[Asterisk-Users] Extension follow me
We are going to have people in our office who do not sit at the same desk throughout the day (or week), and have Cisco 7940 phones using the SIP image. Is it possible to easily set up the phone so that they can enter their extension number and password on the phone, and thus have their extension follow them ? I know that you can change the sip settings, but that requires admin capabilities. I know that this is similar to follow me, but person A might be at B's desk, while B is at A's desk, so I can't use call forwarding, and I don't want to ring x number of extensions in order to find the person. I really want to find the extension Julian. ___ 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] Over 10,000 lines. Will asterisk manage?
Hi all, I am to come up with a proposal to setup a network of over 15,000 lines. I would like to scale down the costs by using Asterisk as the main switching equipment. Let me give u the full scenario. 1. Fiber optic cables are to run from the central exchange to over 2 kilometer radius at selected distribution points. 2. Every subscriber will have a CAT5 cable terminating at his residence/office. This will provide both Internet/Voice and maybe video to the subscriber. 3. SIP phones will be used by the clients, codec U-Law. Bandwidth is no problem since the fiber network will provide over 10Gbit. 4. Fiber will run to the main Telecommunication provider(PSTN) and 2 mobile providers. Questions are which media protocol should I use? How many asterisk servers will I need? Are SIP phones/IAX phones reliable for this kind of project and are they available in such quantities? How many simultaneous calls can I achieve if no transcoding is being done? Keep in mind that their is no need for T1/PRI or any other type of external lines. Asterisk is to switch the voice data only. I believe asterisk will be able to handle this without a problem and its the way forward for a country which is ages back in telecommunications. The client has been approached to buy a switching equipment that can handle the stated amount of lines for a figure of $500,000. Asterisk can definately beat that. Jafar __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.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] Extension follow me
Julian wrote: We are going to have people in our office who do not sit at the same desk throughout the day (or week), and have Cisco 7940 phones using the SIP image. [..] I really want to find the extension Isn't this a case for Queues with callback login? Just a thought rgds pos ___ 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] Extension follow me
As usual, you sit for hours thinking of how to implement something, and send
an email asking for help. Seconds later, you think of a potential solution:
Thinking that the extension is a user, not a phone:
Admin:
1) Record is created with a 4 digit UserID
2) Context to use is stored against this UserID
User Login:
1) The user dials **1#, and enters their 4 digit UserID
2) * looks up a record from the database using the UserID as the key (with
some graceful error handling)
3) * then stores the ${EXTEN} variable as a value against the UserID
Usage (for calls to the User):
1) When someone makes a call to the user, they dial the 4 digit UserID
2) * looks up a record from the database using the UserID as the key (with
some graceful error handling)
3) * gets the current ${EXTEN} variable from the record
4) * dials the ${EXTEN}
Usage (for calls from the user):
1) User dials a number
2) * looks up a record from the database using the ${CALLERIDNUM} as the key
(with some graceful error handling)
3) * gotos to the context defined by the admin
4) * makes the call as normal
User Logout:
1) User dials **1##
2) * looks up a record from the database using the UserID as the key (with
some graceful error handling)
3) * sets the extension to (and thus also indicating that the person
is not available)
This allows anyone to call (for example 5711) and find me, no matter where I
am
This allows for the person to be controlled by the dial plan, not the
extension.
For example, the standard way of checking access permissions is by the
context (SIP/5711 is allowed to dial local but not international). However,
I could go to phone SIP/6712 (on my manager's desk) and call an
international number). With the above method, your dial plan restrictions
follow you.
It also allows for * to see if you are available or not - if not it can
divert straight to voicemail.
Just a straight out of my head idea. Probably full of holes. I would
appreciate any comments.
Julian
- Original Message -
From: Asterisk [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, November 13, 2004 8:09 AM
Subject: [Asterisk-Users] Extension follow me
We are going to have people in our office who do not sit at the same desk
throughout the day (or week), and have Cisco 7940 phones using the SIP
image.
Is it possible to easily set up the phone so that they can enter their
extension number and password on the phone, and thus have their extension
follow them ? I know that you can change the sip settings, but that
requires
admin capabilities.
I know that this is similar to follow me, but person A might be at B's
desk,
while B is at A's desk, so I can't use call forwarding, and I don't want
to
ring x number of extensions in order to find the person.
I really want to find the extension
Julian.
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Re: [Asterisk-Users] Extension follow me
Oh! Man! The simplest solution. Now I feel really stupid. That may well solve the follow me issue. Julian - Original Message - From: Peer Oliver Schmidt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Saturday, November 13, 2004 8:30 AM Subject: Re: [Asterisk-Users] Extension follow me Julian wrote: We are going to have people in our office who do not sit at the same desk throughout the day (or week), and have Cisco 7940 phones using the SIP image. [..] I really want to find the extension Isn't this a case for Queues with callback login? Just a thought rgds pos ___ 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] Over 10,000 lines. Will asterisk manage?
I'm confident asterisk can manage such a setup, but you will need a damn good consultant to set it up. :) (You cannot buy just a huge asterisk machine, you will need some kind of cluster to do this). Joachim (zoa) jafar mohammed wrote: Hi all, I am to come up with a proposal to setup a network of over 15,000 lines. I would like to scale down the costs by using Asterisk as the main switching equipment. Let me give u the full scenario. 1. Fiber optic cables are to run from the central exchange to over 2 kilometer radius at selected distribution points. 2. Every subscriber will have a CAT5 cable terminating at his residence/office. This will provide both Internet/Voice and maybe video to the subscriber. 3. SIP phones will be used by the clients, codec U-Law. Bandwidth is no problem since the fiber network will provide over 10Gbit. 4. Fiber will run to the main Telecommunication provider(PSTN) and 2 mobile providers. Questions are which media protocol should I use? How many asterisk servers will I need? Are SIP phones/IAX phones reliable for this kind of project and are they available in such quantities? How many simultaneous calls can I achieve if no transcoding is being done? Keep in mind that their is no need for T1/PRI or any other type of external lines. Asterisk is to switch the voice data only. I believe asterisk will be able to handle this without a problem and its the way forward for a country which is ages back in telecommunications. The client has been approached to buy a switching equipment that can handle the stated amount of lines for a figure of $500,000. Asterisk can definately beat that. Jafar __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.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] Over 10,000 lines. Will asterisk manage?
--On Saturday, November 13, 2004 00:11 -0800 jafar mohammed [EMAIL PROTECTED] wrote: Hi all, I am to come up with a proposal to setup a network of over 15,000 lines. I would like to scale down the costs by using Asterisk as the main switching equipment. Let me give u the full scenario. I have to agree with another person who said you'll need a decent consultant to set it up...and software to manage it. As for that sort of quantity of SIP devices, the only ones I know you'd be able to get for sure in that quantity would be Cisco. 79XXs or the ATA's. Motorola ATA's are also an option. Outside of that I don't know personally. keeping it all uLaw is probably good in this situation also because transcoding is a pretty heavy hit on the Asterisk server CPU. Without transcoding I think having more than 500 *active* sessions per box should be easy, probably hit a couple thousand even, but that'd have to be tested. With SIP devices it's not the number of devices - well, mostly, at some point the number of registration requests becomes an issue - but the number of active conversations in the system. You can run a virtually unlimited number of SIP clients on a single box, but they couldn't all talk at once, unless you wanted a Chernobyl style melt-down. Probably be about $100 or $200k in PC or other hardware. Keep in mind that you have to have something with a USB controller for 2.4 kernels to source your timing off of, or in 2.6 it can use the RTC I believe. In a pure IP environment you might be better off going to 2.6 anyway. Just my $0.02 ___ 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] Over 10,000 lines. Will asterisk manage?
With a bit of money and hard work - many things are possible. Brandon ___ 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] SPA-3000 Wizard for Asterisk
For your testing pleasure. Feedback welcome: http://voxilla.com/spa3kasterisk.php -- PhoneBoy ___ 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] Over 10,000 lines. Will asterisk manage?
4. Fiber will run to the main Telecommunication provider(PSTN) and 2 mobile providers. [snip] Keep in mind that their is no need for T1/PRI or any other type of external lines. Asterisk is to switch the voice data only. How are you linking to the PSTN referenced in (4) above then? How many concurrent calls have to go to the PSTN? Linus ___ 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: BRI in the US
Hi Brian, One goal is to get BRI support in Zaptel if possible. I'm right now in the planning stage :P Plus BRI is much cooler than pots. Why invent the wheel again, what's wrong with bristuff from junghanns.net? bye, aa _ Listen to music online with the Xtra Broadband Channel http://xtra.co.nz/broadband ___ 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] Cisco IP phones, SIP, Call-Manager Contracts
Hi, There is a lot of talk about Cisco phones, SIP firmware and Contracts to download same. Does using a 7940/60 or other with SIP firmware offer better features/compatibility with Asterisk over using the [default?] Call-Manager firmware and chan_sccp? A lot of people here must have started with Call-Manager then moved on, with all the work that entails, and installed the SIP firmware - I'd love to hear someone's opinion of the difference in using the phones before after. Thanks, Derek --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.778 / Virus Database: 525 - Release Date: 15/10/2004 ___ 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] Cisco IP phones, SIP, Call-Manager Contracts
Hell yes!!! The SIP firmware offers so much more and is better supported with * On Sat, 2004-11-13 at 06:58, Derek Conniffe wrote: Hi, There is a lot of talk about Cisco phones, SIP firmware and Contracts to download same. Does using a 7940/60 or other with SIP firmware offer better features/compatibility with Asterisk over using the [default?] Call-Manager firmware and chan_sccp? A lot of people here must have started with Call-Manager then moved on, with all the work that entails, and installed the SIP firmware - I'd love to hear someone's opinion of the difference in using the phones before after. Thanks, Derek --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.778 / Virus Database: 525 - Release Date: 15/10/2004 ___ 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 -- 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: BRI in the US
Hi, One goal is to get BRI support in Zaptel if possible. I'm right now in the planning stage :P Plus BRI is much cooler than pots. Why invent the wheel again, what's wrong with bristuff from junghanns.net? US bri (afaik) is not EuroISDN, but NI or something like. funny mode Of course US people have their own standards : ulaw instead of alaw, NI instead of euroisdn, T1 instead of E1, miles instead of km and so on... :) /funny mode But since junghanns.net does already the cards (transport layer is the same for both, only layer-3 is different, afaik) perhaps adding to */libpri/zaptel euroisdn bri (from klaus) and us bri could be a great idea. is of course a bigger plus for * itself matteo -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ 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] SysMaster and GPL Violation
I must be missing something with the GPL... Nowhere does it say you need to advertise the open source product in your sales literature. All (from what I gather) is necessary, is to make available the source or instructions to retrieve the source to the end user. This could be on a CD or a sentence in a provided manual. No reference is given to the percent of open source to proprietary software which must be disclosed. Also, if I am not mistaken, if you sell a system with linux on it, don't you have to do the same for the OS? Lots of GPL stuff there. I am not defending sysmaster or anyone else, but I haven't seen (in this discussion) where anybody had reviewed the entire end user product package (outside of a hard drive). My 2cents Regards Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie question
First, I'm really new to asterisk and I'm testing it in order
to improve my first steps...
Recently I installed * asterisk on a FreeBSD Box (5.2.1)
I got it working on my internal LAN (it works fine !).
I was trying to connect my * box through FWD using SIP
but it is not working an I'm very confused about *, in fact
I can't call from my * client (X-Lite) to a FWD number,
but bettwen my * sip authenticated clients yes...
Please somebody can help me or guide me to the right direction ?
Any kind of help will be appreciated and excuseme by my english :o(
Here is attached my config files
There really isn't enough info in those config files to answer
your question. If the bsd box is behind a nat box, then your missing
the sip.conf statements to support that. If you are, consider using
iax2 for the link (instructions are on fwd's site).
Your exten = _7., statement doesn't look right either. Here's what
works on my system (substituting your userid/secret and using iax):
exten = _7.,1,Dial(IAX2/500460:[EMAIL PROTECTED]/${EXTEN:1},60,r)
Modify as necessary for sip.
Also, in your register statement:
register = 500460:[EMAIL PROTECTED]/500460
the /500460 at the end tells fwd to send that extn number when they dial
your * box. So you will need something in your extensions.conf file
that looks something like:
exten = 500460,1,Dial(what ever your want to do)
Without that, incoming calls from fwd have no where to go.
An alternate way to accomplish that is to drop the /500460 from that
register statement, and then have an inbound-fwd context something like:
[fwd]
exten = s,1,Dial(what ever you want to do)
To help troubleshoot your config, break the process down into diagnosing
outbound calling first followed by diagnosing inbound calls. Use the CLI
'sip debug' to identify whether your register statement is actually
working. Once that is successful, then do the same for an outbound call
to one of the fwd test numbers. Once that is successful, then use the
fwd web site option to initiate a test call from their site to your
* system.
Remember, the only thing the register statement does is to tell fwd
where to reach your machine (IP address). Your Dial statement is used
to send calls to fwd, but you still need an incoming 'context' for
inbound fwd sip calls.
Might take a look in the wiki and fwd's web site for * config examples.
http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD
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Re: [Asterisk-Users] SysMaster and GPL Violation
The reason these threads end up rambling on far too much is people post without reading anything pertinent in the previsious messages. SysMaster has been vehemently denying their systems are based on Asterisk, so they have *not* been making any source available, or telling customers where you might find it. However, their system does nothing whatsoever to disguise that is really is *. They seem far too lazy to do any actual work, so probably the code they use is * without any modifications. However, the licence requires they tell customers how to get obtain unmodified source code. Steve Cirelle Enterprises wrote: I must be missing something with the GPL... Nowhere does it say you need to advertise the open source product in your sales literature. All (from what I gather) is necessary, is to make available the source or instructions to retrieve the source to the end user. This could be on a CD or a sentence in a provided manual. No reference is given to the percent of open source to proprietary software which must be disclosed. Also, if I am not mistaken, if you sell a system with linux on it, don't you have to do the same for the OS? Lots of GPL stuff there. I am not defending sysmaster or anyone else, but I haven't seen (in this discussion) where anybody had reviewed the entire end user product package (outside of a hard drive). My 2cents Regards Greg ___ 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] SysMaster and GPL Violation
[snip] If someone believes that they are contributing software to a GPL'd software project, and does not realize that the nature of your disclaimer allows Digium to release their changes under a non-GPL'd license, then that is breaking with the spirit of the GPL. If that is true, then the GPL is not comprehensive enough to cover its own spirit, so what you are saying is that the GPL implementation is fundamentally flawed. The GPL is fundamentally flawed in that it's never been functionally tested and challenged in court, and many IP lawyers believe that there are challenges that it would not survive. The fact that some lawyers may have found further legal loopholes to exploit is not shocking, given the holes in the current implementation. If one can break the spirit of it without breaking it, then something is missing from it that should have there. Many people pay lawyers to find loopholes. I have no doubt that if a large company, of an IBM-like size, wanted to have the GPL found unenforceable, that there are numerous vectors on which to attack it. It is certain that the FSF did not have as many lawyers participating in drafting this license, and that the state of the art in software licenses 13 years ago (the most recent update of the GPL) was less sophisticated and less tested. I've seen good legal teams drive a truck through long legal documents that were considered to be thorough. I've seen courts throw out conservative legal documents for a variety of reasons. The GPL is both long and quite unusual as a legal document goes. To think that it has no attack vector is naive at best. On the other hand, if you are injecting some supernatural spirit (and purposely using that word to conjure imagery of the imaginary intangible qualities that can never be written on paper) of your own into what the GPL actually is, then the GPL is fine as written, which I suspect is the case. The GPL is what it says, and its spirit comes from what it says, and there is no way that anyone can break its spirit as such. Well, the GPL *is* an attempt to legally enforce GNU's concept of free software, which I refer to as the spirit of the GPL. We can be fairly certain that their concept did not translate verbatim into legal language, simply because few things ever translate 100%. Unless you are now claiming to be the author of the GPL, you should stop trying to be an expert on its spirit. The only ones qualified to do so are John Stallman and his attorneys, misguided though they may be. Who's John Stallman? Richard M. Stallman's brother? In the meantime, if you don't like the fact that I've been contemplating the GPL vs the BSD license vs other licenses for many years, that's fine. You do not need to consider me an expert... I don't consider myself one, after all. However, I do believe that I can safely discuss the philosophy of the GNU project at this level of detail without conflicting with their actual position. Yet no matter how much I don't care for the GPL, I find myself believing contributors who don't fully understand the disclaimer merely to be naive, but Digium looking a bit unscrupulous in this regard. Butter him up and then call him unscrupulous in a later paragraph. Beautifully manipulative. I said 'looking a bit unscrupulous'. How better to phrase it? There's something unusual going on. It isn't being disclosed in an obvious manner. People are signing away rights. If you'd read the GPL and the other stuff on the GNU web site, that's fairly clearly not in keeping with some of the principles behind the GPL. Manipulative? Who's being manipulative? I'm discussing the issue. If I've made a point, it's certainly not been by unfair means. That obviously won't fix the moral standing problem that the FSF would Your own use of quotes here suggests something interesting. I'll leave it to the reader to discover what. What, you're dissin' me for suggesting that Digium could at least disclose what's going on? Or are you dissin' me for what the FSF says about authors who release code under multiple licenses (which does not necessarily match up with my own philosophy on the whole matter)? Either way: Get lost. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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] Over 10,000 lines. Will asterisk manage?
On Saturday 13 November 2004 09:11, jafar mohammed wrote: I am to come up with a proposal to setup a network of over 15,000 lines. I would like to scale down the costs by using Asterisk as the main switching equipment. Let me give u the full scenario. Wow, you gained 5000 lines between typing the subject and the body of this email :o) Now *that's* exponential growth! Anyway, I am but a newby, but if i understand correctly, the situation is thus: A SIP phone *could* normally send its media stream directly from phone to phone, if no transcoding is required, but when using Asterisk the media stream will always pass through the server, causing a pottential bottleneck. So, why not use SER to register all the SIP phones, as it doesn't handle the media-streams, just keeps track of the phones and does the 'handshake'. SER is supposed to be able to handle over 50.000 calls at a time, so one SER server would be enough. Then interface this with one (or more) Asterisk servers to connect to the local PSTN. But maybe I'm missing something fundamental, in which case I'm happy to learn. Cheers, Richard ___ 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 outside number along side otherinternal extensions?
On Fri, 12 Nov 2004, Paul Fielding wrote: Hmmm... Interesting that you mention it's not a problem with VOIP companies as they use PRI. The analog trunk I'm connecting to is actually a Vonage line. Thing is, it seems to me that by connecting via the Zap channel to the Vonage ATA I'm effectively cancelling any advantage that Vonage's PRI might have... (?). I don't believe I have any other alternatives for connecting to Vonage's service, but perhaps I'm wrong about that. yeah... well, Vonage probably does use PRI.. in their office (yay, that really helps at your end!). It's rather unfortunate that they insist people convert back to the analog domain to use their service. One option available to you is to buy the softphone option on your account. In the archives for this list, within the last two months or so, you'll find config examples of how to get Asterisk to connect to vonage with a SIP channel on the softphone account. That would make your current goal easier, probably, but more expensive too. Greg ___ 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] SysMaster and GPL Violation
On Sat, 2004-11-13 at 06:03, Joe Greco wrote: [SNIP] However, the specific item that stopped me was the second paragraph of the short disclaimer, because our lawyers would never allow signing of a blanket statement such as and will do nothing to undermine it in the future. (As it was, the remainder of that paragraph would have had to have been sent off to the lawyers, as I don't really have a grasp on how much legal territory that might cover). That sent me off to look at the long disclaimer, at which point it eventually became apparent what you were actually trying to accomplish. [SNIP] So you have read both disclaimers. Yet you state: [SNIP] Digium is making people sign a draconian agreement that gives up rights to patches and features that are integrated into Asterisk, by signing rights over to Digium. [SNIP] Which is definatly the wrong way to see it, because the first disclaimer says, that you disclaim all rights, but not that you pass them over to Digium. In fact you make your source public domain. You didn't read the second paragraph, did you. If making changes public domain was all that would be required, there would be no need for that second paragraph. Or, for that matter, for the first. Merely placing This source code is in the public domain. within the code in question is sufficient for the purpose, though it may be easier for Digium to work that out out-of-band, in which case a first-paragraph-only agreement would make sense. Interestingly enough, placing something in the public domain is potentially riskier than providing it under either the BSD or GPL licenses, because both licenses provide a strong no-warranty clause. There are a number of competing theories on whether or not the author of a public domain bit of code could be liable, with varying amounts of case law, as I understand it: 1) One theory is that you may place code in the public domain, with explicit no-warranty disclaimers (this seems sensible to me). 2) Another theory says that such disclaimers are not legally binding, and that you would need to embed it within a license, copyright agreement, contract, or something like that to prohibit use of the code in the event that the recipient did not agree. 3) Another theory says that liability is only a concern where money has changed hands. There are apparently some finer-grained distinctions in there somewhere. I don't know if I'd want to submit major changes to a project and open myself up to the possibility of having to legally test whether or not a no-warranty clause on a public domain code contribution could be enforced. The second one does neither state, that you sign your copyrights over to Digium. It gives Digium a non-exclusive, non-revocable right to use your changes. That's it. I'd check with our IP lawyer if I really cared. However, it looks a bit more sweeping than that. Even though I'm not a lawyer, I can disprove your statement: if I sign this agreement *and don't even contribute anything*, but were to purchase ownership of a patent covering something that conflicts with Asterisk, this agreement grants Digium rights that you haven't acknowledged. See, that's the ugly thing about legal documents. There are endless things to consider. We can of course agree that it ought not work that way, but that's just pie-in-the-sky. Now, that's all well and fine, you obviously /can/ do it, but what most disturbs me is that people might sign the short form agreement without understanding exactly what it is that they're agreeing to. If someone believes that they are contributing software to a GPL'd software project, and does not realize that the nature of your disclaimer allows Digium to release their changes under a non-GPL'd license, then that is breaking with the spirit of the GPL. It has never been a hidden fact, that Digium runs Asterisk under a Dual License. Digiums Website (http://www.digium.com - Software Products) states: .. Digium specializes in the production of Open Source telecommunications software to accompany our hardware offerings. Most Digium software is licensed under GNU GPL, but may also be licensed commercially from Digium. And then a listing of software, including Asterisk. .. The README states: * LICENSING Asterisk is distributed under GNU General Public License. The GPL also must apply to all loadable modules as well, except as defined below. Digium, Inc. (formerly Linux Support Services) retains copyright to all of the core Asterisk system, and therefore can grant, at its sole discretion, the ability for companies, individuals, or organizations to create proprietary or Open Source (but non-GPL'd) modules which may be dynamically linked at runtime with the portions of Asterisk which fall under our copyright umbrella, or are distributed under more flexible licenses than GPL. .. Which contributor should not
Re: [Asterisk-Users] RE: BRI in the US
Hi Brian, One goal is to get BRI support in Zaptel if possible. I'm right now in the planning stage :P Plus BRI is much cooler than pots. Why invent the wheel again, what's wrong with bristuff from junghanns.net? US BRI is alien. It's not the same as BRI elsewhere. (sigh) ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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] SysMaster and GPL Violation
The reason these threads end up rambling on far too much is people post without reading anything pertinent in the previsious messages. SysMaster has been vehemently denying their systems are based on Asterisk, so they have *not* been making any source available, or telling customers where you might find it. However, their system does nothing whatsoever to disguise that is really is *. They seem far too lazy to do any actual work, so probably the code they use is * without any modifications. However, the licence requires they tell customers how to get obtain unmodified source code. In the event that Digium has not licensed Asterisk to them under a non-GPL license: No, there is no requirement that they tell customers how to obtain unmodified source code (assuming you mean generic Asterisk)... there is, however, a requirement that they offer up the source code that they used to build their Asterisk executables. If that happens to be the same as a distributed Asterisk version, then there is no functional difference, of course, and compliance with the GPL is pretty much as simple as offering people tarballs of that. But merely referring people to the Asterisk web site might not be sufficient compliance... I know the GPL people were debating that at one point. If they've made changes, however, then those changed sources must be made available. That might be interesting stuff, and is precisely the sort of crowbar that the GPL is intended to be. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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] CNG Comfort Noise Generation
Thank you for making this clear for me. Is there any solution for the mentioned phones? Assaf Benharoosh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Saturday, November 13, 2004 1:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CNG Comfort Noise Generation Hi Assaf, Assaf Benharoosh wrote: I have a problem with many phone such as BudgeTone, ariaVoice, PCPhoneline. They are not generating comfort noise (you can hear yourself when you're talking)- with budgetone having CNG sporadically. Is there a way to make this happen on Asterisk - or it must be a phone feature. Does anyone else experiencing this issue with those phones and have a workaround? Assaf Benharoosh Hearing yourself when you talk is not comfort noise. It is sidetone. Comfort noise is simulating the background noise of the room at the far end when nobody is talking and transmission has stopped. Sidetone is always a phone feature. Comfort noise usually is too. 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 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] Over 10,000 lines. Will asterisk manage?
So, why not use SER to register all the SIP phones, as it doesn't handle the media-streams, just keeps track of the phones and does the 'handshake'. SER is supposed to be able to handle over 50.000 calls at a time, so one SER server would be enough. Then interface this with one (or more) Asterisk servers to connect to the local PSTN. But maybe I'm missing something fundamental, in which case I'm happy to learn. I'm guessing, and I'd can't say for sure without seeing the actual physical layout of all of this, that the final solution would probably be a combination of SER and Asterisk with Asterisk getting used for endpoint connections and SER as a routing solution. There are really two virtual topologies that need to be considered to make such a judgment though. First, the actual network structure has to be finely analyzed. You need to know where your bottlenecks exist, latency issues within the network, and other such factors that could cause network issues. During the same time, its also probably a good idea to consider your potential network points of failure so you can plan on strategies should something go wrong. Second, you need to look at the virtual telephone exchange you are creating to understand how and where traffic is going to flow. In certain cases, you may want SIP devices talking to each other such as backend connections, but you really aren't going to want to have SIP endpoint devices doing this as 1) Some countries may and probably will start implementing wiretap requirements that will force you to redesign your entire network. 2) Accounting and control of devices is much harder when your devices are talking P2P. Just look at all the problems the RIAA has when trying to regulate P2P networks. 15,000 endpoints may sound like a lot, but realistically, never more than about 1/8 - 1/4 will be inuse at the same time depending on the environment. Realistically, I see this kind of size system being more of a network design issue than a VoIP one so the key is to make sure you have a good network engineer planning the network and knowing what that network is going to really get used for. Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 833-9720 ___ 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] Broadvoice Patch issues
Is anyone else experiencing a lot of busy signals after this patch? ie Broadvoice becomes disassociated with asterisk.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] wctdm to replaces wcfxs module ?
Hi, Am I correct in saying that the wcfxs kernel module is something of the past, and is now replaced by wctdm ? Regards, Thomas ___ 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] Over 10,000 lines. Will asterisk manage?
On Sat, 13 Nov 2004 10:37:27 -0500, Raymond McKay [EMAIL PROTECTED] wrote: So, why not use SER to register all the SIP phones, as it doesn't handle the media-streams, just keeps track of the phones and does the 'handshake'. SER is supposed to be able to handle over 50.000 calls at a time, so one SER server would be enough. Then interface this with one (or more) Asterisk servers to connect to the local PSTN. But maybe I'm missing something fundamental, in which case I'm happy to learn. I'm guessing, and I'd can't say for sure without seeing the actual physical layout of all of this, that the final solution would probably be a combination of SER and Asterisk with Asterisk getting used for endpoint connections and SER as a routing solution. There are really two virtual topologies that need to be considered to make such a judgment though. First, the actual network structure has to be finely analyzed. You need to know where your bottlenecks exist, latency issues within the network, and other such factors that could cause network issues. During the same time, its also probably a good idea to consider your potential network points of failure so you can plan on strategies should something go wrong. Second, you need to look at the virtual telephone exchange you are creating to understand how and where traffic is going to flow. In certain cases, you may want SIP devices talking to each other such as backend connections, but you really aren't going to want to have SIP endpoint devices doing this as 1) Some countries may and probably will start implementing wiretap requirements that will force you to redesign your entire network. 2) Accounting and control of devices is much harder when your devices are talking P2P. Just look at all the problems the RIAA has when trying to regulate P2P networks. 15,000 endpoints may sound like a lot, but realistically, never more than about 1/8 - 1/4 will be inuse at the same time depending on the environment. Realistically, I see this kind of size system being more of a network design issue than a VoIP one so the key is to make sure you have a good network engineer planning the network and knowing what that network is going to really get used for. Raymond McKay President RAYNET Technologies LLC http://www.raynettech.com (860) 833-9720 ___ 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 ** I'm using MAX TNT'S for PSTN inteface (8T1'S to DS3) It's a gateway so you can do TDM, SIP, ISP, ISDN, SS7 (limited) in one box. SER is the TANDEM, this keeps the audio out of the box. Asterisk is the END OFFICE with all of the class 5 type features, CDR, etc.users with Some users get Asterisk as well, especially for stuff like 911 on a single POTS line. Larger end users might get Asterisk with IAX trunking back to the end office. James Taylor Using Opera's revolutionary e-mail client: 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] New TA from Uniden
Has anybody tried out the new TA from Uniden? DTA200 ___ 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] Over 10,000 lines. Will asterisk manage?
A SIP phone *could* normally send its media stream directly from phone to phone, if no transcoding is required, but when using Asterisk the media stream will always pass through the server, causing a pottential bottleneck. So, why not use SER to register all the SIP phones, as it doesn't handle the media-streams, just keeps track of the phones and does the 'handshake'. SER is supposed to be able to handle over 50.000 calls at a time, so one SER server would be enough. Then interface this with one (or more) Asterisk servers to connect to the local PSTN. But maybe I'm missing something fundamental, in which case I'm happy to learn. Um, Wrong, You can do re-invites and have the media go point-to-point, We do it all the time. ___ 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] SysMaster and GPL Violation
I didn't see anything like that on their website... Is there something written somewhere that they say this? besides, I read all of these posts Greg - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Saturday, November 13, 2004 8:23 AM Subject: Re: [Asterisk-Users] SysMaster and GPL Violation | The reason these threads end up rambling on far too much is people post | without reading anything pertinent in the previsious messages. | | SysMaster has been vehemently denying their systems are based on | Asterisk, so they have *not* been making any source available, or | telling customers where you might find it. However, their system does | nothing whatsoever to disguise that is really is *. They seem far too | lazy to do any actual work, so probably the code they use is * without | any modifications. However, the licence requires they tell customers how | to get obtain unmodified source code. | | Steve | | | Cirelle Enterprises wrote: | | I must be missing something with the GPL... | | Nowhere does it say you need to advertise the open source | product in your sales literature. | | All (from what I gather) is necessary, is to make available | the source or instructions to retrieve the source to the end | user. | | This could be on a CD or a sentence in a provided manual. | | No reference is given to the percent of open source to proprietary | software which must be disclosed. | | Also, if I am not mistaken, if you sell a system with linux on it, | don't you have to do the same for the OS? Lots of GPL stuff | there. | | I am not defending sysmaster or anyone else, but I haven't | seen (in this discussion) where anybody had reviewed | the entire end user product package (outside of a hard drive). | | My 2cents | | Regards | Greg | | | | ___ | 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] wctdm to replaces wcfxs module ?
Thomas: Correct. Actually wctdm *is* wcfxs. They just renamed it. Robert Thomas Andrews wrote: Hi, Am I correct in saying that the wcfxs kernel module is something of the past, and is now replaced by wctdm ? Regards, Thomas ___ 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] Cable for T1 connection: Crossover or straight through?
This is my second asterisk server but the first one with a T100P card. I connected it to the phone company(SBC) jack but have only a busy signal when calling the T1's number and nothing in the asterisk log files to indicate a connection. Do I need to use a crossover cable? Thanks Malcolm ___ 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] SysMaster and GPL Violation
Citat Joe Greco [EMAIL PROTECTED]: [SNIP] There are a number of competing theories on whether or not the author of a public domain bit of code could be liable, with varying amounts of case law, as I understand it: 1) One theory is that you may place code in the public domain, with explicit no-warranty disclaimers (this seems sensible to me). 2) Another theory says that such disclaimers are not legally binding, and that you would need to embed it within a license, copyright agreement, contract, or something like that to prohibit use of the code in the event that the recipient did not agree. 3) Another theory says that liability is only a concern where money has changed hands. There are apparently some finer-grained distinctions in there somewhere. I don't know if I'd want to submit major changes to a project and open myself up to the possibility of having to legally test whether or not a no-warranty clause on a public domain code contribution could be enforced. [SNIP] I'm not in favour for Public Domain disclaims either, but that is something any contributor should take up with himself. [SNIP] The second one does neither state, that you sign your copyrights over to Digium. It gives Digium a non-exclusive, non-revocable right to use your changes. That's it. I'd check with our IP lawyer if I really cared. However, it looks a bit more sweeping than that. Even though I'm not a lawyer, I can disprove your statement: if I sign this agreement *and don't even contribute anything*, but were to purchase ownership of a patent covering something that conflicts with Asterisk, this agreement grants Digium rights that you haven't acknowledged. See, that's the ugly thing about legal documents. There are endless things to consider. We can of course agree that it ought not work that way, but that's just pie-in-the-sky. Section 4 is a bit fishy, i would agree on that, however it doesn't really apply outside the United States (yet, and hopefully never, at least not for Europe). Simply because software patents is a phaenomenon, that not exists in Europe (covering me) at this point. For me personally, if I would go down the road and get a software patent that would conflict with Asterisk, I still had no problem with the fact, that I couldn't make Digium pay. It is the price you pay for using and contributing, if you sign this disclaimer, the IP of Digium. But that is a personal choice. If I wanted to cover me from not giving Digium of any such patents, that I own, I should not contribute (but drag money out of them for my patent :o) ) or disclaim the patches i contribute to the public domain. Anyhow, where is the spirit ? You are advocating FSF and the GPL and taking in consideration, that you ever would have a software patent ? Because a hardware patent can hardly conflict with Asterisk. [SNIP] The one who's looked at the Asterisk web site, has gone to the bugs link, and is then confronted by the short form disclaimer, and doesn't really know or care about Digium, or that there was some requirement that s/he become intimately familiar with some evil company (and be aware that some GPL advocates view companies thusly) and all of the above. Or, to turn this around: What harm would there be in outlining this explicitly within the agreement itself? [SNIP] I do agree, that the Asterisk Website should state something about Asterisk License. I was surprised that there not even is mentioned, that it is GPL. The only mention of GPL is that it is based on a GPL copyrighted PRI stack. [SNIP] Only if they haven't read the essential Documentation (Read README) and the disclaimer before they sign it. I read the essential Documentation. A year ago. I've long since forgotten most of it, as would most people without a photographic memory. Surely, but if you contribute to a project, shouldn't you allways check the license ? Would LICENSE, COPYRIGHT or README be the first places to look, if not on the website ? Where i come back to: [SNIP] Digium is making people sign a draconian agreement that gives up rights to patches and features that are integrated into Asterisk, by signing rights over to Digium. [SNIP] How can you come up with such a claim, that has no base whatsoever ? All right, I concede that the rights aren't being signed over to Digium. That wasn't really the point, and was an error on my part. Please delete everything after that comma, which edits it into a claim that does have a base. I'm just trolling over this, becaue i've seen various people on various lists trying to convince people, that it is fact. Just a couple of days ago, when i was searching on the BlueZ lists, i found somebody who told people not to contribute to Asterisk, because you would sign over your copyright. This was posted in October, quite shocking. People just don't read the contents of Licenses and Disclaimers anymore. They
Re: [Asterisk-Users] SysMaster and GPL Violation
Citat Joe Greco [EMAIL PROTECTED]: [snip] If someone believes that they are contributing software to a GPL'd software project, and does not realize that the nature of your disclaimer allows Digium to release their changes under a non-GPL'd license, then that is breaking with the spirit of the GPL. If that is true, then the GPL is not comprehensive enough to cover its own spirit, so what you are saying is that the GPL implementation is fundamentally flawed. The GPL is fundamentally flawed in that it's never been functionally tested and challenged in court, and many IP lawyers believe that there are challenges that it would not survive. The fact that some lawyers may have found further legal loopholes to exploit is not shocking, given the holes in the current implementation. It has been tested in city/county court in Munich (Germany) and found valid (http://yro.slashdot.org/article.pl?sid=04/07/23/1558219tid=117), not that that might help anybody in the US, but it is a start. Kind regards, Martin List-Petersen ___ 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] wctdm to replaces wcfxs module ?
Citat Robert Lawrence [EMAIL PROTECTED]: Thomas Andrews wrote: Hi, Am I correct in saying that the wcfxs kernel module is something of the past, and is now replaced by wctdm ? Thomas: Correct. Actually wctdm *is* wcfxs. They just renamed it. It would be confusing to continue calling it wcfxs, since there are fxo modules for the board now. Kind regards, Martin List-Petersen -- You shall judge of a man by his foes as well as by his friends. -- Joseph Conrad ___ 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 dual licensing
Brian Capouch wrote: I would like to see you say out loud, just once, that those of us who know all of that and disclaim our work to Digium are not necessarily idiotic boobs who don't know what we're doing. As Joe already pointed out, he doesn't believe this to be the case :-) However, this thread brings to mind a side-issue that I've been bothered about: I have improvements in my local Asterisk tree that I _cannot_ get merged into the main Asterisk tree, no matter how wonderful/exciting/magical they are, because they are based on code written by others, released under the GPL, and those authors will not agree to give Digium an unrestricted license to their code. This is a big concern to me, for two reasons: First is that it can (and will) stifle Asterisk development to some degree, because interested parties cannot just grab best of breed code that they find out there in the wild (licensed under the GPL) and incorporate it into Asterisk. This means that developers must implement _from scratch_ equivalent code if they want it to get into Digium's Asterisk tree. Second is that even if a developer implements the code _from scratch_, if they have seen the original code distributed under the GPL, and their re-implementation ends up being very similar to the original, they cannot legally contribute that code under the terms of Digium's disclaimer, because there is some doubt as to whether they have complete rights over what that they are contributing. Certainly Digium is protected, because the disclaimer absolves them of the burden of proving whether any contributed code is actually being legally contributed or not, but the contributor exposes themselves to possible actions, and it could harm the Asterisk name/brand/reputation if such code was later found to have been improperly contributed. This issue as recently dealt with in the Linux kernel community, but there is less of an issue there because contributions are pure GPL, there is no dual licensing model available. In summary, it bothers me that contributions to Digium's Asterisk tree must be clean room implementations, without reference to existing alternatively-licensed implementations, unless those reference implementations can be re-licensed under Digium's terms. Please understand that I too am very happy that Digium exists, has provided Asterisk to the community, and I'm happy to help them earn an income and continue supporting/extending Asterisk. What I'm concerned about is that Asterisk will not be able to grow as well as it could if these license restrictions were not in place, and since some of us (myself included) are basing business enterprises around Asterisk, I want to see the product be able to do everything it is capable of, in the best way possible, not only the ways that are possible via clean-room implementation. Keep in mind that I am not a lawyer, don't play one on TV, nor have I discussed these issues with one. I do, however, have a very good understanding of the GPL and Digium's long-form disclaimer (or at last I think I do G), and I have discussed these issues with others who I have reason to believe also understand the relevant documents. ___ 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: random echo on TA750
all i have is random echo I have already 4 TA750 with full FXO echocancel=yes and echo training=800 - what should i do? - could it be solved with tweaking echo params on *? - is there any additional devices that can be added between Channel Bank and * to get rid off echo forever? any help would appreciated Paradise Dove ___ 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] SysMaster and GPL Violation
Martin List-Petersen wrote: Surely, but if you contribute to a project, shouldn't you allways check the license ? Would LICENSE, COPYRIGHT or README be the first places to look, if not on the website ? Absolutely, but all license and copyright files in the GPL Asterisk distribution are pure GPL, and do not mention that Digium has an alternative licensing method available to them. In fact, the LICENSE file at the top of the Digium-distributed Asterisk tree is an exact copy of the GPL version 2 as distributed by the FSF. Anyone who downloads this code, learns it, makes changes, and then decides to contribute those changes back to the project will only _then_ learn that they must allow Digium to license their code under non-GPL terms if they want their changes incorporated in to the project. ___ 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 dual licensing
On Sat, 13 Nov 2004 10:59:35 -0700, Kevin P. Fleming [EMAIL PROTECTED] wrote: Brian Capouch wrote: I would like to see you say out loud, just once, that those of us who know all of that and disclaim our work to Digium are not necessarily idiotic boobs who don't know what we're doing. As Joe already pointed out, he doesn't believe this to be the case :-) However, this thread brings to mind a side-issue that I've been bothered about: I have improvements in my local Asterisk tree that I _cannot_ get merged into the main Asterisk tree, no matter how wonderful/exciting/magical they are, because they are based on code written by others, released under the GPL, and those authors will not agree to give Digium an unrestricted license to their code. This is a big concern to me, for two reasons: First is that it can (and will) stifle Asterisk development to some degree, because interested parties cannot just grab best of breed code that they find out there in the wild (licensed under the GPL) and incorporate it into Asterisk. This means that developers must implement _from scratch_ equivalent code if they want it to get into Digium's Asterisk tree. Second is that even if a developer implements the code _from scratch_, if they have seen the original code distributed under the GPL, and their re-implementation ends up being very similar to the original, they cannot legally contribute that code under the terms of Digium's disclaimer, because there is some doubt as to whether they have complete rights over what that they are contributing. Certainly Digium is protected, because the disclaimer absolves them of the burden of proving whether any contributed code is actually being legally contributed or not, but the contributor exposes themselves to possible actions, and it could harm the Asterisk name/brand/reputation if such code was later found to have been improperly contributed. This issue as recently dealt with in the Linux kernel community, but there is less of an issue there because contributions are pure GPL, there is no dual licensing model available. In summary, it bothers me that contributions to Digium's Asterisk tree must be clean room implementations, without reference to existing alternatively-licensed implementations, unless those reference implementations can be re-licensed under Digium's terms. Please understand that I too am very happy that Digium exists, has provided Asterisk to the community, and I'm happy to help them earn an income and continue supporting/extending Asterisk. What I'm concerned about is that Asterisk will not be able to grow as well as it could if these license restrictions were not in place, and since some of us (myself included) are basing business enterprises around Asterisk, I want to see the product be able to do everything it is capable of, in the best way possible, not only the ways that are possible via clean-room implementation. Keep in mind that I am not a lawyer, don't play one on TV, nor have I discussed these issues with one. I do, however, have a very good understanding of the GPL and Digium's long-form disclaimer (or at last I think I do G), and I have discussed these issues with others who I have reason to believe also understand the relevant documents. ___ 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 Brian, Keep this thread. It will make excellent material for Philosophy 320 - Logic and Critical Reasoning, Evaluating Arguments. James Taylor -- Using Opera's revolutionary e-mail client: 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] SysMaster and GPL Violation
On 12/11/2004 16:08 Matteo Brancaleoni said the following: I too demand sysmaster either pay Digium for a non-gpl license or publicly admit the fact that they have repackaged Asterisk and contribute enhancements to Asterisk back to the GPL. *if they have made any enhancements* :) actually, the terms of the GPL do not require sysmaster to state that they are using asterisk. all the GPL requires is that anyone who is given a binary be also given the source (if asked) and that any enhancements made to the original work be also GPLed. as such, sysmaster does not _have_ to give any enhancements back, they just have to give it to their customers. of course, since the rights of the GPL are passed on to the customers, the customer can then distribute the enhancements to anyone they wish. :) -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ 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] Broadvoice asterisk patch
On 11/11/2004 06:08 Steven Sokol said the following: The patch is necessary because (I think I have this correct -- forgive me if I scramble any of the details) the Asterisk SIP channel was not caching the MD5 result of the original authentication dialog, and was instead forcing the BroadVoice system to perform the complete authentication sequence every 16 seconds for every Asterisk system connected. Apparently this causes a huge drain on their application servers. would this patch help those who're not using broadvoice, i.e. does it fix an issue with the way asterisk does not handle SIP registrations correctly ? -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ 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] SysMaster and GPL Violation
Citat Kevin P. Fleming [EMAIL PROTECTED]: Martin List-Petersen wrote: Surely, but if you contribute to a project, shouldn't you allways check the license ? Would LICENSE, COPYRIGHT or README be the first places to look, if not on the website ? Absolutely, but all license and copyright files in the GPL Asterisk distribution are pure GPL, and do not mention that Digium has an alternative licensing method available to them. In fact, the LICENSE file at the top of the Digium-distributed Asterisk tree is an exact copy of the GPL version 2 as distributed by the FSF. Anyone who downloads this code, learns it, makes changes, and then decides to contribute those changes back to the project will only _then_ learn that they must allow Digium to license their code under non-GPL terms if they want their changes incorporated in to the project. No .. the README tells about the Dual License. /Martin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SysMaster and GPL Violation
[snip] Really? Wouldn't it be nice, then, if Digium explicitly stated that this was their intention, in their little agreements? Why aren't YOU stating your own intention with this whole thread, or do you even realize it fully yourself? Your intent, whether you realize it or not, is to effectively chain Mark and all the contributors by the ankles and make them work for your benefit, until they starve to death. You can dance around and say outright that you don't really want that, but everything else you're saying leads to that result. To an extent, screw the FSF's opinion on this. Really? Be very careful. Once you say that, you really begin to slide back down from Mount Principles into the depths of license evil. Capitalizing a false concept doesn't make it any more real. But if you're going to adopt the GPL, and then you're going to cut a big hole in it, then I don't think it's wrong to at least discuss it, in a variety of contexts, including that which would likely be promoted by the FSF. In the same line where they say it is ethically tainted, they also say the copyrightholder can do what ever they want. Of course, because that's legal fact. And you are contradicting yourself, here. First you say its not wrong to discuss it, and then you say its wrong to suggest that the copyright holder can do what they want, which is exactly what Mark is doing. Make up your mind, would ya? Or are you instead saying the copyright law is immoral and should be abolished? That would be nice, wouldn't it. Then you could seize all of Mark's work outright and you wouldn't have to argue and type so much. The fact is, I and everyone else defending Mark are doing so because Mark's right are MY rights, they are their rights, and they are even YOUR rights (yes, everyone get to go along for the ride). We're talking about the right of the creator of something of value to use and dispose of it as he sees fit; otherwise known as property rights. Your saying he doesn't have that right, and I'm saying he does. Just because he adopted the GPL in some form doesn't mean that Asterisk is now public property. Yes, he took the GPL, modified to his own use, and went forward. We all know this, and SO WHAT? Asterisk is his product, its his decision to do with as he sees fit, and you don't have to like it. You can either use Asterisk by whatever terms he makes up, or not, and that is that. All this other crap is you trying to say that Mark has no right to do with Asterisk what he wants, when he made it in the first place, simply because of what terms he chose and how he chose them. Or for some other poster, because he chose to use Asterisk in his corporation and now _somehow_ he should have some say in how its developed/maintained just because he's using it. Well you know what, you can always stop using it if you don't like it, or you never should have started in the first place. You knew what the score was when you started, and if you didn't its your own fault for being too lazy to read and understand. In my opinion, Mark's biggest mistake has been adopting the GPL in the first place. He could have written his OWN general public license verbage and called it something else, and then noone would have this angle to question whether or not he was trying to trick or deceive people into handing over their intellectually property for nothing. -- Tom ___ 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] NAT
the asterisk suport NAT as ser? or need modules from modules or special cofiguration? _ Do You Yahoo!? Información de Estados Unidos y América Latina, en Yahoo! Noticias. Visítanos en http://noticias.espanol.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] BroadVoice
Anybody else having Broadvoice registration problems today? -- Gary White [EMAIL PROTECTED] Network Administrator Internet Pathway 105 D East Church Street Voice: 601-776-3355 P. O. Box 777 Fax: 601-776-2314 Quitman, MS 39355Registered Linux User Number 198875 smime.p7s Description: S/MIME Cryptographic Signature ___ 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] BroadVoice
yes.. started around 12:00 noon EST I get "sip_reg_timeout: Registration for '[EMAIL PROTECTED]" Does anyone know if this is related to the channels patch? Doug Gary White (Network Administrator) wrote: Anybody else having Broadvoice registration problems today? ___ 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] Broadvoice Patch issues
After the broadvoice patch I am getting busy messages also on call in. Is anyone else experiencing a lot of busy signals after this patch? ie Broadvoice becomes disassociated with asterisk.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BroadVoice
Same here... Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Sat, 13 Nov 2004, Doug Shubert wrote: yes.. started around 12:00 noon EST I get sip_reg_timeout: Registration for '[EMAIL PROTECTED] Does anyone know if this is related to the channels patch? Doug Gary White (Network Administrator) wrote: Anybody else having Broadvoice registration problems today? ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-11-13%5Ccf62fbdc4a664e39b123d2ef9ce2d9a4C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] BroadVoice
Its working here, some issues tho. All outbound calls have no CID. -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce Komito Sent: Saturday, November 13, 2004 1:16 PM To: Doug Shubert Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] BroadVoice Same here... Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Sat, 13 Nov 2004, Doug Shubert wrote: yes.. started around 12:00 noon EST I get sip_reg_timeout: Registration for '[EMAIL PROTECTED] Does anyone know if this is related to the channels patch? Doug Gary White (Network Administrator) wrote: Anybody else having Broadvoice registration problems today? --- - ___ 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 This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-1 1-13%5Ccf62fbdc4a664e39b123d2ef9ce2d9a4C=2 -- --- This message has been inspected by DynaComm i:mail --- ___ 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] BroadVoice
Yes. :-( -jeff -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary White (Network Administrator) Sent: Saturday, November 13, 2004 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] BroadVoice Anybody else having Broadvoice registration problems today? -- Gary White [EMAIL PROTECTED] Network Administrator Internet Pathway 105 D East Church Street Voice: 601-776-3355 P. O. Box 777 Fax: 601-776-2314 Quitman, MS 39355Registered Linux User Number 198875 ___ 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] BroadVoice
Working fine for me. I installed their patch like they asked. I'm registering with proxy.dca.broadvoice.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] SysMaster and GPL Violation
Martin List-Petersen wrote: No .. the README tells about the Dual License. Not quite. The README says that Digium can grant others the right to create modules that link with Asterisk at runtime but are not required to be licensed under the GPL. It does not say that Digium can grant others the right to distribute binaries of Asterisk without making the GPL parts of the code available to the recipients of those binaries. In addition, it does not say that Digium can grant others the right to distribute binaries of Asterisk that contain changes to the _core code_ without making those changes available under the GPL. It is my understanding that Digium's disclaimer _does_ give them the ability to license Asterisk in this fashion, but the LICENSE and README files in the GPL Asterisk source do not make this clear to those who have copies of the distribution. ___ 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: random echo on TA750
all i have is random echo I have already 4 TA750 with full FXO echocancel=yes and echo training=800 - what should i do? - could it be solved with tweaking echo params on *? - is there any additional devices that can be added between Channel Bank and * to get rid off echo forever? depends on the source of your echo if its the motherboard latency issue then no external device will help but assuming it external to the b check the external t1 hardware echo can's I use a older tellab 2572 64ms tail inline between the t100 the ta 750 with all asterisk echo related settings OFF I have seen other echo can reported to work ( the telllab 257? can be had for 50/card but the chassis are hard to come by you nned to do leg work with the after market used online suppliers) ___ 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] SysMaster and GPL Violation
Tom Lahti wrote: Or for some other poster, because he chose to use Asterisk in his corporation and now _somehow_ he should have some say in how its developed/maintained just because he's using it. Well you know what, you can always stop using it if you don't like it, or you never should have started in the first place. You knew what the score was when you started, and if you didn't its your own fault for being too lazy to read and understand. Wow, nice way to completely mis-state what I wrote. Are you saying that those of us that are using the product should not be allowed to voice our opinions about its licensing, development and maintenance? That we should all just shut up and take whatever Mark co. give us? If that's the case, then this is most definitely NOT an open-source project at all. My statements in no way implied that _I_ should have any say in how it is developed or maintained, any more than anyone else has a say in that via this list and the other resources made available by Digium. My statements said only that I was _concerned_ that Asterisk's development was hampered by the dual licensing model and that I felt a more normal pure GPL licensing model would help its future. Yes, I have known this from the day I started looking at Asterisk, before I even compiled it for the first time. I also know that if I find it to be a large enough obstacle, that I am free to take the GPL distribution of Asterisk and make my own tree available to anyone who wants it, without contributing all of my changes back to Digium's tree (those that Digium would not accept). From what I understand, the bristuff tree is already dealing with this same issue. I don't want to go that route, I'd much rather be able to contribute all my changes back to Digium's tree, but I can't. If you are saying that I have no right to say that, I'd be curious as to the basis of your opinion :-) ___ 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] BroadVoice
Well, back working now. Guess they were having problems again. -- Gary White [EMAIL PROTECTED] Network Administrator Internet Pathway 105 D East Church Street Voice: 601-776-3355 P. O. Box 777 Fax: 601-776-2314 Quitman, MS 39355Registered Linux User Number 198875 smime.p7s Description: S/MIME Cryptographic Signature ___ 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] SysMaster and GPL Violation
Citat Joe Greco [EMAIL PROTECTED]: [SNIP] There are a number of competing theories on whether or not the author of a public domain bit of code could be liable, with varying amounts of case law, as I understand it: 1) One theory is that you may place code in the public domain, with explicit no-warranty disclaimers (this seems sensible to me). 2) Another theory says that such disclaimers are not legally binding, and that you would need to embed it within a license, copyright agreement, contract, or something like that to prohibit use of the code in the event that the recipient did not agree. 3) Another theory says that liability is only a concern where money has changed hands. There are apparently some finer-grained distinctions in there somewhere. I don't know if I'd want to submit major changes to a project and open myself up to the possibility of having to legally test whether or not a no-warranty clause on a public domain code contribution could be enforced. [SNIP] I'm not in favour for Public Domain disclaims either, but that is something any contributor should take up with himself. Or, preferably, with a good lawyer. It is fairly certain that the no warranty clauses are sound. It is less certain that public domain code would necessarily afford you that protection. :-( [SNIP] The second one does neither state, that you sign your copyrights over to Digium. It gives Digium a non-exclusive, non-revocable right to use your changes. That's it. I'd check with our IP lawyer if I really cared. However, it looks a bit more sweeping than that. Even though I'm not a lawyer, I can disprove your statement: if I sign this agreement *and don't even contribute anything*, but were to purchase ownership of a patent covering something that conflicts with Asterisk, this agreement grants Digium rights that you haven't acknowledged. See, that's the ugly thing about legal documents. There are endless things to consider. We can of course agree that it ought not work that way, but that's just pie-in-the-sky. Section 4 is a bit fishy, i would agree on that, however it doesn't really apply outside the United States (yet, and hopefully never, at least not for Europe). We have some dumb legal stuff. Heh. Look at our most recent Attorney General. Patriot Act. 'Nuff said. Simply because software patents is a phaenomenon, that not exists in Europe (covering me) at this point. For me personally, if I would go down the road and get a software patent that would conflict with Asterisk, I still had no problem with the fact, that I couldn't make Digium pay. It is the price you pay for using and contributing, if you sign this disclaimer, the IP of Digium. But I thought the price I paid and the rights I got was all outlined in the GPL! But that is a personal choice. If I wanted to cover me from not giving Digium of any such patents, that I own, I should not contribute (but drag money out of them for my patent :o) ) or disclaim the patches i contribute to the public domain. However, the agreement doesn't say as of this moment. It talks about future acquisitions, and makes the provision binding upon the contributor. What if you signed this, and some years later Digium went closed-source? It appears to me as though there are still avenues forward in which Digium would have the right to take your work into their private codebase. Sweetheart arrangements *have* gone sour in the past. I certainly have no reason to expect Digium to do any such thing, but are you willing to sign a paper that legally obligates you to do certain things, and cannot be withdrawn or cancelled once tendered? Anyhow, where is the spirit ? You are advocating FSF and the GPL and taking in consideration, that you ever would have a software patent ? We're a business. There's no reason to think that we could never acquire a company that had software patents. Because a hardware patent can hardly conflict with Asterisk. [SNIP] The one who's looked at the Asterisk web site, has gone to the bugs link, and is then confronted by the short form disclaimer, and doesn't really know or care about Digium, or that there was some requirement that s/he become intimately familiar with some evil company (and be aware that some GPL advocates view companies thusly) and all of the above. Or, to turn this around: What harm would there be in outlining this explicitly within the agreement itself? [SNIP] I do agree, that the Asterisk Website should state something about Asterisk License. I was surprised that there not even is mentioned, that it is GPL. The only mention of GPL is that it is based on a GPL copyrighted PRI stack. Um, it's in LICENSE in the Asterisk source tree, and in the headers of the source files. The GPL, that is. [SNIP] Only if they haven't read the essential Documentation (Read README)
Re: [Asterisk-Users] BroadVoice
By the way this was not related to the patch. I installed it Friday and did not start having trouble until today. Well, back working now. Guess they were having problems again. ___ 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 -- Gary White [EMAIL PROTECTED] Network Administrator Internet Pathway 105 D East Church Street Voice: 601-776-3355 P. O. Box 777 Fax: 601-776-2314 Quitman, MS 39355Registered Linux User Number 198875 smime.p7s Description: S/MIME Cryptographic Signature ___ 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 dual licensing
[off-list] Brian Capouch wrote: I would like to see you say out loud, just once, that those of us who know all of that and disclaim our work to Digium are not necessarily idiotic boobs who don't know what we're doing. As Joe already pointed out, he doesn't believe this to be the case :-) Correct, of course. :-) I've said repeatedly that authors can do whatever they want. [Then you go on to bring up several interesting points, which bear a fair amount of consideration.] I've studied the GPL at arm's length for some years now. We long ago made a business decision to license code under BSD for a variety of reasons, and I've studied the ins and outs of that decision in detail over the years. I am finding it fascinating to contemplate the complexities of bending the GPL in this manner, since it isn't an issue which I've extensively considered, and your comments extend my understanding of the ins and outs of this in a direction I hadn't even started to consider. Thanks. ;-) ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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 dual licensing
Please [off-list] - Original Message - From: Joe Greco [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, November 13, 2004 4:10 PM Subject: Re: [Asterisk-Users] Asterisk dual licensing [off-list] Brian Capouch wrote: I would like to see you say out loud, just once, that those of us who know all of that and disclaim our work to Digium are not necessarily idiotic boobs who don't know what we're doing. As Joe already pointed out, he doesn't believe this to be the case :-) Correct, of course. :-) I've said repeatedly that authors can do whatever they want. [Then you go on to bring up several interesting points, which bear a fair amount of consideration.] I've studied the GPL at arm's length for some years now. We long ago made a business decision to license code under BSD for a variety of reasons, and I've studied the ins and outs of that decision in detail over the years. I am finding it fascinating to contemplate the complexities of bending the GPL in this manner, since it isn't an issue which I've extensively considered, and your comments extend my understanding of the ins and outs of this in a direction I hadn't even started to consider. Thanks. ;-) ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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] SysMaster and GPL Violation
Citat Joe Greco [EMAIL PROTECTED]: If so, you are allowed to fork or distribute your own patches and not sign any of the disclaimers. This is what Klaus-Peter Junghanns does with the bristuff patch (adds various addons, including full DSS1 BRI support to Asterisk, Zaptel and Libpri and enabled Asterisk for a whole range of standard ISDN cards) and chan_capi. He refuses to sign the disclaimer because he doesn't agree with the Dual License that Asterisk is maintaining and his work contain contributions from others, that are GPL only. It just doesn't go in the asterisk mainstream source then. Might be cumbersome, but that is the way it is. I don't even think there's anything wrong with that, and it is interesting to hear that there's already a forked project. Currently it's just a patchset against the asterisk, zaptel, libpri stable release, that is distributed seperatly, but you could categorise that a fork allready. Kind regards, Martin List-Petersen -- One good thing about music, Well, it helps you feel no pain. So hit me with music; Hit me with music now. -- Bob Marley, Trenchtown Rock ___ 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] spandsp problem
From the * machine, I'm able to fax (txfax) just the first page of a multi-page document. I've tried this on a Sharp UX-P200 as well as an HP 5510 machines with the same result. The document was assembled using gs from a series of .ps documents. The fax machines are connected to an Adtran 750 channel bank and then to the * machine with a T-1 crossover cable. I can fax from one machine to the other in this configuration without problems. zttool doesn't register any missed interrupts. I'm using spandsp-0.0.1k The span-0.0.2pre4 doesn't work for me. The logs all have the following: ... (faxing first page) MPS: 4f HDLC underflow in state 13 Changed phase from 4 to 3 Slow carrier up/down/up/down, etc. XCN: fa XCN with final frame tag In state 13 Disconnecting Changed phase from 3 to 7 Changed phase from 7 to 8 Hungup While I was composing this email, I tried the fax again on the HP, and it worked. That tells me that the problem is intermittent and that there's nothing wrong with the document. Where should I be looking? Should I put the fax machines on a TDM card to eliminate the T-1? Should I be trying to get the spandsp-0.0.2 version to work? Thanks for your help. Mike -- Michael Welter Introspect Telephony Corp. Denver, Colorado US +1.303.674.2575 [EMAIL PROTECTED] www.introspect.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] SysMaster and GPL Violation
Are you saying that those of us that are using the product should not be allowed to voice our opinions about its licensing, development and maintenance? That we should all just shut up and take whatever Mark co. give us? If that's the case, then this is most definitely NOT an open-source project at all. Not at all. I believe you should voice whatever opinion you have, but you should bear in mind while doing so that Mark is under no obligation to even listen to you, and should not be. Those like Joe seem to be searching for some _legal_ way to do just that, which disgusts me because I know when they are successful it sets a legal precedent that could be used against _me_. As long as you aren't pursuing some legal angle whereby you can take over control of Asterisk, whether in part or in whole, then bitch away! That's what free speech is all about. Let me try to make my point about property rights as clear as I can. The right to property means the right to use and dispose of the property. If someone is holding a gun to your head telling you what to do with it, it isn't _really_ yours, no matter how much lip service is paid to the fact that you're the one actually touching the property. If someone is forcing you (and I mean force in the truest sense, i.e. the laws of a government) to destroy your property or hand it over to someone else, then it is really the governments and they are just letting you pretend to own it. A good example of this is the current state of the ILECs in the US. What do you think would happen if SBC, Qwest etc. woke up tomorrow and blew up all their switches and said well, they belonged to us, we paid excise taxes on them, we could do with them what we want. There would be a whole lot of board members in jail, for starters, for destroying the public telephone network. Well, who does it belong to, the public, or the phone company? It can't be both. Only one of them has the right to blow it up, and I'll give you one guess as to which. Property rights are not a matter of degree. You cannot be sort of pregnant, you can't be somewhat dead, and you can't kind of own something, where others are partially in control of its use and disposal. It's either yours or it isn't. Communism, ala the FSF and Stallman, don't work. Look at the history of communist states that have existed and those that are left and tell me that system works. I'd really love a good laugh. If you don't think the FSF is a communist establishment, go read the GNU Philosophy on the FSF web site. Everything is about making the collective, the public, or whatever else you want to call it, more important than the individual, and that is the basic principle upon which communism is built. Their idea is that noone has a right to his own ideas; that whatever you as developers may dream up ought to be the rightful property of the public, and Stallman says so over and over and over. And where does that leave you, the developer, motivationally? Where does that leave you at the end of the month when its time to pay rent and buy groceries? It doesn't take rocket science to figure out why it doesn't and _cannot_ work. The only rational way for men to deal with each other is through trading value for value, which is why I said the Asterisk licensing sets up a trade. You can use Asterisk not really for free, but in exchange for what Mark can add to Asterisk along the way, rather than being compensated monetarily. -- Tom ___ 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] SysMaster and GPL Violation
[snip] Really? Wouldn't it be nice, then, if Digium explicitly stated that this was their intention, in their little agreements? Why aren't YOU stating your own intention with this whole thread, or do you even realize it fully yourself? I do. I don't believe you do. Your intent, whether you realize it or not, is to effectively chain Mark and all the contributors by the ankles and make them work for your benefit, until they starve to death. You can dance around and say outright that you don't really want that, but everything else you're saying leads to that result. Nope. If anything, I'd have preferred to see Asterisk licensed under a BSD license, where this wouldn't even be a consideration. You'll note that I've said a number of times that I believe authors are entitled to do as they wish with their work. That's a central idea to this discussion, but not only from a Mark's rights point of view. To an extent, screw the FSF's opinion on this. Really? Be very careful. Once you say that, you really begin to slide back down from Mount Principles into the depths of license evil. Capitalizing a false concept doesn't make it any more real. Most people would agree that the FSF has a highly principled - even if unrealistic - view of what free software should mean. We don't all agree with it, but if you use the GPL and then you selectively pick and choose certain principles, that would generally be considered sliding down the slippery slope. But if you're going to adopt the GPL, and then you're going to cut a big hole in it, then I don't think it's wrong to at least discuss it, in a variety of contexts, including that which would likely be promoted by the FSF. In the same line where they say it is ethically tainted, they also say the copyrightholder can do what ever they want. Of course, because that's legal fact. And you are contradicting yourself, here. First you say its not wrong to discuss it, and then you say its wrong to suggest that the copyright holder can do what they want, which is exactly what Mark is doing. I'm not contradicting myself; I would never say anything other than the copyright holder can do what they want, and I really don't know where you got that from. Having the right to do what you want, however, does not make you totally exempt from other considerations. Make up your mind, would ya? Or are you instead saying the copyright law is immoral and should be abolished? That would be nice, wouldn't it. Then you could seize all of Mark's work outright and you wouldn't have to argue and type so much. I really have very little interest in Asterisk. I've been using it to set up an office PBX. Under the terms of the GPL, I'm free to do that, and to make any local changes I would like, without any flak from Mark or anyone else here. I've already seized that work, just as you have. You want to criticize someone? Go find someone who's distributing it in violation of the GPL. The fact is, I and everyone else defending Mark are doing so because Mark's right are MY rights, No, Mark's rights are Mark's rights. If he graciously grants you a GPL license to use his code, that is a privilege which could be revoked in the future - there's no guarantee you will get your hands on Asterisk 2.0.0. (He cannot revoke your right to use something that was GPL'd, but that's about it...) they are their rights, and they are even YOUR rights (yes, everyone get to go along for the ride). We're talking about the right of the creator of something of value to use and dispose of it as he sees fit; otherwise known as property rights. No, we're not talking about that at all. You've completely missed the point. We're talking about property rights of contributors. Your saying he doesn't have that right, and I'm saying he does. I said no such thing. Just because he adopted the GPL in some form doesn't mean that Asterisk is now public property. Well, not exactly. The GPL's intent is to try to create a meta-class of public software. Mark still owns title to it, of course, but the GPL grants rights in a manner designed to accomplish certain goals. Yes, he took the GPL, modified to his own use, and went forward. We all know this, and SO WHAT? Asterisk is his product, its his decision to do with as he sees fit, and you don't have to like it. You can either use Asterisk by whatever terms he makes up, or not, and that is that. Actually, he didn't take the GPL and modify it. To do that would require him to call the license something besides GPL. The point is about the finer points of contributed code ownership, and what happens from a licensing point of view.. All this other crap is you trying to say that Mark has no right to do with Asterisk what he wants, when he made it in the first place, simply because of what terms he chose and how he chose them. All this other crap you accuse me of saying...
Re: [Asterisk-Users] RE: BRI in the US
Andreas Anderson wrote: Hi Brian, One goal is to get BRI support in Zaptel if possible. I'm right now in the planning stage :P Plus BRI is much cooler than pots. Why invent the wheel again, what's wrong with bristuff from junghanns.net? bye, aa _ Listen to music online with the Xtra Broadband Channel http://xtra.co.nz/broadband ___ 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 Here in the states we have this critter called SPID (Service Provision Identifier (?)) that is not supported by bristuff. -- Michael Welter Introspect Telephony Corp. Denver, Colorado US +1.303.674.2575 [EMAIL PROTECTED] www.introspect.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] SPA-3000 Wizard for Asterisk
On Sat, 2004-11-13 at 01:48 -0800, Dameon D. Welch-Abernathy wrote: For your testing pleasure. Feedback welcome: http://voxilla.com/spa3kasterisk.php -- PhoneBoy PhoneBoy! What is the point of testing it you don't even know how to ship whatever you sell internationally and/or resolving the problem with a customer if they need your help. The last time we ordered from you the SPA-3000 was defective, we shipped you the defective and you ship us back another one but you put the value for shipping the same one like for sale; you did not indicate on the shipping papers that it was Repair/Return shipment so we end up paying over $50.00 for brokerage and taxes (and it suppose to be free); that is almost 50% value what the unit is worth! I called you several time left a message on your answering machine, I faxed a message to fix it but you didn't bother to respond. All it was needed is to send a fax to UPS Re-Rate Department and mention that shipment such and such was Repair and Return. If you need my invoice or tracking number I can provide that to you as well. Unless you learn how to solve a problem in a business like manner, I can not recommend any of the product you sell to anybody especially international customers. -- #Joseph ___ 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] Over 10,000 lines. Will asterisk manage?
On Saturday 13 November 2004 17:55, Billy Huddleston wrote: So, why not use SER to register all the SIP phones, as it doesn't handle the media-streams, just keeps track of the phones and does the 'handshake'. SER is supposed to be able to handle over 50.000 calls at a time, so one SER server would be enough. Then interface this with one (or more) Asterisk servers to connect to the local PSTN. But maybe I'm missing something fundamental, in which case I'm happy to learn. Um, Wrong, You can do re-invites and have the media go point-to-point, We do it all the time. Ok, sounds good. So if I understand correctly Asterisk can also be used as a SIP gateway instead of SER? And when the media passes point-to-point one Asterisk server would be able to handle connecting thousands of calls concurently? Or is there still a reason why a SIP - SER - Asterisk - PSTN setup would be preferable? Thanks, Richard. ___ 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] SysMaster and GPL Violation
No .. the README tells about the Dual License. Try reading it as though you didn't know about the dual license. If you read it without that knowledge, it sounds as though they are trying to provide a way within the terms of the GPL to allow linking with stuff that isn't GPL'd. This could be important for a telephony application, what with patented codec's and stuff like that. I certainly took it that way. It does say: If you have any questions, whatsoever, regarding our licensing policy, please contact us. but I don't consider that to be a disclosure of a dual license. As a matter of fact, the ChangeLog says: * Asterisk 0.1.1 -- Revised License -- Pure GPL, nothing else Knowing about the dual licensing, I think it may be possible to torture the language within the README to at least hint that there might be one. Reading the README without knowing about the dual licensing, I don't get dual licensing disclosure out of it at all. Maybe someone could get out some crayons and draw me that diagram I was promised by someone a few messages back. :-) ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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] SysMaster and GPL Violation
Are you saying that those of us that are using the product should not be allowed to voice our opinions about its licensing, development and maintenance? That we should all just shut up and take whatever Mark co. give us? If that's the case, then this is most definitely NOT an open-source project at all. Not at all. I believe you should voice whatever opinion you have, but you should bear in mind while doing so that Mark is under no obligation to even listen to you, and should not be. Those like Joe seem to be searching for some _legal_ way to do just that, which disgusts me because I know when they are successful it sets a legal precedent that could be used against _me_. As long as you aren't pursuing some legal angle whereby you can take over control of Asterisk, whether in part or in whole, then bitch away! That's what free speech is all about. What exactly are you accusing me of? I don't even get it. Just for the record, if I'm looking for some legal way to do something, then it'll be in the hands of our lawyers, and you'll most likely find out on letterhead or by being served. ... but that's not applicable here, since I'm not. I was at one point looking for a legal way to contribute a few Asterisk changes. I checked the disclaimers. I knew we couldn't sign them. End of that. However, I believe it's worth discussing the reasons that BSD- or GPL-licensed changes to a GPL'd project would not be accepted by that project, not on merit of the changes, but because of rights. Let me try to make my point about property rights as clear as I can. The right to property means the right to use and dispose of the property. If someone is holding a gun to your head telling you what to do with it, it isn't _really_ yours, no matter how much lip service is paid to the fact that you're the one actually touching the property. If someone is forcing you (and I mean force in the truest sense, i.e. the laws of a government) to destroy your property or hand it over to someone else, then it is really the governments and they are just letting you pretend to own it. A good example of this is the current state of the ILECs in the US. What do you think would happen if SBC, Qwest etc. woke up tomorrow and blew up all their switches and said well, they belonged to us, we paid excise taxes on them, we could do with them what we want. There would be a whole lot of board members in jail, for starters, for destroying the public telephone network. Interestingly enough, that might only be a problem for an ILEC. CLEC's are generally not obligated to provide service (or even to exist). We've seen CLEC's fold and terminate service in the past. Well, who does it belong to, the public, or the phone company? It can't be both. Only one of them has the right to blow it up, and I'll give you one guess as to which. Property rights are not a matter of degree. You cannot be sort of pregnant, you can't be somewhat dead, and you can't kind of own something, where others are partially in control of its use and disposal. It's either yours or it isn't. Heh. The spammers would like to own your computer in that way. ;-) Communism, ala the FSF and Stallman, don't work. Look at the history of communist states that have existed and those that are left and tell me that system works. I'd really love a good laugh. If you don't think the FSF is a communist establishment, go read the GNU Philosophy on the FSF web site. Everything is about making the collective, the public, or whatever else you want to call it, more important than the individual, and that is the basic principle upon which communism is built. Their idea is that noone has a right to his own ideas; that whatever you as developers may dream up ought to be the rightful property of the public, and Stallman says so over and over and over. And where does that leave you, the developer, motivationally? Where does that leave you at the end of the month when its time to pay rent and buy groceries? It doesn't take rocket science to figure out why it doesn't and _cannot_ work. I came to a somewhat similar conclusion years ago. However, I hope that you would concede that there is a certain attractiveness to the general philosophy. So much software is locked up for no good reason. Closed development is a waste of resources. Think of where we might be if people weren't busily duplicating work. On the flip side, we have the BSD license capitalism, which relies perhaps too heavily on the willingness of contributors to contribute changes back. There's no real middle ground. If we lived in a world where programmers were tenured positions and did not have to worry about those pesky business fundamentals, I might actually be persuaded that the GPL was an ideal license. As it is, I like to believe that most people are inherently good, and as such, I believe that the BSD license is
[Asterisk-Users] Remote answer not detected
Here's my a section of my simple extensions.conf exten = s,1,Answer exten = s,2,playback(thx4call) exten = s,3,Dial(Zap/1|15) ; Calls channel 1 exten = s,4,playback(trying_bert) exten = s,5,Dial(Zap/4/2326932|15) exten = s,6,Voicemail,u100 exten = s,7,hangup exten = s,104,Voicemail,b100 exten = s,105,hangup exten = s,106,Voicemail,u100 It works, but when the call is routed out on ZAP/4 (at priority 5), Asterisk seems to not realize the call is answered. After 15 seconds it proceeds to voicemail interrupting the call. Can anyone help? DB ___ 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] SysMaster and GPL Violation
On Saturday 13 November 2004 12:19 pm, Martin List-Petersen wrote: It has been tested in city/county court in Munich (Germany) and found valid (http://yro.slashdot.org/article.pl?sid=04/07/23/1558219tid=117), not that that might help anybody in the US, but it is a start. Kind regards, Martin List-Petersen It is also being used by IBM against SCO. And so if IBM attorneys think it's good, there's good chance it is. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ 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] Remote answer not detected
hi Il dom, 2004-11-14 alle 00:13, DB ha scritto: Here's my a section of my simple extensions.conf snip exten = s,5,Dial(Zap/4/2326932|15) exten = s,6,Voicemail,u100 snip It works, but when the call is routed out on ZAP/4 (at priority 5), Asterisk seems to not realize the call is answered. After 15 seconds it proceeds to voicemail interrupting the call. Can anyone help? eh, perhaps with some details about your zap... ie what card? zaptel.conf? zapata.conf? matteo, still without divinatory powers -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ 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] NAT
On Saturday 13 November 2004 01:57 pm, Walter Willis wrote: the asterisk suport NAT as ser? or need modules from modules or special cofiguration? Hmm, your English is a bit too crippled to understand. I'm guessing you are asking if Asterisk supports NAT as something (server?) And then if it needs modules and special configurations. If you are asking if Asterisk can work through NAT then the answer is yes. But the real question is whether or not you are going to use SIP or IAX. IAX does NAT very well whereas SIP is problematic. You need to go to the wiki and read up about Asterisk to use it. It requires a lot of work to understand and yes, you need to configure it. Go to http://www.voip-info.org/wiki-Asterisk and read up on it. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ 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] Make good use of time. (was: SysMaster and GPL Violation)
Ok I'm not one to beat around the bush here... so here goes. This has got to be the most wasted energy I have ever seen. If only half of you put as much energy into pissing and moaning on the list as we do working to make asterisk better IT WOULD TEN TIMES BETTER!!! (Not that it isn't already, but it could be many times better). So I put it to you like this... SHUT THE HELL UP PEOPLE... Work on documentation or work on fixing bugs. This mindless arguing on the list gets us NOWHERE. We must all work together! A good place to start is http://bugs.digium.com and http://www.voip-info.org Now every one play nice before I call your mommie!!! bkw ___ 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] Remote answer not detected
Havent had that problem but look here http://www.voip-info.org/wiki-Asterisk+config+zapata.conf I would try callprogress=yes - Original Message - From: DB [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, November 13, 2004 6:13 PM Subject: [Asterisk-Users] Remote answer not detected Here's my a section of my simple extensions.conf exten = s,1,Answer exten = s,2,playback(thx4call) exten = s,3,Dial(Zap/1|15) ; Calls channel 1 exten = s,4,playback(trying_bert) exten = s,5,Dial(Zap/4/2326932|15) exten = s,6,Voicemail,u100 exten = s,7,hangup exten = s,104,Voicemail,b100 exten = s,105,hangup exten = s,106,Voicemail,u100 It works, but when the call is routed out on ZAP/4 (at priority 5), Asterisk seems to not realize the call is answered. After 15 seconds it proceeds to voicemail interrupting the call. Can anyone help? DB ___ 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] isdn to sip gw
Hello Im trying to have my normal incoming calls automatically forwarded to my SIP phone, or even better, directly to a given number thorough my SIP service provider. Example: Im visiting the office in Argentina or Spain, someone call to our office in Italy (a 'normal' PSTN call), then the Asterisk forward the call, thorough SIP, to the 'normal' PSTN number of the office in Argentina or Spain. What minimum hardware i need? I have Linux, some ISDN cards and some SIP (Budgetone 102) phones. Thanks for the help out there -- Pablo Povarchik Quality Colocation and Dedicated Servers services Colocation facilities include Fremont California, London UK and Trento Italy +--- FuturaHost.Com - Industrial Business Class ISP + | Web Hosting - Dedicated Servers - Colocation | [EMAIL PROTECTED] - http://futurahost.com/ - (+39) 0461 592710 | Get a high quality full cabinet with 5Mbps full burst included |for only ¤700/month, availability also in London +-+ ___ 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] Best setup for BudgeTone
Hi all, I'd like to know what's most reliable configuration for BudgeTone 101 in the following setup: PSTN | Legacy phones == Alcatel Omnipcx == QuadBRI-Asterisk1 | | IAX trunk | Asterisk2 == 25 Bugetone 101 System works, but people complain about frequent disconnect (hangups), or phone not ringing. I've read the wiki, but information seems a bit old. This is extracted from sip.conf: [362] type=friend context=interne callerid=xyz xxx362 ;host=dynamic host=192.168.4.42 username=362 disallow=all allow=ulaw allow=alaw dtmfmode=rfc2833 (similar entry for every phone). The phones are configured as recommended on the wiki. I recently moved to static IP but it doesn't seem to improve reliability. BT firmware is 1.0.5.16, and asterisk is stable version (1.0.2). Any advice to improve reliability (ie no disconnect nor phone not ringing) is welcome :) Thanks, Jean-Denis ___ 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] NAT
thank you, my English is terrible, I don't usually use it and ti is not my language. unfortunately they don't exist clever of mail in another language. and I don't have with the one who to practice it. XD I go he is necessary to have to practice it but. thanks you for the you help me. jijijiji _ Do You Yahoo!? Información de Estados Unidos y América Latina, en Yahoo! Noticias. Visítanos en http://noticias.espanol.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] NAT
Walter Willis wrote: thank you, my English is terrible, I don't usually use it and ti is not my language. unfortunately they don't exist clever of mail in another language. and I don't have with the one who to practice it. XD I go he is necessary to have to practice it but. thanks you for the you help me. jijijiji Escriba, entonces, en Espanol. Hay varios aqui que lo hablan. Mas o menos :--) Quizas lo podemos ayudar. 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] NAT
yee gracias por la ayuda pero de todos modos tengo que practicarlo. lo leo lo entiendo pero no lo escribo ni lo habloo! XD _ Do You Yahoo!? Información de Estados Unidos y América Latina, en Yahoo! Noticias. Visítanos en http://noticias.espanol.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] spandsp problem
Hi Michael, Michael Welter wrote: From the * machine, I'm able to fax (txfax) just the first page of a multi-page document. I've tried this on a Sharp UX-P200 as well as an HP 5510 machines with the same result. The document was assembled using gs from a series of .ps documents. The fax machines are connected to an Adtran 750 channel bank and then to the * machine with a T-1 crossover cable. I can fax from one machine to the other in this configuration without problems. zttool doesn't register any missed interrupts. I'm using spandsp-0.0.1k The span-0.0.2pre4 doesn't work for me. What happens with 0.0.2pre4? For most people that version gives better results than 0.0.1k. It seems to fix most of the quirks people have had. The logs all have the following: ... (faxing first page) MPS: 4f HDLC underflow in state 13 Changed phase from 4 to 3 Slow carrier up/down/up/down, etc. XCN: fa XCN with final frame tag In state 13 Disconnecting Changed phase from 3 to 7 Changed phase from 7 to 8 Hungup spandsp says we have more pages to send. The remote fax machine sends a disconnect message and hangs up. It sounds like the remote fax machine is at fault. Does it always fail with one machine, and succeed with another? Some machines can be set to receive a maximum of X pages. Could it be soemthing like that. While I was composing this email, I tried the fax again on the HP, and it worked. That tells me that the problem is intermittent and that there's nothing wrong with the document. Where should I be looking? Should I put the fax machines on a TDM card to eliminate the T-1? Should I be trying to get the spandsp-0.0.2 version to work? Try to get 0.0.2 working. It works well enough for other people than it is about to loose its pre status. 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] Cable for T1 connection: Crossover or straight through?
On November 13, 2004 12:11 pm, Malcolm Bader wrote: This is my second asterisk server but the first one with a T100P card. I connected it to the phone company(SBC) jack but have only a busy signal when calling the T1's number and nothing in the asterisk log files to indicate a connection. Do I need to use a crossover cable? Is the T100P's light Green or flashing red? Green means it sees the other side, and the other side sees it. i.e. the cabling is fine. Orange (well it's trying to be yellow) means that the other side can't see the T100P, but the T100P is seeing the other side. -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] manager api: how to handle failed calls
Hello again,
I am reposting this issue since I realized I posted in an existing
thread ( I am sorry about that).
I am still faced with the same problem since long time:
The question is how to correctly handle failed calls.
In my application I want to make hundreds of outgoing calls
automatically.
When the callee pick up the phone he gets a playback message and give
an acknowledge by means of dtmf code.
I make use of manager command originate, something like
Action:originate
channel: ZAP/g1/
Variable:X|Y|Z
extension: test
the extension test is something like
[test]
exten s,1 , wait ()
exten s, 2 , answer ()
exten s, 3 playback(XX)
The problem is since I don't use the application dial inside the
extension I
cannot get any value from
DIALSTATUS or HANGUPCAUSE variable
I tried several strategies:
1)
change the logic and use local pseudo channel
In the originate command if I use channel: local/[EMAIL PROTECTED]/n
where test1 is:
[test1]
exten = _.,1,Dial(ZAP/g1/g${EXTEN})
exten = _.,2,NoOp( 2 HANGUPCAUSE is ${HANGUPCAUSE})
exten = _.,3,NoOp( 2 DIALSTATUS is ${DIALSTATUS})
exten = _.,4,NoOp( number is ${number})
exten = _.,5,Hangup
I got the correct HANGUP value ( ie BUSY) but unfortunately I cannot
see the variables set on the originate command.
I wonder why not?
2)
I tried to use the Async=True parameter in the originate command and
magically
* make a goto to extent failed.
Unfortunately in such extension I can't see any value in HANGUPCAUSE
or
DIALSTATUS. Why?
3) this is the strategy I am currently using
In the manager api when I send the originate command I wait for the
response: if it is Error - reason : Originate failed
I assume that such a call fails for one of this reason: busy, chan
unavailable, wrong number.
Unfortunately this solution, even not giving detailed information about
the reason of failure doesn't work any more with a new pri line the
italian pri provider telecom installed recently. With this pri
interface , the originate command now return Success: Call queueed
even if the line is busy. This is a very strange behaviour probably due
to a pri different information not correctly catched.
So how can I know what is going on when a call fail?
Regards
--
Ing. Luca Casavola
(Technical Manager)
Software Products Italia
Milano / Roma / Firenze
tel. ++39 055/33651
fax ++39 055/340558
e-mail: [EMAIL PROTECTED]
www: http://www.softpi.it
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Re: [Asterisk-Users] spandsp problem
Steve Underwood wrote: Hi Michael, What happens with 0.0.2pre4? For most people that version gives better results than 0.0.1k. It seems to fix most of the quirks people have had. It didn't work with the HP3150. I have a new HP5510, and I'll try 0.0.2pre4 again. spandsp says we have more pages to send. The remote fax machine sends a disconnect message and hangs up. It sounds like the remote fax machine is at fault. Does it always fail with one machine, and succeed with another? Some machines can be set to receive a maximum of X pages. Could it be soemthing like that. You're absolutely correct. In a Hp-Sharp test (fax to fax) the Sharp disconnected after one page. I'll focus my efforts on the HP machines. Try to get 0.0.2 working. It works well enough for other people than it is about to loose its pre status. Ok, I'll do that. To eliminate T-1 timing issues, I removed both the fax machines from the channel bank and put them on a TDM22B card. Thanks for your help. -- Michael Welter Introspect Telephony Corp. Denver, Colorado US +1.303.674.2575 [EMAIL PROTECTED] www.introspect.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] SysMaster and GPL Violation
Uh ok...So when will Asterisk be a licensed product? Will it take the form of a Redhat sort of platform... Fedora with Redhat the pay me money side of the house? Just a simple question: When can we expect to see Asterisk the licensed as in paid for version ? Brandon Are you saying that those of us that are using the product should not be allowed to voice our opinions about its licensing, development and maintenance? That we should all just shut up and take whatever Mark co. give us? If that's the case, then this is most definitely NOT an open-source project at all. ___ 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] SysMaster and GPL Violation
Are you saying that those of us that are using the product should not be allowed to voice our opinions about its licensing, development and maintenance? That we should all just shut up and take whatever Mark co. give us? If that's the case, then this is most definitely NOT an open-source project at all. -Original Message- From Brandon Patterson Sent: Saturday, November 13, 2004 7:15 PM Uh ok...So when will Asterisk be a licensed product? Will it take the form of a Redhat sort of platform... Fedora with Redhat the pay me money side of the house? Just a simple question: When can we expect to see Asterisk the licensed as in paid for version ? Brandon Right now. As far as I know, you just need to contact Digium's sales department and negotiate a licensing agreement with 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
[Asterisk-Users] Re: getting callerid from spa3k to asterisk
if i have two sip contexts for my spa3k, on inbound and one outbound, e.g. [spa3k-in] type=friend host=dynamic port=5061 auth=md5 secret=pfui qualify=1000 canreinvite=yes context=ext-in42 [spa3k-out] type=peer auth=md5 secret=pfui username=outpass fromuser=outpass host=spa3k.bogus.com port=5061 nat=no canreinvite=yes context=ext-in42 and the spa3k's PSTN / Subscriber Information / User ID: = spack-in, the incoming connection from spa3k to * is being routed to the spa3k-out context, not the spa3-in context. see appended. i suspect this is a bug in * 1.0.1. i found the problem, or at least a work-around. if i reverse the order of the above two sip contexts, the incoming call is properly routed to the spa3k-in sip context as opposed to the wrong one, spa3k-out. my guess is that * is traversing a list and taking the first context which has the ip address and port it wants without checking the context name against the name which was received over the wire. so it depends on what order the contexts are inserted in the list. aii! 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] my asterisk drops connection when remote side puts me on hold?
I've got a TDM100P card with a fxo and fxs module in the US. I'm using kewlstart for all ports. I've noticed that when I make a call out from an analog phone out the POTS line that if after talking to the party I called (in this case the phone company itself) they put me on hold asterisk disconnects the call immediatly. I've looked around the web pages, but can't figure out what might be causing this and how to fix it - can anyone give me a clue? Thanks Steve Prior ___ 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] Cisco ATA and G729
Hi all, I am new to asterisk. I was able, but not without pain to install it on a FreeBSD box. I set up a cisco ATA 186 and the SJlabs softphone to work with the PBX. Three remarks: * On the SJphone, i use the GSM and the G711 (ulaw and alaw) codec. In the h323.conf file i enabled those codec. Everything works great!!! * However, when i set my Cisco ATA to G711, i can't hear any sound unless I press at least two or threekeys(any random keys). I am using the demo context of extension.conf file. Can that be due to a fast start problem? Anyone knows how to checkthe faststartcapabilities of an ATA 186? *Also when i set my ATA codec to g729 and in asterisk i allow=g729, i get a very low weird sound. What is that due to? I am guessing that i don't have the codec installed on the system. Is there an open source g729 codec available for FreeBSD? Any help will be very much appreciated, Thanks. Kido ___ 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] Remote answer not detected
Il dom, 2004-11-14 alle 00:13, DB ha scritto: Here's my a section of my simple extensions.conf snip exten = s,5,Dial(Zap/4/2326932|15) exten = s,6,Voicemail,u100 snip It works, but when the call is routed out on ZAP/4 (at priority 5), Asterisk seems to not realize the call is answered. After 15 seconds it proceeds to voicemail interrupting the call. Can anyone help? eh, perhaps with some details about your zap... ie what card? zaptel.conf? zapata.conf? matteo, still without divinatory powers Hi - thanks for the reply - here's that info: card is TDM22B zaptel.conf: == fxoks=1-2 # Make sure that the FXS(green) modules are closest fxsks=3-4 # This is for the FXO module(s) becaus defaultzone=us loadzone=us == zapata.conf: === [trunkgroups] [channels] switchtype=national signalling=fxo_ls rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes callprogress=yes progzone=us signalling=fxo_ks echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=yes echotraining=400 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=asreceived group=1 context=MD_line1 ; Points to the default context of your extensions.conf channel = 1 signalling=fxo_ks echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=yes echotraining=400 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=asreceived group=2 context=MD_line2 ; Points to the default context of your extensions.conf channel = 2 signalling=fxs_ks group=3 context=incoming_9141252 channel= 3 ; Again if you only have one FXO module remove the '-4' signalling=fxs_ks group=4 context=incoming_3493729 channel= 4 ; Again if you only have one FXO module remove the '-4' DB ___ 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] SysMaster and GPL Violation
Guys, in fact we will give an Applause to Sysmaster guys that are doing a great job in their products (world wide sales), this guys are doing money , and in this point is ,, Mark (and/or Digium is not receiving money for that). But Back to basics of Open source you can sale , modify , distribute and so on , so on so on,, BUT give the credits of the developer (in this case that is the only thing thery are not respecting) Now ,, Guys this is an oportunity to have a back benefit of that ,, because why I will pay a 150K usd for a NORFA (sysmaster new system)if for much less I can have an Asterisk up and running, In my point of view technically Asterisk (as it is right now) is GREAT lets take advance of that ,, instead of 100's of brains trying to make asterisk more and more and more features GUYS FOR GOD SAINT lets do it NEAT (GUI, Administration etc) Why dont Asterisk comunity opens a group for Asterisk Simplification This is the Opinion of a Non guru fellow ben available for a monthly donation for that proyect (what we need is to be Several like me to pay good programers and develop a GUI) and off course with a licencing that for all the donators and contributors the GUI has no Cost (including Digium) but for others will have a cost. AGAIN , this is only my point of view and I respect every coments about this :) Regards Humberto On Sat, 13 Nov 2004 19:30:06 -0700, Brian [EMAIL PROTECTED] wrote: Are you saying that those of us that are using the product should not be allowed to voice our opinions about its licensing, development and maintenance? That we should all just shut up and take whatever Mark co. give us? If that's the case, then this is most definitely NOT an open-source project at all. -Original Message- From Brandon Patterson Sent: Saturday, November 13, 2004 7:15 PM Uh ok...So when will Asterisk be a licensed product? Will it take the form of a Redhat sort of platform... Fedora with Redhat the pay me money side of the house? Just a simple question: When can we expect to see Asterisk the licensed as in paid for version ? Brandon Right now. As far as I know, you just need to contact Digium's sales department and negotiate a licensing agreement with 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 ___ 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] Cable for T1 connection: Crossover or straightthrough?
You've got a 50/50 shot. Try the crossover. http://www.cisco.com/en/US/products/hw/routers/ps233/products_tech_note0 9186a00800a3f09.shtml#topic2 It would be more helpful for you to send your /etc/zaptel.conf file and /etc/asterisk/Zapata.conf file. You should have something like the following for your zaptel.conf file: #zaptel.conf span=1,1,0,esf,b8zs loadzone = us defaultzone=us also do an %asterisk -r and send the info from the CLI that is show when you try to dial something. It's pretty intuitive output. Tim. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Saturday, November 13, 2004 6:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cable for T1 connection: Crossover or straightthrough? On November 13, 2004 12:11 pm, Malcolm Bader wrote: This is my second asterisk server but the first one with a T100P card. I connected it to the phone company(SBC) jack but have only a busy signal when calling the T1's number and nothing in the asterisk log files to indicate a connection. Do I need to use a crossover cable? Is the T100P's light Green or flashing red? Green means it sees the other side, and the other side sees it. i.e. the cabling is fine. Orange (well it's trying to be yellow) means that the other side can't see the T100P, but the T100P is seeing the other side. -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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.794 / Virus Database: 538 - Release Date: 11/10/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.794 / Virus Database: 538 - Release Date: 11/10/2004 ___ 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] Cisco ATA and G729
Im not sure about the G711 codec on the ATA, but I know you need to purchase the g729 from digium. http://www.digium.com/index.php?menu=asterisk_g729 pretty inexpensive at $10 each. Thats for concurrent connections to the server. Tim. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of kido noagbodji Sent: Saturday, November 13, 2004 8:37 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco ATA and G729 Hi all, I am new to asterisk. I was able, but not without pain to install it on a FreeBSD box. I set up a cisco ATA 186 and the SJlabs softphone to work with the PBX. Three remarks: * On the SJphone, i use the GSM and the G711 (ulaw and alaw) codec. In the h323.conf file i enabled those codec. Everything works great!!! * However, when i set my Cisco ATA to G711, i can't hear any sound unless I press at least two or threekeys(any random keys). I am using the demo context of extension.conf file. Can that be due to a fast start problem? Anyone knows how to checkthe faststartcapabilities of an ATA 186? *Also when i set my ATA codec to g729 and in asterisk i allow=g729, i get a very low weird sound. What is that due to? I am guessing that i don't have the codec installed on the system. Is there an open source g729 codec available for FreeBSD? Any help will be very much appreciated, Thanks. Kido --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.794 / Virus Database: 538 - Release Date: 11/10/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.794 / Virus Database: 538 - Release Date: 11/10/2004 ___ 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/AgentCallbackLogin Problems
I am having a few problems with my queue. I am using the AgentCallbackLogin
feature. When the call comes to the
user, it does not announce the call to the agent. It waits until you enter the #.
After you hit #. It will
play the queue-support announcement to the agent and tell them to press # if
they want the call.
The agent will have to hit # multiple times if they want the
call to come through.
Any suggestions?
Tim.
Extenstions.conf
exten =
999,1,AgentCallbackLogin(${CALLERIDNUM}|@default)
exten =
999,2,Hangup
exten =
1,1,Playback(welcome)
exten =
1,2,SetVar(QUEUE_PRIO=10)
exten =
1,3,Queue(cssupport|t||queue-support|120)
Queues.Conf
[cssupport]
music = random
announce =
queue-support
strategy = rrmemory
context = exitqueue
timeout = 45
retry = 10
wrapuptime=60
maxlen = 2
announce-frequency = 60
announce-holdtime = yes
announce-round-seconds = 10
queue-youarenext = queue-youarenext
queue-thereare = queue-thereare
queue-callswaiting = queue-callswaiting
queue-holdtime = queue-holdtime
queue-minutes =
queue-minutes
queue-seconds =
queue-seconds
queue-thankyou = queue-thankyou
queue-lessthan =
queue-less-than
joinempty = no
leavewhenempty = yes
member =
Agent/309
member =
Agent/311
CLI
-- Executing Playback(Zap/1-1, welcome) in new
stack
--
Playing 'welcome' (language 'en')
--
Executing SetVar(Zap/1-1, QUEUE_PRIO=10) in new
stack
--
Executing Queue(Zap/1-1,
cssupport|t||queue-support|120) in new stack
--
Started music on hold, class 'random', on Zap/1-1
--
Stopped music on hold on Zap/1-1
--
Playing 'queue-youarenext' (language 'en')
--
Told Zap/1-1 in cssupport their queue position (which
was 1)
--
Playing 'queue-thankyou' (language 'en')
--
Started music on hold, class 'random', on Zap/1-1
Nov 13 23:28:26 NOTICE[4125]: app_queue.c:749 wait_for_answer:
No one is answering queue 'cssupport'
Nov 13 23:28:36 NOTICE[4125]: app_queue.c:749 wait_for_answer:
No one is answering queue 'cssupport'
Nov 13 23:28:47 NOTICE[4125]: app_queue.c:749 wait_for_answer:
No one is answering queue 'cssupport'
Nov 13 23:28:57 NOTICE[4125]: app_queue.c:749 wait_for_answer:
No one is answering queue 'cssupport'
Nov 13 23:29:08 NOTICE[4125]: app_queue.c:749 wait_for_answer:
No one is answering queue 'cssupport'
---
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[Asterisk-Users] re: DVG-1120
Hello, I know the d-link units (DVG-1120 ATA and their router as well) are supposed to work well with asterisk...does anyone know if the units that come with ATT callvantage are locked, or can they be used w/asterisk or SER? And if they are locked, is it linksys no way out locking or a simple password thing? thanks, yair ___ 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
