etc). Not being documented is probably it's second biggest problem. Third is that it isn't as feature-complete as RTP/RTCP, etc. ___ 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] Linux and Windows
Jay Milk wrote: I make six figures from Windows. I have yet to earn a penny with Linux. That hardly makes your opinion unbiased. You are obviously a windows shop or you suggest windows. There are many of us who make our money mainly from Linux. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] Linux and Windows
As far as I understand, corporation, and Govt's like commercial products because of the issue of liability. That is why the Commercial market and Govt market don't accept open source solutions. Both governments (Germany anyone?) and commercial entities (quite a few people have made money with Asterisk) accept open source. Open source != non commercial. In fact, sometimes they’ll want to have source for even closed source products. It might not be feasible (I’ve had access and looked at the Windows source, and it’s huge… I doubt anyone outside of MS would be able to do a thorough inspection.) What is perplexing about the whole situation is that the licensing agreements with commercial software 99.9% of the time indemnify the vendor of any and all liability. So, what we have is the "feeling" of security... Perhaps Linus should convince the various entities that distribute Linux to include a nice fluffy security blanket with the licensing agreement embroidered on it? That way the attorneys can get that "warm fuzzy feeling" they so desire. Isn’t that what Red Hat is doing? As far as security, it’s not always just a feeling. A company with a lot of money invested in a product (say, MS), is going to make sure the product suits most of their customers as best as possible. A non-commercial open source product might not always have that goal, instead trying to build “better technology”. Go read Raymond Chen’s blog: http://weblogs.asp.net/oldnewthing/. Read the history category. See the evil, horrible compatibility hacks that are in Windows just to make sure apps keep working as people upgrade, even when vendors do Bad Things? This is the kind of commitment that makes companies feel secure that things will continue to work. Open source doesn’t have anything to do with this argument. The real argument is non-commercial versus commercial. So it’s basically coming down to “companies like to deal with other companies rather than a couple of coders doing something for fun” … well, duh. I speak from much experience regarding this matter... The U.S. Govt won't accept an open-source solution even if it is the only option to cover their ass. They'd rather leave their cheese out in the wind than cover it with an open source solution. So the U.S. Govt has never used linux anywhere? Wow. Question: Why isn't there a commercial solution available in some cases? Answer: What company in their right mind would engineer a competing product to a solution that costs $0.00 ??? Again making the mistake that open source equates non-commercial. -Michael ___ 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: How far is IAX to be a Standard
On Mon, 1 Nov 2004 17:14:37 -0600, Michael Giagnocavo <[EMAIL PROTECTED]> wrote: > Hi there, I'm in Guatemala. My current connectivity is via a modem (poorly > implemented EVDO or CDMA, whichever is actually working at the moment). > Before that we were using satellite. The telco tried to hook up ADSL (@ $229 > for 512K) but fried the phone line, and two weeks later it's still messed > up. Hardly what I'd call comfortable or first world. Precisely. First world environment! Try travelling in Africa, the Middle East and South Asia, where everything is still mostly dialup and many of the phone wires installed go back before the time when plastic was invented. > Simply writing a better spec and then hoping > that magically the entire world is going to support it Come on. You know as well as I do that this is not representing the situation at all. In fact it fails to give credit to Mark and others who have been working hard to make IAX what it is today. It's not fair to say that all they did was "write a better spec and hope". IAX is a reality, it is already being supported by an increasing number of VoIP service providers who use it to deliver millions of minutes of phone calls. IAX is now used and supported even by Jeff Pulver and FWD, recognised by many in the industry as a pioneer of VoIP. There is at least one chip manufacturer who has chosen to built IAX support into their chip and there are several phone manufacturers supporting IAX or are about to support IAX. This is definitely well beyond "writing a better spec and hope". rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] Linux and Windows
Not to say that all Govt's are like this, but we employ a LOT of open source. It comes down to a money issue. But besides that, we still use Windows, and YES it does have its place. I don't think that this thread belongs on this list. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Karl J. Vesterling Sent: Monday, November 01, 2004 5:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Linux and Windows As far as I understand, corporation, and Govt's like commercial products because of the issue of liability. That is why the Commercial market and Govt market don't accept open source solutions. What is perplexing about the whole situation is that the licensing agreements with commercial software 99.9% of the time indemnify the vendor of any and all liability. So, what we have is the "feeling" of security... Perhaps Linus should convince the various entities that distribute Linux to include a nice fluffy security blanket with the licensing agreement embroidered on it? That way the attorneys can get that "warm fuzzy feeling" they so desire. I speak from much experience regarding this matter... The U.S. Govt won't accept an open-source solution even if it is the only option to cover their ass. They'd rather leave their cheese out in the wind than cover it with an open source solution. Question: Why isn't there a commercial solution available in some cases? Answer: What company in their right mind would engineer a competing product to a solution that costs $0.00 ??? At 05:59 PM 11/1/2004, you wrote: Jay Milk wrote: Why are you so angry? At the risk of throwing oil on the fire, I would submit that Benjamin was *kidding* at the beginning of that mail, and trolling at the end. I agree with him about the "quit anytime I want," but I digress. . . It does appear his flamebait was eagerly pounced upon. Why would the Window$ user$ mind the nose-tweaking? They've got 90%+ of the market. B. ___ 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 Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Yahoo Messenger: karl_vesterling ICQ: 1548052 AOL Instant Messenger: n2vqm Telephone: Washington DC: (202) 448-3009 Extension 0 Annapolis MD: (240) 524-6706 Extension 0 Seattle WA: (360) 516-1822 Extension 0 Niagara Falls NY: (716) 286-9175 Extension 0 Buffalo NY: (716) 608-1121 Extension 0 United Kingdom: 0870 3403428 Extension 0 ___ 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] Linux and Windows
>This has nothing to do with anger, but it has all to do with truthfulness. > >How many times have you had discussions with Windoze folks trying to >give you good reasongs why Windoze should have its place when in >reality it only came down to the one thing that they would not admit: >all they really cared about is the ability to play all those games. > >So, I say, if you want to play games and that is what drives your >choice, then say so and don't come up with all these pretense >arguments. But I also say that playing games is not an activity that >one should expect to get paid for. Are you for real? If I only ran linux, I'd not even be able to connect to the Internet, since all I have is an EVDO/CDMA PCMCIA modem that barely has working drivers for Windows. Note that I'm not trying to push any agenda (although I am quite pro-MSFT). I just think that people need to take some levelheaded positions, and while ranting about how Windows has no place in the workplace might make you feel good and gain respect from leet kids, it makes it seem like you're rather out of touch with business. -Michael ___ 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] Linux and Windows
As far as I understand, corporation, and Govt's like commercial products because of the issue of liability. That is why the Commercial market and Govt market don't accept open source solutions. What is perplexing about the whole situation is that the licensing agreements with commercial software 99.9% of the time indemnify the vendor of any and all liability. So, what we have is the "feeling" of security... Perhaps Linus should convince the various entities that distribute Linux to include a nice fluffy security blanket with the licensing agreement embroidered on it? That way the attorneys can get that "warm fuzzy feeling" they so desire. I speak from much experience regarding this matter... The U.S. Govt won't accept an open-source solution even if it is the only option to cover their ass. They'd rather leave their cheese out in the wind than cover it with an open source solution. Question: Why isn't there a commercial solution available in some cases? Answer: What company in their right mind would engineer a competing product to a solution that costs $0.00 ??? At 05:59 PM 11/1/2004, you wrote: Jay Milk wrote: Why are you so angry? At the risk of throwing oil on the fire, I would submit that Benjamin was *kidding* at the beginning of that mail, and trolling at the end. I agree with him about the "quit anytime I want," but I digress. . . It does appear his flamebait was eagerly pounced upon. Why would the Window$ user$ mind the nose-tweaking? They've got 90%+ of the market. B. ___ 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 Best Regards, Karl J. Vesterling E-Mail: [EMAIL PROTECTED] Yahoo Messenger: karl_vesterling ICQ: 1548052 AOL Instant Messenger: n2vqm Telephone: Washington DC: (202) 448-3009 Extension 0 Annapolis MD: (240) 524-6706 Extension 0 Seattle WA: (360) 516-1822 Extension 0 Niagara Falls NY: (716) 286-9175 Extension 0 Buffalo NY: (716) 608-1121 Extension 0 United Kingdom: 0870 3403428 Extension 0 ___ 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: How far is IAX to be a Standard
Hello IAX really isn't the 'one and only' perfect signaling protocol because many people forget one thing IAX has one technical issue (by design) which makes it difficult to ever get accepted by the big boys, a real big problem for carriers who have big loads on their systems like we do. With IAX the audio (RTP) and signaling goes embedded over one port. We all know that the big advantage ofcourse is that this makes it an excellent performer behind a NAT, but the big disadvantage is that there is not any DSP chip available in the market which is able to get the codecs encoded and put into this embedded rtp+signaling channel, and I wander if there ever will be because another piece of software does the signaling (asterisk in this case) asterisk would have to 'tell' the DSP chip the signaling packets to embed into the IAX/RTP channel.. That would be a whole new DSP standard, Will any chipmaker (besides digium) ever see the need to design such a chip? Anyhow, the situation now, is that there is no DSP chip, that means .. Your main processor has to encode the channel in total (3 to 4 E1's absolute is the max possible with dual xeon 3 ghz I read somewhere in this case) Another method is to send the incoming IAX on asterisk out again with SIP to a gateway with hardware DSP's.. (like we do).. This needs less performance ofcourse because asterisk doesn't have to do codec encoding, but nevertheless will still have to transcode to get the signaling and RTP merged and submerged from/to this one IAX port to separate Signaling/RTP ports.. Our setup now is the second scenario.. And my first (rough) calculations are that a dual xeon 3.0 ghz can handle about 500 concurrent channels in this scenario... Ever wandered why there isn't any codec (DSP) hardware availiable for asterisk?? I think here is the answer, because it is very hard to make, Digium should then be able to design a totally new DSP chip design .. And that's much more difficult than to design an E1 board. Our case is that we have about 200 E1's of voip (h323 and sip) traffic and are still expanding. If we would have this all on IAX this would be unmanageable, we would need 50 linux boxes. Conclusion.. IAX Is a good performer behind NAT and perfect for small setups but to work in an enterprise, Much work has to be done. Niels. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Giagnocavo Sent: Monday, November 01, 2004 11:13 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Re: How far is IAX to be a Standard Unless SIP just plain does not work, I think it'll be hard (for IAX to get excellent acceptance), without a lot of good marketing and other efforts by Digium. At VON, only a few people even understood what Asterisk was, let alone even had heard of IAX. Even with IT people controlling things you still see a lack of good decisions. How many people do SSL POP3? How many people use S/MIME? All very simple things that don't require reworking of dedicated hardware or huge industry efforts. My only point is that we can't just rely on a better design to somehow magically win out. Getting Digium to create a standard with input from other vendors would be a huge plus and help pave the path forward. -Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, November 01, 2004 12:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: How far is IAX to be a Standard I see your point and it is well taken but I feel that with convergence you are going to see more IT staff in charge of phone systems. In turn, I see more research and informed decisions going on, not just a consumer following what is being pushed. That being said, maybe I am just looking at a grey sky through blue sunglasses. > >From: [EMAIL PROTECTED] > [mailto:asterisk-users->[EMAIL PROTECTED] On Behalf Of Steve > Totaro > >>I predict a paradigm shift, rendering what is historical, null. > > Is this in general, applying to all technology, or just to telecom? (i.e., > will Hollywood ship videos in hi-def on EVD?) > > -Michael > >> >No worries here. What works best will win out eventually. >> >> Not sure where you get that idea, as historically it's not that way :S. >> >> Companies will make SIP work reasonably enough. What will win out is >> whatever is marketed and sold the best. Getting published specs, inc. >> being >> a published "standard" is part of that marketing. >> >> -Michael >> >> - Original Message - >> From: "Randy Bush" <[EMAIL PROTECTED]> >> To: "Voip Business" <[EMAIL PROTECTED]> >> Cc: <[EMAIL PROTECTED]> >> Sent: Monday, November 01, 2004 12:44 PM >> Subject: [Asterisk-Users] Re: How far is IAX to be a Standard >> >> what does the RFC's guys and the Pseudo-Cisco IETF think about this Protocol? >>> >>> the internet vendor task force has a mass
Re: [Asterisk-Users] Linux and Windows
On Mon, 1 Nov 2004 16:56:27 -0600, Jay Milk <[EMAIL PROTECTED]> wrote: > Why are you so angry? This has nothing to do with anger, but it has all to do with truthfulness. How many times have you had discussions with Windoze folks trying to give you good reasongs why Windoze should have its place when in reality it only came down to the one thing that they would not admit: all they really cared about is the ability to play all those games. So, I say, if you want to play games and that is what drives your choice, then say so and don't come up with all these pretense arguments. But I also say that playing games is not an activity that one should expect to get paid for. > don't like it, don't use it, In fact I don't. > but don't insult the people who do. Who is the one making is the insult here? The one with the pretense argument or the one who says one shouldn't expect to get paid for playing games? > accept that there are valid uses Vaild uses, sure. Question is whether that includes the workplace. I believe it doesn't. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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: How far is IAX to be a Standard
>On Mon, 1 Nov 2004 16:13:15 -0600, Michael Giagnocavo ><[EMAIL PROTECTED]> wrote: >> Unless SIP just plain does not work, I think it'll be hard (for IAX to >get >> excellent acceptance), > >Funny you should say that from the comfort of your first world >environment. In many countries internet infrastructure is such that >SIP does indeed simply not work, but IAX does. Hi there, I'm in Guatemala. My current connectivity is via a modem (poorly implemented EVDO or CDMA, whichever is actually working at the moment). Before that we were using satellite. The telco tried to hook up ADSL (@ $229 for 512K) but fried the phone line, and two weeks later it's still messed up. Hardly what I'd call comfortable or first world. >Of course IAX, once going RFC, will have to start its own life outside >of Asterisk. My only point has been that IAX needs support. Marketing (in forms of standardization, or even "open source gurus" spouting stuff) needs to encourage people to pick it up. Simply writing a better spec and then hoping that magically the entire world is going to support it is, well, an interesting concept. (Hey, I hope it works!) >> My only point is that we can't just rely on a better design to somehow >> magically win out. Getting Digium to create a standard with input from >other >> vendors would be a huge plus and help pave the path forward. > >I totally disagree. IAX is not a standards committee protocol. It's a >grassroots thing. If it catches on, and we have any reason to believe >that it will, given how far it has already come, then it will be >through grassroots implementations. It will be through small vendors >who seek an edge and use IAX to get that edge, forcing other small >vendors to follow. Once there is a critical mass, even the bigger >vendors won't be able to ignore it. Well, I guess that's certainly one approach. I still don't see why this is a replacement or excuse for not trying other approaches. Nothing says you have to be purely grassroots. Also, you can bet that some companies who are making $$$,$$$ selling boxes just to "fix" SIP are going to be throwing lots of money against stuff that puts them out of business... -Michael ___ 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] Re: Centrex
On Mon, 1 Nov 2004 Tim Sailer wrote: > OK folks, > I'm trying to help get another remote Asterisk box up and running. > The system currently has a single FXO card, but it doesn't seem to > be working, my guess is because the inbound line is CENTREX. Knowing > nothing about Centrex, can someone tell me if I'm right, and need an > FXS card? Centrex in its most common form is physically the same as a loop start POTS line. If you can plug a plain old phone into it, get dial tone, and make and receive calls an FXO card will work. You'll need to set the dialplan to input a leading 9 to call out to the PSTN from the Centrex line, and you can also build a dialplan to call other Centrex stations of the business with typically three or four digits. Generally these inter-office calls are free of message unit charges. If a regular phone rings but the Asterisk doesn't detect ringing, try swapping the two wires of the line, also make sure that the Asterisk has a good ground. Some ringing circuits are fussy that way. -- Jay Hennigan - CCIE #7880 - Network Administration - [EMAIL PROTECTED] WestNet: Connecting you to the planet. 805 884-6323 WB6RDV NetLojix Communications, Inc. - http://www.netlojix.com/ ___ 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] Linux and Windows
>You can play with your toys at home, but there is absolutely no reason >why your employer should provide you toys at work simply because you >have them at home. [rest of sermon snipped] This is a joke right? While I'm amazed at your enlightenment of how a computer is supposed to be used, in my case, I'm a consultant that works from home. I'm also a hobbyist computer user that enjoys all aspects of what computing has to offer. I've setup many terminal server environments for clients that utilize "windoze" (isn't that funny, it's like the OS is sleeping!) that only allow your Orwellian approach to computer usage, so I'm not foreign to the concept. These systems are cheap and extremely fast & easy to administer. It costs my clients more to drop a phone on a new employee's desk then it does to drop a terminal (hence my interest in VOIP). But I digress.. Thanks for the morality lesson, may the sun always rise on your staff. -Bryan ___ 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] Linux and Windows
Yes, if you take his last mail out of context, it could have been kidding/trolling. But if you read it in the context of his many other emails, you'll understand that he's quite serious about his opinion of Windows and its users. And it's just plain sad to see someone who's otherwise so insightful and articulate resort to such poor generalizations. I make six figures from Windows. I have yet to earn a penny with Linux. I like both, I use both. > -Original Message- > From: Brian Capouch [mailto:[EMAIL PROTECTED] > Sent: Monday, November 01, 2004 5:00 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Linux and Windows > > > Jay Milk wrote: > > Why are you so angry? > > At the risk of throwing oil on the fire, I would submit that Benjamin > was *kidding* at the beginning of that mail, and trolling at the end. > > I agree with him about the "quit anytime I want," but I digress. . . > > It does appear his flamebait was eagerly pounced upon. Why would the > Window$ user$ mind the nose-tweaking? They've got 90%+ of the market. > > B. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/aster> isk-users > To > UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [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: How far is IAX to be a Standard
On Mon, 1 Nov 2004 16:13:15 -0600, Michael Giagnocavo <[EMAIL PROTECTED]> wrote: > Unless SIP just plain does not work, I think it'll be hard (for IAX to get > excellent acceptance), Funny you should say that from the comfort of your first world environment. In many countries internet infrastructure is such that SIP does indeed simply not work, but IAX does. IAX is so vastly superior to SIP, that the comparison shouldn't be things like VHS versus Betamax, but it should be more like horse carriages versus motorcars. Besides, only five years ago, SIP was the underdog and H323 was all the rage. At that point people were very sceptical if anybody would even implement SIP because it was yet another standard created by a committee, many of which had failed to catch on in the past. In this respect, IAX has a clear advantage in that it isn't designed by a committee and that it has already got a significant installed base. > At VON, only a few people even understood what Asterisk was, let > alone even had heard of IAX. Well, you are of course entitled to your opinion, but some folks with far more clout and credibility that you and I and pretty much everybody on this list seem to think differently. A few weeks ago there was a statement from an open source guru at an event making the mainstream news all over the world and the statement was this: "Watch out for Asterisk, it will be bigger than Linux". This wasn't coming from an Asterisk zealot driven by wishful thinking. It was a guy with enough clout to make top headlines in the mainstream media. Of course IAX, once going RFC, will have to start its own life outside of Asterisk. > My only point is that we can't just rely on a better design to somehow > magically win out. Getting Digium to create a standard with input from other > vendors would be a huge plus and help pave the path forward. I totally disagree. IAX is not a standards committee protocol. It's a grassroots thing. If it catches on, and we have any reason to believe that it will, given how far it has already come, then it will be through grassroots implementations. It will be through small vendors who seek an edge and use IAX to get that edge, forcing other small vendors to follow. Once there is a critical mass, even the bigger vendors won't be able to ignore it. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] Linux and Windows
Jay Milk wrote: Why are you so angry? At the risk of throwing oil on the fire, I would submit that Benjamin was *kidding* at the beginning of that mail, and trolling at the end. I agree with him about the "quit anytime I want," but I digress. . . It does appear his flamebait was eagerly pounced upon. Why would the Window$ user$ mind the nose-tweaking? They've got 90%+ of the market. B. ___ 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] Linux and Windows
Why are you so angry? Did Bill Gates take your lunch money? If you don't like it, don't use it, but don't insult the people who do. I used both, for different reasons. Neither is perfect. Seriously, let's lose the rhetoric, accept that there are valid uses for linux and windows, and GET ON WITH OUR LIVES. > -Original Message- > From: Benjamin on Asterisk Mailing Lists > [mailto:[EMAIL PROTECTED] > Sent: Monday, November 01, 2004 4:35 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Linux and Windows > > > On Mon, 1 Nov 2004 11:06:28 -0700, public > <[EMAIL PROTECTED]> wrote: > > > I'm an unashamed windows user. > > Nobody is. It's like alcoholics, they don't admit that they > have a drinking problem, because they can stop any time they > like, right? > > > So I like to game, sue me ;) > > Very good point. > > Leading right to the only reason why Windoze even exists in the > workplace: People who abuse their employers and waste their > time at work with entertainment instead of doing their jobs. > > You can play with your toys at home, but there is absolutely > no reason why your employer should provide you toys at work > simply because you have them at home. > > A workplace is a workplace, not a kindergarden and toys > should not be tolerated at work. That includes Windoze. > > rgds > benjk > > -- > Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 > Shibuya, Tokyo, Japan. > > NB: Spam filters in place. Messages unrelated to the * > mailing lists may get trashed. > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/aster> isk-users > To > UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [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] UDP Fragmentation Problem
Rich Adamson schrieb: As far as I am aware there is no such thing as a fragmented UDP packet; each packet is sent out on its own, there is no coherency between UDP packets like there is with TCP packets. I could be very wrong here, it's been a late night with the kids. :-) Packet fragmentation is at the IP layer, so UDP will have fragmented packets too. But... the OS should handle that and Asterisk shouldn't find out - it's a all or none policy, so it should receive the whole packet at once or nothing. How I can setup Linux to handle UDP fragments? Not sure why the concern with fragmentation, it should not be an issue with any modern linux distribution and there is nothing to setup. The only issue that I've heard about in recent months/years relative to fragmentation is the SonicWall firewall just can't seem to get it right. In their case, any udp packet greater then about 1500 bytes does not get reassembled propery, and its still an issue in the latest firmware. If you really think you've got a fragmentation problem, I'd like to see a packet trace (eg, ethereal) of those packets. Here it is ;-) Okay, looked at the pcap and see the fragmentation, but that does not indicate your asterisk IP stack is not handling it properly. Might compare a 'sip debug' with those packets to see if data is reassembled. Since both pieces of the original fragmented packet did in fact arrive at your destination, the only issue left is whether your IP stack reassembled them properly. I'd suspect another problem is lurking unrelated to fragmentation. I think you're right it seems to be the client side in Colombia. Tomorrow I will perform a trace there. You've got a idea problem it could be? Regards Bastian Rich ___ 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] Linux and Windows
> Very good point. > > Leading right to the only reason why Windoze even exists in the > workplace: People who abuse their employers and waste their time at > work with entertainment instead of doing their jobs. > > You can play with your toys at home, but there is absolutely no reason > why your employer should provide you toys at work simply because you > have them at home. > > A workplace is a workplace, not a kindergarden and toys should not be > tolerated at work. That includes Windoze. This is a joke right? While I'm amazed at your enlightenment of how a computer is supposed to be used, in my case, I'm a consultant that works from home. I'm also a hobbyist computer user that enjoys all aspects of what computing has to offer. I've setup many terminal server environments for clients that utilize "windoze" (isn't that funny, it's like the OS is sleeping!) that only allow your Orwellian approach to computer usage, so I'm not foreign to the concept. These systems are cheap and extremely fast & easy to administer. It costs my clients more to drop a phone on a new employee's desk then it does to drop a terminal (hence my interest in voip). But I digress.. Thanks for the morality lesson, may the sun always rise on your staff. -Bryan ___ 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] Linux and Windows
Great, now that you've proved you're an 31337 professional, not a "windoze luser", equated Windows to drugs, and made it clear that Windows isn't used in any real business, can we move on to another topic? -Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin on Asterisk Mailing Lists Sent: Monday, November 01, 2004 4:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Linux and Windows On Mon, 1 Nov 2004 11:06:28 -0700, public <[EMAIL PROTECTED]> wrote: > I'm an unashamed windows user. Nobody is. It's like alcoholics, they don't admit that they have a drinking problem, because they can stop any time they like, right? > So I like to game, sue me ;) Very good point. Leading right to the only reason why Windoze even exists in the workplace: People who abuse their employers and waste their time at work with entertainment instead of doing their jobs. You can play with your toys at home, but there is absolutely no reason why your employer should provide you toys at work simply because you have them at home. A workplace is a workplace, not a kindergarden and toys should not be tolerated at work. That includes Windoze. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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 and GnuGK on the same box?
And on further inspection. The iMerge never sends anything to asterisk after the incoming call was terminated. John John Gray wrote: We have asterisk and the Lucent iMerge working together now. http://www.voip-info.org/tiki-index.php?page=Asterisk+Lucent+iMerge+Configuration On thing that isn't working well yet. If you call in from the outside, wait for the phone to ring on the inside, then hang up. Asterisk doens't seem to realize that the call has been terminated. It keeps ringing the phone. John John Gray wrote: Hello Gang, I'm trying to get asterisk to play with a Lucent iMerge. It seems to that GnuGK talks to it a bit better. So I'm trying to get this: PSTN->iMerge->GnuGK->Asterisk. I'd like to get GnuGK and Asterisk running on the same box. Do they get in each others way? Any tricks to getting them both going and talking to one another on the same box? If they conflicting on ports, I suppse an option is to assign the box two IP and have them listening on two different IPs? Thanks, John -- John Gray [EMAIL PROTECTED] AgoraNet, Inc. (302) 224-2475 102 E. Main Street, Suite 303 (302) 224-2552 (fax) Newark, De 19711http://www.agora-net.com ___ 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 and GnuGK on the same box?
We have asterisk and the Lucent iMerge working together now. http://www.voip-info.org/tiki-index.php?page=Asterisk+Lucent+iMerge+Configuration On thing that isn't working well yet. If you call in from the outside, wait for the phone to ring on the inside, then hang up. Asterisk doens't seem to realize that the call has been terminated. It keeps ringing the phone. John John Gray wrote: Hello Gang, I'm trying to get asterisk to play with a Lucent iMerge. It seems to that GnuGK talks to it a bit better. So I'm trying to get this: PSTN->iMerge->GnuGK->Asterisk. I'd like to get GnuGK and Asterisk running on the same box. Do they get in each others way? Any tricks to getting them both going and talking to one another on the same box? If they conflicting on ports, I suppse an option is to assign the box two IP and have them listening on two different IPs? Thanks, John -- John Gray [EMAIL PROTECTED] AgoraNet, Inc. (302) 224-2475 102 E. Main Street, Suite 303 (302) 224-2552 (fax) Newark, De 19711http://www.agora-net.com ___ 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] Linux and Windows
On Mon, 1 Nov 2004 11:06:28 -0700, public <[EMAIL PROTECTED]> wrote: > I'm an unashamed windows user. Nobody is. It's like alcoholics, they don't admit that they have a drinking problem, because they can stop any time they like, right? > So I like to game, sue me ;) Very good point. Leading right to the only reason why Windoze even exists in the workplace: People who abuse their employers and waste their time at work with entertainment instead of doing their jobs. You can play with your toys at home, but there is absolutely no reason why your employer should provide you toys at work simply because you have them at home. A workplace is a workplace, not a kindergarden and toys should not be tolerated at work. That includes Windoze. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] Queue Prioritization
I have a client that uses two queues for a customer support application. One queue is for english speaking customers. The other is for spanish. This is a small company and they have one dedicated call agent for answering spanish calls. They don't get many spanish calls, so this agent is also part of the english queue. They wish to prioritize spanish queues for this agent over english. Without any manipulation the opposite is happening. They english queue seems to swamp the spanish queue. IE no mater how long a spanish queue member waits if there is an english queue call it will go to the spanish operator. We are using dynamic queue logins, since the customer prefers this interface. Is there a way to make the spanish queue have higher priority? I don't know of a way to prioritize inter queues, but it certainly seems that English gets checked first. I have tried swaping the order in the queue.conf file and I thought this had some impact, but the client still reports problems. Are they handled alphebetically or some other way? Anyone know how to resolve this? -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Visit Winfield Public Schools at http://usd465.com - 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
Re: [Asterisk-Users] goto() results in invalid extension
Michael Rowley wrote: Yeah, yeah, rtfm, I know... :) This is after several editions. The original was trying to refer to the incoming DID of 6044, so was programatically correct at Goto(afterhours,6044,1) Plus, the GotoifTime's should work. Actually, the afterhours should work, if I have an extension of 'afterhours' in the current context. I didn't notice this until you pointed it out... but it was correct in previous revisions... and I was tyring to test it during the week, when the previous Goto's should have taken precidence... and they _all_ failed. Any reason why the rest should give me 'invalid extension'? >>> exten => 6044,1,GotoifTime(08:30-11:59|mon-fri|*|*?officehours,s,1) >>> exten => 6044,3,GotoifTime(12:00-12:59|mon-fri|*|*?lunch,s,1) >>> exten => 6044,2,GotoifTime(13:00-16:29|mon-fri|*|*?officehours,s,1) >>> exten => 6044,4,Goto(afterhours,1) It shouldn't make a difference, but you are altering terminators half way through the line...i.e. try exten => 6044,1,GotoifTime(08:30-11:59|mon-fri|*|*?officehours|s|1) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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] Speech to Text Conversion
has anybody found anything which works for speech to text translation? Implementation being instead of (or as well as) vm wav file being sent in email, a text translation would accompany the wav file Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Website Design www.cirelle.net ProSpeed High Speed Dial-up - 5 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ 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] calling an iaxy
Thomas Niesel wrote:
Hallo rich allen
On Mon, 1 Nov 2004 12:30:43 -0900 you wrote:
iH
i have an IAXy which i can make calls from but am unable to call. when
i dial the extension assigned, i get the following from the console;
-- Executing Dial("SIP/5801-b665", "IAX2/[EMAIL PROTECTED]") in new
stack
-- Called [EMAIL PROTECTED]
-- Call accepted by 192.168.0.5 (format ULAW)
Nov 1 12:28:33 NOTICE[163850]: chan_iax2.c:5546 socket_read: Rejected
call to 192.168.0.5, format 0x4 incompatible with our capability
0xff03.
Hm, I'm not an expert on iaxY but it looks like that the codec is the
prob.
If both sides do not find a common codec the call will be rejected.
Try to call with alaw or gsm and see if it helps.
The IAXy doesn't do anything but ulaw and adpcm. I would stick with
ulaw for testing.
I would start looking in the iax.conf entry for the iaxy for the culpit.
B.
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Re: [Asterisk-Users] calling an iaxy
Hallo rich allen
On Mon, 1 Nov 2004 12:30:43 -0900 you wrote:
> iH
>
> i have an IAXy which i can make calls from but am unable to call. when
> i dial the extension assigned, i get the following from the console;
>
> -- Executing Dial("SIP/5801-b665", "IAX2/[EMAIL PROTECTED]") in new
> stack
> -- Called [EMAIL PROTECTED]
> -- Call accepted by 192.168.0.5 (format ULAW)
> Nov 1 12:28:33 NOTICE[163850]: chan_iax2.c:5546 socket_read: Rejected
> call to 192.168.0.5, format 0x4 incompatible with our capability
> 0xff03.
Hm, I'm not an expert on iaxY but it looks like that the codec is the
prob.
If both sides do not find a common codec the call will be rejected.
Try to call with alaw or gsm and see if it helps.
> -- Hungup 'IAX2/5899/1'
>== No one is available to answer at this time
>
>
> what might i be missing?
>
> thanks
> - hcir
>
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--
Tho/\/\as
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Re: [Asterisk-Users] Centrex
It shouldn't matter if the inbound line is centrex, it's just a phone line
ringing into asterisk at that point. For the outbound side of the equation,
you'll have to dial 9 (or other digit as defined by the system) get dialtone
and send the dialed digits.
You might try:
exten => _9XX,1,dial(zap/g3/${EXTEN})
replace zap/g3 with whatever you're using to dial outbound. This should put
you on the track to figuring out how to get centrex outbound working.
- Original Message -
From: "Tim Thompson" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<[EMAIL PROTECTED]>
Sent: Monday, November 01, 2004 3:13 PM
Subject: RE: [Asterisk-Users] Centrex
> Centrex is a type of line and I do not believe there is a compatible card
> for *.
>
> FXS isn't going to cut it. Centrex is a digital type line.
>
> Tim.
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer
> Sent: Monday, November 01, 2004 1:35 PM
> To: Asterisk Users
> Subject: [Asterisk-Users] Centrex
>
> OK folks,
> I'm trying to help get another remote Asterisk box up and running.
> The system currently has a single FXO card, but it doesn't seem to
> be working, my guess is because the inbound line is CENTREX. Knowing
> nothing about Centrex, can someone tell me if I'm right, and need an
> FXS card?
>
> Thanks,
> Tim
>
> --
> ><
> >> Tim Sailer >< Coastal Internet, Inc. <<
> >> Network and Systems Operations >< PO Box 726 <<
> >> http://www.buoy.com >< Moriches, NY 11955 <<
> >> [EMAIL PROTECTED] >< (631) 399-2910 (888) 924-3728 <<
> ><
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Re: [Asterisk-Users] UDP Fragmentation Problem
> As far as I am aware there is no such thing as a fragmented UDP > packet; each packet is sent out on its own, there is no coherency > between UDP packets like there is with TCP packets. > > I could be very wrong here, it's been a late night with the kids. :-) > > >>> > >>>Packet fragmentation is at the IP layer, so UDP will have fragmented > >>>packets too. But... the OS should handle that and Asterisk shouldn't > >>>find out - it's a all or none policy, so it should receive the whole > >>>packet at once or nothing. > >> > >>How I can setup Linux to handle UDP fragments? > > > > > > Not sure why the concern with fragmentation, it should not be an issue > > with any modern linux distribution and there is nothing to setup. > > > > The only issue that I've heard about in recent months/years relative > > to fragmentation is the SonicWall firewall just can't seem to get it > > right. In their case, any udp packet greater then about 1500 bytes does > > not get reassembled propery, and its still an issue in the latest firmware. > > > > If you really think you've got a fragmentation problem, I'd like to see > > a packet trace (eg, ethereal) of those packets. > > > > Here it is ;-) Okay, looked at the pcap and see the fragmentation, but that does not indicate your asterisk IP stack is not handling it properly. Might compare a 'sip debug' with those packets to see if data is reassembled. Since both pieces of the original fragmented packet did in fact arrive at your destination, the only issue left is whether your IP stack reassembled them properly. I'd suspect another problem is lurking unrelated to fragmentation. Rich ___ 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: How far is IAX to be a Standard
Unless SIP just plain does not work, I think it'll be hard (for IAX to get excellent acceptance), without a lot of good marketing and other efforts by Digium. At VON, only a few people even understood what Asterisk was, let alone even had heard of IAX. Even with IT people controlling things you still see a lack of good decisions. How many people do SSL POP3? How many people use S/MIME? All very simple things that don't require reworking of dedicated hardware or huge industry efforts. My only point is that we can't just rely on a better design to somehow magically win out. Getting Digium to create a standard with input from other vendors would be a huge plus and help pave the path forward. -Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, November 01, 2004 12:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: How far is IAX to be a Standard I see your point and it is well taken but I feel that with convergence you are going to see more IT staff in charge of phone systems. In turn, I see more research and informed decisions going on, not just a consumer following what is being pushed. That being said, maybe I am just looking at a grey sky through blue sunglasses. > >From: [EMAIL PROTECTED] > [mailto:asterisk-users->[EMAIL PROTECTED] On Behalf Of Steve > Totaro > >>I predict a paradigm shift, rendering what is historical, null. > > Is this in general, applying to all technology, or just to telecom? (i.e., > will Hollywood ship videos in hi-def on EVD?) > > -Michael > >> >No worries here. What works best will win out eventually. >> >> Not sure where you get that idea, as historically it's not that way :S. >> >> Companies will make SIP work reasonably enough. What will win out is >> whatever is marketed and sold the best. Getting published specs, inc. >> being >> a published "standard" is part of that marketing. >> >> -Michael >> >> - Original Message - >> From: "Randy Bush" <[EMAIL PROTECTED]> >> To: "Voip Business" <[EMAIL PROTECTED]> >> Cc: <[EMAIL PROTECTED]> >> Sent: Monday, November 01, 2004 12:44 PM >> Subject: [Asterisk-Users] Re: How far is IAX to be a Standard >> >> what does the RFC's guys and the Pseudo-Cisco IETF think about this Protocol? >>> >>> the internet vendor task force has a massive amount invested in >>> sip. so there will be a lot of 'guidance' to have it published >>> as an informational rfc. if iax catches on in the market, then >>> they'll have to play. otherwise, expect to have a hard time >>> getting iax on the ivtf standards track. >>> >>> randy >>> >>> ___ 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] Eezee phone?
On Monday 01 November 2004 18:35, Kanuri, Seshu (Company IT) wrote: > They work wonderful on SIP and H323 and hence > can extend the life of H323 Termination gateways like AS5300 etc. Hi, Just to point out that AS5300 will terminate SIP natively if you upgrade your IOS version to something recent. Cisco did't mention when they'll include support for IAX ;o) ___ 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] adding an artificial delay to *
*blink blink* --On Monday, November 01, 2004 18:33 +0400 Vahan Yerkanian <[EMAIL PROTECTED]> wrote: Greetings, Is there a way to add artificial delay to the rtp stream? Due to regulations in our country, it is required to add 400ms delay to *some* VoIP calls. What country is that BTW? Just about the wierdest, and certainly one of the daftest regulations I've ever heard of. Is this possible with any module? ___ 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] Directory app and extension
On Mon, 01 Nov 2004 16:21:46 -0500, David Filion <[EMAIL PROTECTED]> wrote: > > So, the question is does anyone know of a way to get the extension > number when the dial plan context is entered via Directory(), and if so how? > > David Filion David, You could always right your own directory application (AGI if you want) that reads from Postgres. It would be pretty easy. -Chuji ___ 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] adding an artificial delay to *
Vahan Yerkanian wrote: Greetings, Is there a way to add artificial delay to the rtp stream? Due to regulations in our country, it is required to add 400ms delay to *some* VoIP calls. Is this possible with any module? Sorry I don't know if it is possible with a module (none that I know of), but you could simply route your calls via New Zealand, and then through the UK and then to US to terminate to destination. This all begs the question [EMAIL PROTECTED] Why do you have regulations in your country requiring you to make VOIP crap? Government owned telco? Which country? BTW: Depending on your volume of calls, I might be able to offer you the New Zealand leg of your crazy journey! :-) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ 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 phone auth
On Mon, 1 Nov 2004 17:50:35 -0300, "Guido Rebert" <[EMAIL PROTECTED]> wrote:
>Do anyone know about soft phones, commercials or not, that asks for
>username/passw at launch?
It would be fairly easy to patch iaxComm to do so.
>
>Guido Rebert
>Network Manager
>GrupoPyD - +54 11 4800
>
>
>-Mensaje original-
>De: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] En nombre de Guido Rebert
>Enviado el: Lunes, 01 de Noviembre de 2004 05:18 p.m.
>Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>Asunto: RE: [Asterisk-Users] EIC & *
>
>
>As I suposed... You have there a pretty nice solution.
>
>The T100p has an "ethernet" port I think your Dialogic´s CIC server board
>also has an "ethernet" port... As mine dialogic is coaxial like...
>
>You have over there a nice stuff!
>
>Any idea with exchange/crm? Asterisk talking with eic/cic...?
>
>Guido Rebert
>Network Manager
>GrupoPyD - +54 11 4800
>
>
>-Mensaje original-
>De: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] En nombre de Tom Neville
>Enviado el: Lunes, 01 de Noviembre de 2004 01:51 p.m.
>Para: Asterisk Users Mailing List - Non-Commercial Discussion
>Asunto: Re: [Asterisk-Users] EIC & *
>
>
>We're doing just that. :) I was going to reply off-line, but this
>should apply to just about any PBX that has PRIs going in and
>T1/Channel banks coming out.
>
>We've got a CIC system with 4 PRI ports and 10 Station ports (T1s going
>to Adtran TA-750s.) Our current system is built on ISA cards. The CIC
>VOIP boards are PCI, so we would have to replace all the hardware.
>(Way too expensive.)
>
>What we have working right now is a T1 (normally connected to an Adtra
>TA-750) going into a T100P. This is used to bring "stations" into the
>* box. Each channel corresponds to a specific user.. ie
>station-cb10-01 in the CIC world is me. Channel 1 is me in the *
>world. Using zapata.conf I map each channel to a context..
>
>zapata.conf
>---
>context = exttomn
>channel => 1
>
>context = extdela
>channel => 2
>
>Those contexts then ring whatever VOIP lines we want:
>
>extensions.conf
>---
>[exttomn]
>exten => s,1,Dial(SIP/7001&SIP/[EMAIL PROTECTED])
>
>[fromtomn]
>exten => _NXX,1,Dial(Zap/1/${EXTEN})
>exten => _NXXNXX,1,Dial(Zap/1/${EXTEN});
>exten => _1NXXNXX,1,Dial(Zap/1/${EXTEN});
>
>exten => _9,1,Dial(Zap/1)
>
>To place outbound calls, calls from the SIP phones are mapped into
>[fromUSERNAME] contexts. This causes outbound calls from these
>extensions to be placed through the users normal station. (The _9 is
>to allow the user to dial 9 to pickup the channel without actually
>dialing anything, incase you hit PICKUP in the CIC client by mistake.
>:) If the user is not there, the dial to the station is allowed to
>just timeout and CIC pulls the call back and drops them in the CIC
>voicemail (Unified messaging stuff..) This allows me to stay
>"Available, no ACD" to get my calls bounced to all of my phones...
>
>As for inbound, we ran a PRI into a T100P with PRI_NET signaling.
>(Providing a PRI to the Interactive box.) Through that, I am able to
>take VOIP inbound calls (eventually routing them from an AS5400 also
>providing dialup in other cities) into the CIC box. CallerID is
>reported correctly to the CIC box and inbound DID also works as it
>should.
>
>I then created a new TrunkGroups in CIC and put the * lines into that.
>(Our other lines are also grouped in TrunkGroups.) This allows me to
>route outbound long distance out the PRI to the * box and on to
>whatever VOIP provider we're using that day. :)
>
>Tom
>
>
>On Nov 1, 2004, at 10:56 AM, Guido Rebert wrote:
>
>> Has anyone done some integration between Interactive Intelligence
>> (cic,
>> eic...) and Asterisk?
>> Thanks all
>> Guido
>>
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Re: [Asterisk-Users] Centrex
On Mon, 1 Nov 2004 15:34:47 -0500, Tim Sailer <[EMAIL PROTECTED]> wrote: > OK folks, > I'm trying to help get another remote Asterisk box up and running. > The system currently has a single FXO card, but it doesn't seem to > be working, my guess is because the inbound line is CENTREX. Knowing > nothing about Centrex, can someone tell me if I'm right, and need an > FXS card? Tim, Have you messed around with your txgain & rx gain in zapata.conf? Also, make sure your distinictive ring isn't enabled... I'm betting it's a configuration issue. -Bryan ___ 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] Centrex
At 03:13 PM 11/1/2004, you wrote: Centrex is a type of line and I do not believe there is a compatible card for *. FXS isn't going to cut it. Centrex is a digital type line. Tim. Not true. Centrex can be delivered over any type of switched circuit. I have had it on both POTs and ISDN. It is a virtual PBX hosted at the CO. Tom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Sailer Sent: Monday, November 01, 2004 1:35 PM To: Asterisk Users Subject: [Asterisk-Users] Centrex OK folks, I'm trying to help get another remote Asterisk box up and running. The system currently has a single FXO card, but it doesn't seem to be working, my guess is because the inbound line is CENTREX. Knowing nothing about Centrex, can someone tell me if I'm right, and need an FXS card? Thanks, Tim -- >< >> Tim Sailer >< Coastal Internet, Inc. << >> Network and Systems Operations >< PO Box 726 << >> http://www.buoy.com >< Moriches, NY 11955 << ___ 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] Unable to write frame to channel: Success - MeetMeproblem
Nov 1 16:54:05 NOTICE[12238881]: Call failed to go through, reason 0 Nov 1 16:54:05 NOTICE[12009491]: Unable to request channel SIP/lbarr_page Nov 1 16:54:05 NOTICE[12009491]: Call failed to go through, reason 0 Nov 1 16:54:05 NOTICE[12304421]: Unable to request channel SIP/noclobby1_page Nov 1 16:54:05 NOTICE[12304421]: Call failed to go through, reason 0 Nov 1 16:54:08 NOTICE[12042261]: Call failed to go through, reason 3 Nov 1 16:54:09 NOTICE[12353576]: Call failed to go through, reason 3 Nov 1 16:54:11 NOTICE[1496]: Call completed to SIP/drodden_page Starting to get a little frustrated here; several users have called and screamed at me. This paging system used to work. When I page, everybody's phone answers but everybody gets dead silence, the conference room is completely broken. It's disappointing really. I'd like to think it's the zaprtc timing, but it's worked flawlessly in the past, nothing has changed except zaptel/libpri/asterisk I didn't change the kernel, or the way the module was loaded or the rtcsetup program running in the background. I may be forced to downgrade but I don't know to what version. Sigh. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Rodan Sent: Monday, November 01, 2004 3:30 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Unable to write frame to channel: Success - MeetMeproblem After I upgraded from the CVS Head to the latest CVS 1.0 Stable, my paging application doesnt work. Using the Wiki as a guidance, I made a line 2 on all phones with auto answer. When someone wants to page out, they dial an extension and it brings everyone into the conference, with everyone muted. This system used to work flawless, but now when I use the extension, it brings everybody into the conference without a problem, but it's silent, until the timeout kicks in. Everybody can't hear the person speaking, the who initiated the conference, its broken. My log files show: Nov 1 15:15:22 NOTICE[8716311]: Call failed to go through, reason 3 Nov 1 15:15:22 NOTICE[8831023]: Call failed to go through, reason 3 Nov 1 15:15:23 NOTICE[8486946]: Call completed to SIP/rkrisel_page Nov 1 15:15:24 WARNING[8339481]: Unable to write frame to channel: Success Nov 1 15:15:24 WARNING[8519716]: Unable to write frame to channel: Success Nov 1 15:15:24 WARNING[8323096]: Unable to write frame to channel: Success Any ideas? My timing is provided by zaptelrtc (the zaprtc module and the rtcsetup binary). This is how its been for some time. Ive recompiled a and reinstalled Asterisk/Zaptel/LibPRI/ZaptelRTC to no avail. >From my extensions.conf file: --- [paging] exten => *,1,AbsoluteTimeout(15) exten => *,2,agi(pageall) exten => *,3,MeetMe(,xdqp) exten => *,4,Hangup [add-to-paging] exten => start,1,AbsoluteTimeout(15) exten => start,2,MeetMe(,dmqp) exten => start,3,Hangup exten => h,1,Hangup exten => t,1,Hangup exten => T,1,Hangup --- One of my .call files: --- Channel: SIP/rkrisel_page Context: add-to-wupaging Extension: start Priority: 1 CallerID: Office Pager <> WaitTime: 3 --- And what I have in meetme.conf --- conf => --- And what I have in my agi script pageall: --- #!/bin/sh /bin/cp /var/lib/asterisk/paging/*.call /var/spool/asterisk/outgoing --- ___ 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] UDP Fragmentation Problem
Rich Adamson schrieb: I've got no success to get a friend in Bogota (Colombia) connected to my Asterisk. He has got a ISDN Internet connection and the UDP packets will be fragmented. It seems that the MTU of this connection is round about 400 to 500 Bytes. Therefore most UDP-SIP packages are fragmented. Is Asterisk not able to handle fragmented UDP packages? Is it possible to use SIP over TCP with X-Lite? Or has somebody another hint for me? As far as I am aware there is no such thing as a fragmented UDP packet; each packet is sent out on its own, there is no coherency between UDP packets like there is with TCP packets. I could be very wrong here, it's been a late night with the kids. :-) Packet fragmentation is at the IP layer, so UDP will have fragmented packets too. But... the OS should handle that and Asterisk shouldn't find out - it's a all or none policy, so it should receive the whole packet at once or nothing. How I can setup Linux to handle UDP fragments? Not sure why the concern with fragmentation, it should not be an issue with any modern linux distribution and there is nothing to setup. The only issue that I've heard about in recent months/years relative to fragmentation is the SonicWall firewall just can't seem to get it right. In their case, any udp packet greater then about 1500 bytes does not get reassembled propery, and its still an issue in the latest firmware. If you really think you've got a fragmentation problem, I'd like to see a packet trace (eg, ethereal) of those packets. Here it is ;-) Regards Bastian javier-sip-1.pcap Description: Binary data ___ 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] 'Unregistered Channel Type' when parsing zapata. conf on * startup
-Using FC2 -Digium T100P card -lsmod | grep wcfxo yields: wcfxo 16288 0 zaptel 232580 3 wct1xxp,wcfxo,wcfxs so my drivers are cool -zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 -zapata.conf: [channels] language=en channel =>1-23 signalling=cpe_pri asterisk -vvvc yields: [chan_zap.so] [snip] Unregistered channel type 'Zap' Oucherror while writing audio data: :Broken pipe Found a few examples relating to this on the mailing lists but not for a T1 card, and almost all of them pointed to a broken / misconfigured driver install. Can anyone help? TIA ___ 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] Centrex
On November 1, 2004 04:13 pm, Tim Thompson wrote: > Centrex is a type of line and I do not believe there is a compatible card > for *. > > FXS isn't going to cut it. Centrex is a digital type line. Not around here (Canada) -- Centrex is POTS but the PBX functionality is all "hosted" with the telco. -A. ___ 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] Passing a PIN in SIP Parameters
Hi, We have a system that takes an e164 number and does a translation to convert that e164 number to the PIN that is tied to a certain account. We configure customer's devices to use those PINs so they don't ever have to use a PIN. They just plug the device into the internet, and it works. My question is this: If I pass the e164 number as a username, password or some other field in the SIP parameters, can I retrieve it from a global variable in extensions.conf ? -- -- Heritage Communications Corporation Melbourne, FL USA 32935 http://www.hcc.net ___ 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] ZTdummy
centrex can be digital or analog. if you can plug a standard 2500 set into the line and get dial tone, then it is analog and will work just like any other POTS line (* would need an FXO card). you may or may not need to dial an access digit. 9 is the normal access digit but it may be anything or none at all (assume dial 9) - hcir On Nov 1, 2004, at 12:20 PM, Tomas Carnecky wrote: Paul Rodan wrote: It's picky about what USB controller you have. It refused to work on my server because it had the wrong kind of USB controller, go figure. So I used zaprtc and it worked fine. If you have problems with zrdummy, let me know and I'll see if I can help with zaptelrtc. The trickiest part is to make sure you don't have the Real Time Clock (rtc) compiled in the kernel. ___ 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] ASTCC - Anyone has a Dial Plan that is working?
My ASTCC is working, in the sense that it gives me a voiceprompt when I dial the extension configured for the ASTCC by saying "welcome" and "12 digit card number" but it is not asking me for the prompts to Enter Destination Number etc. Can anyone in the list send me the dialplan in sequence so that My users will be asked to enter the password, then the Destination number and after they enter the destination number, the uesrs will be connected to the destination. I need a couple of variations, if anyone can share them with me. Also if anyone has done some customization of the ASTCC or ASTPP I would like to get these files as wel. Thanks in advance Seshu Kanuri NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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] calling an iaxy
iH
i have an IAXy which i can make calls from but am unable to call. when
i dial the extension assigned, i get the following from the console;
-- Executing Dial("SIP/5801-b665", "IAX2/[EMAIL PROTECTED]") in new
stack
-- Called [EMAIL PROTECTED]
-- Call accepted by 192.168.0.5 (format ULAW)
Nov 1 12:28:33 NOTICE[163850]: chan_iax2.c:5546 socket_read: Rejected
call to 192.168.0.5, format 0x4 incompatible with our capability
0xff03.
-- Hungup 'IAX2/5899/1'
== No one is available to answer at this time
what might i be missing?
thanks
- hcir
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