[Asterisk-Users] Queue application problem
Hi, I've posted this problem before, but no response. I'm using iaxcomm for agents, and sometimes when there are agents waitng, incomming calls are not connected to agents for 20~30 seconds. In that case one agent is displayed in "Ringing" state. How can i avoid this situation? Any response is highly appreciated. Thanks. queue.conf -- [general] ;monitor-format = gsm [default] timeout = 4 maxlen = 0 music=default [que1] leavewhenempty=no music=default strategy=leastrecent joinempty=yes eventwhencalled=yes retry=1 CLI> shoq eueues -- que1 has 12 calls (max unlimited) in 'leastrecent' strategy (32s holdtime), W:0, C:883, A:411, SL:0.0% within 0s Members: IAX2/agent05 (dynamic) (Not in use) has taken no calls yet IAX2/agent11 (dynamic) (Not in use) has taken no calls yet IAX2/agent23 (dynamic) (Not in use) has taken no calls yet IAX2/agent16 (dynamic) (Not in use) has taken no calls yet IAX2/agent09 (dynamic) (Not in use) has taken no calls yet IAX2/agent06 (dynamic) (Not in use) has taken no calls yet IAX2/agent12 (dynamic) (Not in use) has taken 1 calls (last was 44 secs ago) IAX2/agent15 (dynamic) (Ringing) has taken no calls yet Callers: 1. Zap/38-1 (wait: 1:32, prio: 1) 2. Zap/49-1 (wait: 0:51, prio: 1) 3. Zap/51-1 (wait: 0:47, prio: 1) 4. Zap/52-1 (wait: 0:40, prio: 1) 5. Zap/39-1 (wait: 0:28, prio: 1) 6. Zap/41-1 (wait: 0:21, prio: 1) 7. Zap/53-1 (wait: 0:19, prio: 1) 8. Zap/54-1 (wait: 0:16, prio: 1) 9. Zap/43-1 (wait: 0:05, prio: 1) 10. Zap/55-1 (wait: 0:05, prio: 1) 11. Zap/44-1 (wait: 0:04, prio: 1) 12. Zap/58-1 (wait: 0:04, prio: 1) iax.conf -- [general] port=5036 disallow=all allow=alaw jitterbuffer=yes maxjitterbuffer=300 maxexccessbuffer=50 tos=0x04 qualify=no [agent00] type=friend username=agent00 secret=agent00 context=agent host=dynamic notransfer=yes __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Could someone look at channels/chan_zap.c
I'm banging my head against a brick wall trying to get CallerID
recognised in Australia.
I have CLID presentation enabled and I know that it works. I also have
distinctive ring tones enabled in zapata.conf
Around about line 5924 in channels/chan_zap.c is where the caller ID and
distinctive ring tone recognition starts for Bellcore FSK signalling
5924 } else if (p->use_callerid && p->cid_start ==
CID_START_RING) {
5925 /* FSK Bell202 callerID */
5926 cs = callerid_new(cid_signalling);
and at line 5961 there is this comment:
5961 /* Let us
detect callerid when the telco uses distinctive ring */
but what follows appears to have no resemblence to identifying CLID.
The problem is that I cannot see, or work out what is supposed to go on
after that. I am getting distinctive ring tones but an not getting CLID.
Any help out there, or anyone who can explain what the code is supposed
to be doing?
--
Howard.
LANNet Computing Associates - Your Linux people
--
When you just want a system that works, you choose Linux;
When you want a system that works, just, you choose Microsoft.
--
Flatter government, not fatter government;
Get rid of the Australian states.
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Re: [Asterisk-Users] asterisk -RT
On Mon, Oct 24, 2005 at 12:04:40PM +0800, Ronald Wiplinger wrote: > I use the command asterisk -RT to connect to a running asterisk box. > > There must be some changes to the latest CVS upgrade: > > 1. it does not remember anything anymore what I have done in the > previous connection. Why do you use 'asterisk -R' and not 'asterisk -r'? When you exit a CLI shell with ctrl-C it does not save the history. Is this a bug? Try quiting with 'quit', though I'm not sure if it has a different effect on asterisk -R . > I could reconnect to the asterisk box and with > arrow up I could see all my last commands, now no more. > > 2. I still cannot see any colors, > > > The original asterisk starts via > 31240 ?S 0:00 /bin/sh /usr/sbin/safe_asterisk > 31245 ?Sl 0:00 asterisk -vvvgpT -c Asterisk only checks for a color terminal at startup. Asterisk does not color messages in the remote CLI, but rather, when issuing the messages. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!
On 10/24/05, trixter aka Bret McDanel <[EMAIL PROTECTED]> wrote: > > And this should be why I am allowed to create a single PDF of the file > > to be distributed to our mirrors (which btw, I have done, so once the > > mirrors are updated, you will be able to get the file in a single PDF > > as opposed to multiple, individual chapters). > > > > Hope that makes everyone happy :) > > While the most important person to make happy is O'Reilly, I cant > imagine they would have a problem with this. But having everything as a > single pdf (which anyone could have done themselves since there are so > many tools to do this) I think is slightly better. Searching is easier > when you can search the whole thing instead of chapter by chapter. Plus > is makes it easier to put it on my PDA :) Ask, and ye shall receive! Thanks to Jeremy McNamara and William Suffill for updating their links so quickly. I'm sure Andrew Latham will get his updated as soon as he gets his email in the morning :) Thanks to all! You have access to both the multipart, and the single part versions of the file now. Available here: http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 Check Mirror 2 (USA) for now for the single part, just tested the links, and not quite updated on Jeremy's server yet (might be by the time you read this though). -- Leif Madsen - http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 Hardware Recommendations [ATTN: Digium marketing]
On 10/23/05, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: > On Sunday 23 October 2005 21:40, Kevin Bockman wrote: > > I agree on both points. I'm not sure if anyone from Digium actually > > reads the -users lists though. > > Kevin Fleming slogs through this list just as I do. It's a lot to keep up > with. Hell I think he even does it off the clock, as I do. > > I don't work for Digium, though. :-) > > -A. At Astricon I talked to a lot of the Digium guys and I was suprised at how many of them lurk on this list. Many of them read it and pass on the interesting posts to others in the company but most of them very rarely post on most of those issues. I learned that they do take comments on the list very seriously even though they may not comment on them publicly. A lot of us read the list and the forums off the clock, Hell it's 1:30AM here and I'm on the list again, what's wrong with me I need to get to sleep :) MATT--- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 Hardware Recommendations
On 10/23/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote: > don't know if there is much difference in "maintaining" a Sangoma > install vs a Digium install. I do know the setup may be somewhat > different, but that's a "one time" deal (sometimes). There really isn't much difference once you've gotten it installed. With the Sangoma cards it's just another script to run before you start Asterisk > As far as the Digium card, I don't know if all the TE2XXP have the > newer firmware. I do wish I could trade my TE4XXP for ones with the > new firmware. I will need to call Digium on that one. TE2XXP cards do have the v2 firmware just like the new TE4XXP cards. You can get your TE4XXP cards upgraded by Digium for just the cost of mailing it in to them. I've had 6 cards upgraded already so far, and as I can bring other servers down I will send their cards in to be upgraded too. Just call Digium and get an RMA number for a firmware upgrade. You should have your card back within a week of sending it. > Before the firmware upgrade, I would think there was a more > significant reason to go with Sangoma (from a technical PoV). But > now I guess it's a different story and for the amount of traffic > I'm expecting, I don't know if I'll realize the potential performance > difference between Sangoma and Digium, as explained in your blog entry. True, and now the technical performance difference between the two brands is minimal. > I am definitely pro supporting the Asterisk cause and if that means > that a % of my Digium purchase would go to it, I'd gladly do it. > Anyway, the price difference between the two is less than $80. There is a significant profit on these cards for both companies. The actual manufacturing and support costs for these cards is just a fraction of what they charge. It's best illustrated by the fact that govarion.com makes the old tor2 design cards and sells them new for less than half the cost of either the Digium or Sangoma cards. > The only thing I wished was that the Digium cards worked in 3.3V and > 5V motherboards without having to specify which one you are going to > deploy it on. I got somewhat screwed on the TE410P because of that > reason :( That's an issue with the chipset that Digium chose to use for the cards. I do believe that if you explain your issue with it that Digium may exchange your card for an TE405P when you send it in for an upgrade. If you get the right support person that is. > The warranty issue is a big difference. Why couldn't Digium compete > on that one? It's only a business/marketing decision, not a technical > one anyway. I agree, I had a T400P card die earlier this year. I had purchased it about 2 and a half years before and I called Digium about warranty support. The support person I talked to (don't remember his name) said that they didn't have any replacements for the T400P but I could get a discount on a new card if I traded it in. The card is currently serving as a $1500 paperweight on my desk. I do need to mention that none of the other Digium(12) or Sangoma(4) cards that I've bought have failed in any way in over the last 3 years. Sangoma recently raised their warranty on new cards from 3 to 5 years a couple months ago. I called them and asked for a clarification as to why and they said it was because they almost never got cards back and if it helps sell their cards they will stand by them for 5 years instead of just 3. As for Digium, when I was at Astricon last week I did hear rumours of retail-packaged Digium cards that would come with a 5-year warranty at a higher cost. They are currently selling a retail analog card like this and I was told to expect the same option for the T1/E1 cards. So I guess stay tuned and see if they follow through on that. > Anyway, thanks for your input. You're welcome, let me know how you decide. MATT--- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!
On 10/24/05, Wai-Sun Chia <[EMAIL PROTECTED]> wrote: > Regarding the pdfs, I seem to be missing some pages between the > COPYRIGHT.pdf (page iv) and foreword.pdf (page ix).. > > Is this one of those "blank page intentionally inserted here" moments > or is something really dropped off here? Hrmm... flipping through my hard copy, that seems to be the table of contents. I'll see about getting an updated version of it and adding it to the PDF. For now, the table of contents can be found at http://www.oreilly.com/catalog/asterisk/toc.html Thanks! -- Leif Madsen - http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!
On 10/24/05, trixter aka Bret McDanel <[EMAIL PROTECTED]> wrote: > > And this should be why I am allowed to create a single PDF of the file > > to be distributed to our mirrors (which btw, I have done, so once the > > mirrors are updated, you will be able to get the file in a single PDF > > as opposed to multiple, individual chapters). > > > > Hope that makes everyone happy :) > > While the most important person to make happy is O'Reilly, I cant > imagine they would have a problem with this. But having everything as a > single pdf (which anyone could have done themselves since there are so > many tools to do this) I think is slightly better. Searching is easier > when you can search the whole thing instead of chapter by chapter. Plus > is makes it easier to put it on my PDA :) I agree, and can see no reason why I can't do just that :) Will let you know once the mirrors update. -- Leif Madsen - http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!
Regarding the pdfs, I seem to be missing some pages between the COPYRIGHT.pdf (page iv) and foreword.pdf (page ix).. Is this one of those "blank page intentionally inserted here" moments or is something really dropped off here? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!
On Mon, 2005-10-24 at 01:00 -0400, Leif Madsen wrote: > However, I was basically just erring on the side of caution -- > re-distribution of the works is permitted (this is why we are allowed > to have so many mirrors!). I would assume repackaging of the work > would be considered part of the "redistribution" aspect of that > allowance. > > And this should be why I am allowed to create a single PDF of the file > to be distributed to our mirrors (which btw, I have done, so once the > mirrors are updated, you will be able to get the file in a single PDF > as opposed to multiple, individual chapters). > > Hope that makes everyone happy :) While the most important person to make happy is O'Reilly, I cant imagine they would have a problem with this. But having everything as a single pdf (which anyone could have done themselves since there are so many tools to do this) I think is slightly better. Searching is easier when you can search the whole thing instead of chapter by chapter. Plus is makes it easier to put it on my PDA :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!
On 10/24/05, trixter aka Bret McDanel <[EMAIL PROTECTED]> wrote: > I dont know who has access to post blog entries at creativecommons.org > but Glenn Otis Brown wrote in support of repackaging saying it helps the > original author (in his example songwriters). > http://creativecommons.org/weblog/entry/3729 > > Now, there is a clause that applies to you (Lief) specifically in the > license. > "Any of these conditions can be waived if you get permission from the > copyright holder." > > Unless that was assigned to orielly, which is may be the case. You are correct -- copyright is held by O'Reilly Media Inc. However, I was basically just erring on the side of caution -- re-distribution of the works is permitted (this is why we are allowed to have so many mirrors!). I would assume repackaging of the work would be considered part of the "redistribution" aspect of that allowance. And this should be why I am allowed to create a single PDF of the file to be distributed to our mirrors (which btw, I have done, so once the mirrors are updated, you will be able to get the file in a single PDF as opposed to multiple, individual chapters). Hope that makes everyone happy :) -- Leif Madsen - http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!
On Mon, 2005-10-24 at 00:18 -0400, Leif Madsen wrote: > The problem here is that you CAN'T make derivitive works. I think you > have the wrong license (however I can't verify since I can't get to > the Creative Commons website right now). There is no derivitive works > allowed of the book (and no commercial works either). > Hmm.. well creative commons forbids commercial works being made from anything protected under it. So that part at least is the same. The license linked to from http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 is http://creativecommons.org/licenses/by-nc-nd/2.0/ca/ so that is a slightly different license. The creativecommons.org page does state: Your fair use and other rights are in no way affected by the above. And I believe it is highly regarded as fair use (although I dont have legal cites I think its certainly within the spirit of the creative commons doctrine to allow repackaging). I dont know who has access to post blog entries at creativecommons.org but Glenn Otis Brown wrote in support of repackaging saying it helps the original author (in his example songwriters). http://creativecommons.org/weblog/entry/3729 Now, there is a clause that applies to you (Lief) specifically in the license. "Any of these conditions can be waived if you get permission from the copyright holder." Unless that was assigned to orielly, which is may be the case. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trying to clarify ideas about spands, libtiff & FC4
I've been using spandsp 0.0.2pre18 and 0.0.2pre21 with libtiff 3.7.3, Asterisk 1.0.7 (now 1.0.9) and the Linux kernel 2.6.11.5 kernel with results that are good enough for me (I'm using fax over IP with the G.711 codec). On Sunday 23 October 2005 13:23, Carlos Alperin wrote: > I spent more than 3 weeks, with some little help of people that belongs to > this forum, and after try differents combinations of versions this is my > conclusion: > > > > I tried RH9, FC4 & FC4 64 > > I tried with CVS 1.0.2, and Stable 1.0.9 > > I tried with spandsp 0.0.2pre18, 0.0.2pre20 & 0.0.2pre21 > > Libtiff 3.5.7 & libtiff devel 3.5.7 > > Libtiff 3.7.1 & libtiff devel 3.7.3 (I couldn't find 3.7.1) > > > > My conclusion is: > > > > If I need to be able to use fax with Spandsp, app_rxfax.c & app_txfax.c > with libtiff 3.5.7 (and libtiff devel 3.5.7) there is no way to do that on > FC4 (get conflict with GTK2+) > > So it looks like I have to go back to RH9 and at least upgrade to kernel > 2.4.31, and try again. > > > > This is under the presumption that Spandsp, & the rest are going to work. > (Looking at the forum, that is not a 100% fact). > > > > It should be a way to save us a lot of time, if somebody can unify all the > requeriments on each OS, so we can decide before to start which direction > to follow. > > > > The reason for RH9 & FC4 is because they're more familiar. But if someone > can show me a working configuration, I don't hesitate to move the platform. > > > > By the way, the 64 bits platform still looks to be very unstable and not so > fast to implement with Asterisk. > > > > To the digium support: I understand that your recommendation is to go to > 2.6 kernel, but if I need to run spandsp, how to do that without libtiff > 3.5.7. > > > > The general experience is libtiff 3.7.1 locks the asterisk when the machine > boots. > > > > Please feel free to send every kind of disappointments opinions. That is > going to feel me much better that no answers. > > (Even if you can show me how stupid I was doing all kind of mistakes) > > > > Regards, > > > > Carlos Alperin -- Juan Jose Comellas ([EMAIL PROTECTED]) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!
On 10/24/05, trixter aka Bret McDanel <[EMAIL PROTECTED]> wrote: > On Sun, 2005-10-23 at 23:57 -0400, Leif Madsen wrote: > > On 10/20/05, Darrick Hartman <[EMAIL PROTECTED]> wrote: > > > Is there any reason why the book wasn't released as a single pdf rather > > > than the individual chapter pdf's? Using pdftk, I merged the pdfs back > > > into a single document (11mb), then zipped it back up. Is there any > > > restriction that would prevent me from mirroring this as a complete pdf > > > rather than individual pdfs? > > > > Hrmmm... that is a good question, because I guess technically you're > > not changing it. However, for now, lets just leave it as be. Perhaps > > tomorrow I will speak with Jared and the people providing the mirrors > > and ask them to update the file with the PDF after I have merged it. > > As it is under the creative commons license it would be those terms that > would limit what you can and cannot do. > > http://creativecommons.org/licenses/by-nc-sa/2.0/legalcode > You are free to make derivative works, however it requires you to > release under the creative commons license any such works. So whether > repackaging it is deemed to be a derrative work or not you are free to > do that. If its not deemed a derrative work then there wouldnt be a > problem either. > > Repackaging shouldnt really be covered as derrative work anyway because > it doesnt change the content which is what is protected, not the > packaging method. So long as the content is 100% the same you dont have > to do anything, but if you do change the content (adding, removing, > correcting, etc) then you have to rerelease it as creative commons, > which it sounds like you wanted to keep it the same anyway so that isnt > an issue. The problem here is that you CAN'T make derivitive works. I think you have the wrong license (however I can't verify since I can't get to the Creative Commons website right now). There is no derivitive works allowed of the book (and no commercial works either). However, I do agree that repackaging it as a single PDF probably doesn't constitute a change, and thus is fine as long as the original work is unaltered 100%. Either way, I'm going to go and spend a few minutes now and repackage the PDF for y'all :) -- Leif Madsen - http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!
On Sun, 2005-10-23 at 23:57 -0400, Leif Madsen wrote: > On 10/20/05, Darrick Hartman <[EMAIL PROTECTED]> wrote: > > Is there any reason why the book wasn't released as a single pdf rather > > than the individual chapter pdf's? Using pdftk, I merged the pdfs back > > into a single document (11mb), then zipped it back up. Is there any > > restriction that would prevent me from mirroring this as a complete pdf > > rather than individual pdfs? > > Hrmmm... that is a good question, because I guess technically you're > not changing it. However, for now, lets just leave it as be. Perhaps > tomorrow I will speak with Jared and the people providing the mirrors > and ask them to update the file with the PDF after I have merged it. As it is under the creative commons license it would be those terms that would limit what you can and cannot do. http://creativecommons.org/licenses/by-nc-sa/2.0/legalcode You are free to make derivative works, however it requires you to release under the creative commons license any such works. So whether repackaging it is deemed to be a derrative work or not you are free to do that. If its not deemed a derrative work then there wouldnt be a problem either. Repackaging shouldnt really be covered as derrative work anyway because it doesnt change the content which is what is protected, not the packaging method. So long as the content is 100% the same you dont have to do anything, but if you do change the content (adding, removing, correcting, etc) then you have to rerelease it as creative commons, which it sounds like you wanted to keep it the same anyway so that isnt an issue. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk -RT
I use the command asterisk -RT to connect to a running asterisk box. There must be some changes to the latest CVS upgrade: 1. it does not remember anything anymore what I have done in the previous connection. I could reconnect to the asterisk box and with arrow up I could see all my last commands, now no more. 2. I still cannot see any colors, The original asterisk starts via 31240 ?S 0:00 /bin/sh /usr/sbin/safe_asterisk 31245 ?Sl 0:00 asterisk -vvvgpT -c Any ideas? bye Ronald Wiplinger ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!
On 10/20/05, Darrick Hartman <[EMAIL PROTECTED]> wrote: > Leif Madsen wrote: > > PS: If the Asterisk Documentation Project website becomes slow due to > > the number of people accessing it at once, we appoligize and > > appreciate your patience. For those of you who are able to obtain the > > full copy, please consider helping us out by creating mirrors and > > torrents and posting them to the list by replying to this thread. > > Thanks! > > > > > Is there any reason why the book wasn't released as a single pdf rather > than the individual chapter pdf's? Using pdftk, I merged the pdfs back > into a single document (11mb), then zipped it back up. Is there any > restriction that would prevent me from mirroring this as a complete pdf > rather than individual pdfs? Hrmmm... that is a good question, because I guess technically you're not changing it. However, for now, lets just leave it as be. Perhaps tomorrow I will speak with Jared and the people providing the mirrors and ask them to update the file with the PDF after I have merged it. Thanks! -- Leif Madsen - http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Trying to clarify ideas about spands, libtiff & FC4
By the way, After follow all the rules, RH9 with Libtiff 3.5.7, Libtiff-devel 3.5.7, OpenSSL-Devel, Readline41, Ncurses4, Ncurses C++ Devel, SOX, Asterisk 1.0.9 Stable, Spandsp 0.0.2pre21, App_txfax.c & app_rxfax.c dated October 21, 2005. This time everything was smoth and nice. Sunday 23:30 the compilation ended after redo the full machine at least more than 10 times. I got it no errors. NOW, CAN SOMEONE EXPLAIN ME WHY ASTERISK DIES WITH CODE 1? Sorry, I'm close to drop this to the garbage can. Carlos ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Fwd: Re: [Asterisk-Users] Problem with asterisk 1.0.9 and sip and dtmf
I don't know if this noise is related to our noise.Note: forwarded message attached.--- Begin Message --- Silence Supp Enable is No. Sergey Okhapkin wrote: Check if you have Silence Suppression disabled on PSTN line of spa-3000 (admin/advansed/PSTN line). On Sat, 2005-10-22 at 18:05 -0400, Mike Bernson wrote: I have asterisk running with sipura 3000 connect to PSTN and sipura 2000 connected to phones. On inbound calls I am getting what sounds like DTMF tone when someone is talking on the PSTN side of the phone. It sound like someone is hitting key on the phone while talking. Is there any way to stop this from happing. Here is the PSTN and one ext from the sip.conf PSTN line [199] username= type=friend secret= record_out=Adhoc record_in=Adhoc qualify=yes port=5061 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=incomming canreinvite=no ext [206] username= type=friend secret= record_out=Adhoc record_in=Adhoc qualify=yes port=5061 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-sip canreinvite=no ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users--- End Message --- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Trying to clarify ideas about spands, libtiff & FC4
Good for you Slackware 10.x (exactly 10.?) to be more accurate? What version of Asterisk? What version of Spandsp? What version of Libtiff? What version of Libtiff-devel? And the million dollars question: Is the fax working? (Lets say more than 50% of the cases?) Thanks for your info. I'm trying to help myself & a lot of people like me that couldn't make the fax working. Thanks, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Derek Whitten Sent: Sunday, October 23, 2005 1:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Trying to clarify ideas about spands,libtiff & FC4 Haven't had any issues here with slackware 10.x On Sunday 23 October 2005 09:23 am, Carlos Alperin wrote: > I spent more than 3 weeks, with some little help of people that belongs to > this forum, and after try differents combinations of versions this is my > conclusion: > > > > I tried RH9, FC4 & FC4 64 > > I tried with CVS 1.0.2, and Stable 1.0.9 > > I tried with spandsp 0.0.2pre18, 0.0.2pre20 & 0.0.2pre21 > > Libtiff 3.5.7 & libtiff devel 3.5.7 > > Libtiff 3.7.1 & libtiff devel 3.7.3 (I couldn't find 3.7.1) > > > > My conclusion is: > > > > If I need to be able to use fax with Spandsp, app_rxfax.c & app_txfax.c > with libtiff 3.5.7 (and libtiff devel 3.5.7) there is no way to do that on > FC4 (get conflict with GTK2+) > > So it looks like I have to go back to RH9 and at least upgrade to kernel > 2.4.31, and try again. > > > > This is under the presumption that Spandsp, & the rest are going to work. > (Looking at the forum, that is not a 100% fact). > > > > It should be a way to save us a lot of time, if somebody can unify all the > requeriments on each OS, so we can decide before to start which direction > to follow. > > > > The reason for RH9 & FC4 is because they're more familiar. But if someone > can show me a working configuration, I don't hesitate to move the platform. > > > > By the way, the 64 bits platform still looks to be very unstable and not so > fast to implement with Asterisk. > > > > To the digium support: I understand that your recommendation is to go to > 2.6 kernel, but if I need to run spandsp, how to do that without libtiff > 3.5.7. > > > > The general experience is libtiff 3.7.1 locks the asterisk when the machine > boots. > > > > Please feel free to send every kind of disappointments opinions. That is > going to feel me much better that no answers. > > (Even if you can show me how stupid I was doing all kind of mistakes) > > > > Regards, > > > > Carlos Alperin -- -BEGIN GEEK CODE BLOCK- Version: 3.1 GCS/GIT> d-@ s+:+ a? C+++ BLHIS$ U+++ P+> L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+> h r+++ y --END GEEK CODE BLOCK-- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Trying to clarify ideas about spands, libtiff& FC4
Same question that before: Mandrake/Mandrive 2.6.13.4 (thanks for the info) What version of Asterisk? What version of Spandsp? What version of Libtiff? What version of Libtiff-devel? And the million dollars question: Is the fax working? (Lets say more than 50% of the cases?) Thanks for your info. I'm trying to help myself & a lot of people like me that couldn't make the fax working. Thanks, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle Sent: Sunday, October 23, 2005 1:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Trying to clarify ideas about spands, libtiff& FC4 Carlos Alperin wrote: > I spent more than 3 weeks, with some little help of people that > belongs to this forum, and after try differents combinations of > versions this is my conclusion: > > > > Please feel free to send every kind of disappointments opinions. That > is going to feel me much better that no answers. > > (Even if you can show me how stupid I was doing all kind of mistakes) > > I've got good experience with Mandrake/Mandriva with a 2.6.13.4 kernel Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax softphone
How about any IAX softphones for the pocket pc platform? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with Festival...
Hello there, I'm having problems with Festival text-to-speech generator. Apparently, Asterisk connects to the Festival server, but no audio is generated. Does anybody know: a) if Asterisk is compatible with Festival 1.4.2? b) it is possible to download new voices for text2wave (for the above version of Festival)? If so, how? If you are not happy with Festival, which tts would you use? Is it free? Thanks in advance, Leo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 Hardware Recommendations [ATTN: Digium marketing]
On Sunday 23 October 2005 21:40, Kevin Bockman wrote: > I agree on both points. I'm not sure if anyone from Digium actually > reads the -users lists though. Kevin Fleming slogs through this list just as I do. It's a lot to keep up with. Hell I think he even does it off the clock, as I do. I don't work for Digium, though. :-) -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 Hardware Recommendations [ATTN: Digium marketing]
Waldo Rubinstein wrote: The only thing I wished was that the Digium cards worked in 3.3V and 5V motherboards without having to specify which one you are going to deploy it on. I got somewhat screwed on the TE410P because of that reason :( The warranty issue is a big difference. Why couldn't Digium compete on that one? It's only a business/marketing decision, not a technical one anyway. I agree on both points. I'm not sure if anyone from Digium actually reads the -users lists though. Kevin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Filenaming for Incoming Queue Call Recordings (Reposted from "changing the filename of incoming call recordings")
KRTorio wrote:
Where in queues.conf? Could you please point out where? Thanks
Check /usr/src/asterisk/configs/queues.conf.sample if you have updated.
Now to state the obvious:
; Calls may be recorded using Asterisk's monitor resource
; This can be enabled from within the Queue application, starting recording
; when the call is actually picked up; thus, only successful calls are
; recorded, and you are not recording while people are listening to MOH.
; To enable monitoring, simply specify "monitor-format"; it will be
disabled
; otherwise.
;
; You can specify the monitor filename with by calling
;Set(MONITOR_FILENAME=foo)
; Otherwise it will use MONITOR_FILENAME=${UNIQUEID}
;
; monitor-format = gsm|wav|wav49
;
; If you wish to have the two files joined together when the call ends,
set this
; to yes.
;
; monitor-join = yes
Kevin
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Re: [Asterisk-Users] T1 Hardware Recommendations
Matt, Thanks for the reply. I had read your blog entry a few days ago and that's when I started doubting whether or not to go with Sangoma. I've only heard good things of Sangoma. However, I only have experience with TE410 and because of my limited experience with Asterisk, I was a bit concerned with switching to another card vendor from the one I've gotten used to. From a practical point of view, I don't know if there is much difference in "maintaining" a Sangoma install vs a Digium install. I do know the setup may be somewhat different, but that's a "one time" deal (sometimes). As far as the Digium card, I don't know if all the TE2XXP have the newer firmware. I do wish I could trade my TE4XXP for ones with the new firmware. I will need to call Digium on that one. Before the firmware upgrade, I would think there was a more significant reason to go with Sangoma (from a technical PoV). But now I guess it's a different story and for the amount of traffic I'm expecting, I don't know if I'll realize the potential performance difference between Sangoma and Digium, as explained in your blog entry. I am definitely pro supporting the Asterisk cause and if that means that a % of my Digium purchase would go to it, I'd gladly do it. Anyway, the price difference between the two is less than $80. The only thing I wished was that the Digium cards worked in 3.3V and 5V motherboards without having to specify which one you are going to deploy it on. I got somewhat screwed on the TE410P because of that reason :( The warranty issue is a big difference. Why couldn't Digium compete on that one? It's only a business/marketing decision, not a technical one anyway. Anyway, thanks for your input. - Waldo On Oct 23, 2005, at 8:24 PM, Matt Florell wrote: Hello, Watch out there, that's a very touchy issue. I'll try to lay out the technical and non-technical points of view. First the purely technical point of view: The Digium TE210P/TE205P is basically the TE4XXP(quad card) with only two ports included instead of four. It uses almost exactly the same firmware and drivers as the TE410P which as of a few months ago was greatly improved, see my review: http://astguiclient.blogspot.com/2005/09/digium-405p-v2-review.html The Sangoma a102u dual T1/E1 card requires additional software to be installed on the server to use it, and that software also needs to be configured, which does mean that it takes longer to setup. The sangoma card does fit straight-up in a 2U server chassis while the Digium dual card would require a horizontal PCI riser card to do the same. Also, (again PURELY TECHNICAL HERE) The sangoma card comes with a 5 year warranty while the Digium card comes with a 2 year warranty and the Sangoma card is guaranteed to work on all PCI2-compliant motherboards while Digium has a list of motherboards that their cards will not work in. Also, the Sangoma card will work in 3v and 5v PCI slots while the Digium cards are 5v or 3v specific. Now here is the non-technical comparison: Both cards work well and have been used in production servers running steadily for months at a time and both companies offer free setup support. However, I will be very quietly cursed if I do no mention that much more of your purchase price of the Digium cards goes towards the further development of Asterisk. Hardware sales make up a large portion of Digium's revenues and are the primary reason why Asterisk and Digium have grown so much over the last 3 years. There is much more to the story, but I've rambled on long enough and I hope I've given you enough information to answer your question for yourself. MATT--- On 10/23/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote: Hi guys, Just a quick question. I need to buy a dual T1 card and I'm debating between TE210P or the Sangoma A102u. Any recommendations? Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Filenaming for Incoming Queue Call Recordings (Reposted from "changing the filename of incoming call recordings")
Your answer was in queues.conf that's why you only got 1 reply. Where in queues.conf? Could you please point out where? Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 Hardware Recommendations
Hello, Watch out there, that's a very touchy issue. I'll try to lay out the technical and non-technical points of view. First the purely technical point of view: The Digium TE210P/TE205P is basically the TE4XXP(quad card) with only two ports included instead of four. It uses almost exactly the same firmware and drivers as the TE410P which as of a few months ago was greatly improved, see my review: http://astguiclient.blogspot.com/2005/09/digium-405p-v2-review.html The Sangoma a102u dual T1/E1 card requires additional software to be installed on the server to use it, and that software also needs to be configured, which does mean that it takes longer to setup. The sangoma card does fit straight-up in a 2U server chassis while the Digium dual card would require a horizontal PCI riser card to do the same. Also, (again PURELY TECHNICAL HERE) The sangoma card comes with a 5 year warranty while the Digium card comes with a 2 year warranty and the Sangoma card is guaranteed to work on all PCI2-compliant motherboards while Digium has a list of motherboards that their cards will not work in. Also, the Sangoma card will work in 3v and 5v PCI slots while the Digium cards are 5v or 3v specific. Now here is the non-technical comparison: Both cards work well and have been used in production servers running steadily for months at a time and both companies offer free setup support. However, I will be very quietly cursed if I do no mention that much more of your purchase price of the Digium cards goes towards the further development of Asterisk. Hardware sales make up a large portion of Digium's revenues and are the primary reason why Asterisk and Digium have grown so much over the last 3 years. There is much more to the story, but I've rambled on long enough and I hope I've given you enough information to answer your question for yourself. MATT--- On 10/23/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote: > Hi guys, > > Just a quick question. I need to buy a dual T1 card and I'm debating > between TE210P or the Sangoma A102u. Any recommendations? > > Thanks, > Waldo > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600
On Sunday 23 October 2005 19:59, C F wrote: > Well, I don't think that that's what I hear when I enable it. It works > really nice for me. Chacon son gout. :-) if you are listening to someone and you start to talk (to interrupt say) their voice disappears immediately. Or if there's background noise on your side they'll cut in and out. For myself and my customers that's totally unacceptable. But then again I'm of German stock, we tend to have a more... agressive way of communicating. :-) -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600
Well, I don't think that that's what I hear when I enable it. It works really nice for me. On 10/23/05, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: > On Sunday 23 October 2005 18:30, C F wrote: > > Why? > > Because it sounds like ass. I (and my customers) are used to the full-duplex > nature of the telephone system. Half duplex sounds very unnatural. > > -A. > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600
On Sunday 23 October 2005 18:30, C F wrote: > Why? Because it sounds like ass. I (and my customers) are used to the full-duplex nature of the telephone system. Half duplex sounds very unnatural. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T1 Hardware Recommendations
Hi guys, Just a quick question. I need to buy a dual T1 card and I'm debating between TE210P or the Sangoma A102u. Any recommendations? Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with asterisk 1.0.9 and sip and dtmf
Silence Supp Enable is No. Sergey Okhapkin wrote: Check if you have Silence Suppression disabled on PSTN line of spa-3000 (admin/advansed/PSTN line). On Sat, 2005-10-22 at 18:05 -0400, Mike Bernson wrote: I have asterisk running with sipura 3000 connect to PSTN and sipura 2000 connected to phones. On inbound calls I am getting what sounds like DTMF tone when someone is talking on the PSTN side of the phone. It sound like someone is hitting key on the phone while talking. Is there any way to stop this from happing. Here is the PSTN and one ext from the sip.conf PSTN line [199] username= type=friend secret= record_out=Adhoc record_in=Adhoc qualify=yes port=5061 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=incomming canreinvite=no ext [206] username= type=friend secret= record_out=Adhoc record_in=Adhoc qualify=yes port=5061 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-sip canreinvite=no ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Testing AreskiCC
> When I try to create sip/iax friend from web interface it says > "Could not open buddy file '/etc/asterisk/additional_areskicc_sip.conf' > I tried creating the file manually without luck. Make sure the user your web-server runs as can write to and read from the file. > > Second I am unable to dial any phone nos after card verification. > As soon as the card is verified with the remaining balance it straight > forward tells invalid-digit, Make sure the length of the PIN entered is correct and that you have entered in a phone number. __ Yahoo! FareChase: Search multiple travel sites in one click. http://farechase.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600
Why? On 10/23/05, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: > On Sunday 23 October 2005 18:02, C F wrote: > > Sorry guys I forgot to mention that in my setup I always enable > > agressive in zconfig > > Yuck. I find the agressive echo canceller totally unacceptable. > > -A. > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iConnectHere (or DeltaThree) trunk settings
I've stumbled upon a very interesting phenomenon. In setting up a trunk from iConnectHere (ich) I mistakenly input type=from-pstn type=friend This, in [EMAIL PROTECTED] using AMP. No "User Context" entries are made and a DID entry (with the full 11 digit number) is in the DID settings. I now receive calls incoming from ich.. For the past month (or more) I have not been able to set this up. Needless to say, I tried removing this 'fluke' - replacing type with context. Anything that looks right results in my not receiving calls from ich.. The entry looks like Type=from-pstn&friend In the trunks input form, even if you change it to two "type" entries on two separate lines.. I suspect this is AMP. But the .conf file shows them on two separate lines. Could someone tell me what type=from-pstn is ? AAH1.5 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600
On Sunday 23 October 2005 18:02, C F wrote: > Sorry guys I forgot to mention that in my setup I always enable > agressive in zconfig Yuck. I find the agressive echo canceller totally unacceptable. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One SIP dead, all SIP dead -- sipmedia gone?
The CEO of SIPMEDIA lives down the road from me. I will find out what's going on, and report back. What I do know is that it's a real company and not one running behind a lemonade stand, backed by major players in the industry. I can of course not give out more than this on the structure of the company, since all I know is from a personal basis and not from gossip. I do not have and never had any business connection with him/them. But my clients that do use him are very very happy. On 10/23/05, Jay Milk <[EMAIL PROTECTED]> wrote: > > -Original Message- > > From: Joe Greco [mailto:[EMAIL PROTECTED] > > > > > juju to have all your IPs in one block. > > > > Actually, they didn't have all their DNS servers in one block. > > Touche... I didn't even double-check, just assumed. > > > while, we ran with sequentially numbered servers that were in > > completely different cities, thanks to the magic of OSPF and > > True. > > > > I don't think they were reselling, and I actually thought I had a > > ... > > > availability, they've been the best provider out of all I tried > > > (Vonage, Broadvoice, voicepulse, iax.cc, ... ) > > > > Best price is occasionally a bad sign. > > Granted. But I did say "between pricing and availability". Their BYOD > plan is $5/month and includes 60 outgoing minutes, "unlimited" incoming, > and sip-only access. After explaining that I wanted several lines and a > discount, they asked me quite bluntly what kind of incoming volume I > expected; when they learned that I'm a residential user with minimal > volume, they allowed me to drop the included outgoing minutes and > adjusted their pricing south. Seems fair and well-calculated to me. I > could have gotten cheaper lines from the yokels at sixtel/iax.cc... > Well, if they ever fixed their "instant" 3-months ordering system. > > For sipmedia, good response, quick set-up, proper fraud-protection... > All around good, just some recurring problems with their upstream > (bandwidth) providers from what I can tell. They're reselling L3. > Three failures in seven months isn't great, and the response on this > last one was lacking -- but they're still doing better than the local > utility, which leaves us an average of eight hours/month without > electricity :) > > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600
Sorry guys I forgot to mention that in my setup I always enable agressive in zconfig On 10/23/05, Andrew Kohlsmith <[EMAIL PROTECTED]> wrote: > On Sunday 23 October 2005 16:20, [EMAIL PROTECTED] wrote: > > Followup, I set a -2.0 gain from my asterisk t1 > pbx, and echo seems > > mostly gone. > > > > A note, I also turned on the aggressive suppressor in zconfig.h > > It's the turning on the agressive mode that did it. Agressive mode works by > turning the zaptel channels into half-duplex channels. That's not a side > note. :-) > > -A. > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hardware setup question
I have just a quick setup question about how some of you have hardware setup. Basically, for a system that has an average volumes of calls in an office setting, are you using one or two network cards. I am just wondering if it owuld be any advantage to having one NIC for the extensions and one NIC for your trunks. Robert ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Terrible echo with Te110P and Adit 600
On Sunday 23 October 2005 16:20, [EMAIL PROTECTED] wrote: > Followup, I set a -2.0 gain from my asterisk t1 > pbx, and echo seems > mostly gone. > > A note, I also turned on the aggressive suppressor in zconfig.h It's the turning on the agressive mode that did it. Agressive mode works by turning the zaptel channels into half-duplex channels. That's not a side note. :-) -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Terrible echo with Te110P and Adit 600
Followup, I set a -2.0 gain from my asterisk t1 > pbx, and echo seems mostly gone. A note, I also turned on the aggressive suppressor in zconfig.h Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gregory Wiktor - ADCom Corp. Sent: Saturday, October 22, 2005 11:07 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Terrible echo with Te110P and Adit 600 I have a similar setup... I set the canceller on the incoming PSTN lines, but turn it off on the FXS. I have no local internal echo over the t1, but moderate over the PSTN. I managed to tweak it a little and most of my outbound (local side) echo is minimized, but still there a little. I have no incoming echo. You mind elaborating on where you are getting the echo? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wright Sent: Tuesday, October 18, 2005 10:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Terrible echo with Te110P and Adit 600 8 POTS lines - Adit 600 - TE110P - Dell Precision 530 Dual Xeon running 1.0.9 and 1.2 (tried both) The echo is insurmountable. I have tried everything, and the pots lines are clean. If I go from an FXO on the Adit 600 straight to an FXS, I get no echo from an analog phone. I put an 128ms T1 echo canceller in between the adit and the TE110P, and the echo was still horrible. I finally disabled the Zapata echo cancellerand WHAMMO! It's perfect now. The TE110P is on it's own IRQ.. and the machine has PLENTY of horsepower. Any ideas so I don't have to spend $1000 on an echo canceller? -Darren ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linksys pap2 behind Linksys RT31
Sebastian Milioto wrote: Hi all, I have a public ip in Linksys RT31 (2 FXS port + 3 swtich port + 1 uplink port). I want to add behind it, a Linksys pap2 (uplink port + 2 FXS port) with private ip. I understand that I have to configure Port forwarding or port triggering (really I'm not sure which one). Is someone already configured this toplogy? Could you help me with that, please? I have had very weird experiences with running any kind of SIP device behind those router/ata combo devices. I've tried both a SPA-200 and a Polycom 501 with very odd results (ie. calls that should go to router/ata's FXS actually go to Polycom *as well*, and when you answer one the other keeps ringing. If you answer the other you end up in a really crappy quality 3 way call. Hanging up either device will terminte the call altogether) Just a heads up... Trev ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Adit 3104 configuration
I just installed several 3104s in S. Calif. Didn't have any problems--I was able to call from one line to another on the same unit and between lines on different units. Jerry Jones wrote: Has anyone been able to get the 3104 to register more than one line correctly? It seems to work OK for the first line, but as soon as I turn on more than one it appears that only the last one is actually registering corectly. The 3104 sometimes indicates the line is registered, but * says not. This looks like a very useful unit and would really like to get it to work. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue Join Event
Tressler, Joshua A wrote: Saul, What you are suggesting follows along the lines of what I am currently trying however I have determined that if the incoming call has no callerid, then the channel name is just Zap/1-1/ . For some reason asterisk doesn't even add the - to the end of the channel name In zap channels :) zap/1-1 is the - u don't will get zap/1-2 at least i have a 300 phones system runing and never got 1... b/c only will happen when u send another call in the same channel... if the user hangs up :) u will get another event.. i think i can look here for that event too if u need .. and the same will happen if the agent hangup u get another event etc etc Event: Leave Privilege: call,all Channel: SIP/s-f36c Queue: operator Count: 0 Server: asterisk1 Database problems can be reduced to the min.. u can creat an id unique for u when u move to the db but in the asterisk enviroment the only way u have to follow the call is the channel...and if the channel is renamed b/c u got a park call or something u will get the rename event regard Saul My concern is that we could get a call that wouldn't go completely through the queue (aka, the user hangs up, or a db problem) and then an hour later get another private call on the same Zap/1-1/ channel and then we could have an issue of the uniqueness of the record. Do you follow my scenario? I really thought that this may work until the above problem. Do you have any to this issue? Again, I appreciate your help with this. Thanks again, Josh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Queue Join Event
Saul, What you are suggesting follows along the lines of what I am currently trying however I have determined that if the incoming call has no callerid, then the channel name is just Zap/1-1/ . For some reason asterisk doesn't even add the - to the end of the channel name My concern is that we could get a call that wouldn't go completely through the queue (aka, the user hangs up, or a db problem) and then an hour later get another private call on the same Zap/1-1/ channel and then we could have an issue of the uniqueness of the record. Do you follow my scenario? I really thought that this may work until the above problem. Do you have any to this issue? Again, I appreciate your help with this. Thanks again, Josh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Saul Diaz Sent: Sunday, October 23, 2005 12:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Queue Join Event Tressler, Joshua A wrote: > I did a quick Google search of the lists and I hope that I am not > asking a question that has already been answered recently. > > I have been working on a interface to use with our CRM software. I am > using the manager interface and mysql to store the changes. The only > issue I am having is when a caller joins the queue. > > Currently, I can show the status of phones (ready, not ready, ringing, > ringing ack, in call, etc). What I am wanting to do is to be able to > track the status of the call in the database and do things with it > accordingly. I am able to accomplish this and make it work exactly as > I want, but it requires a modification to the source. For some reason, > the JOIN event in the manager interface doesn't seem to have the > unique call id. Almost every other event does, but JOIN doesn't for > some reason. Can anyone explain why it doesn't? > > My boss asked us to remove our hack to the source and find another way > as it we want to be able to update versions of asterisk and not modify > the source. I thought that I could get around this by using the > NEWEXTEN event that happens just before the join, but I can't tie the > two events together. > I think you boss was right.. that will allow 100% compatibility with all asterisk versions as far the events are there lets see the events for a moment HMMM the call enter go to the ivr so u will get 1 event like this for the entire IVR Event: Newexten Privilege: call,all Channel: SIP/s-f36c Context: open Extension: 0606 Priority: 7 Application: Queue AppData: operator Uniqueid: 1127422073.9183 Server: asterisk1 Ok this is the JOIN event still u can find things there that will allow u to relate the call see the channel parameter. Event: Join Privilege: call,all Channel: SIP/s-f36c CallerID: ... CallerIDName: ... Queue: operator Position: 1 Count: 1 Server: asterisk1 Now the asterisk is preparing himself for call an agent... Event: Newchannel Privilege: call,all Channel: SIP/9915004-e198 State: Down CallerID: CallerIDName: Uniqueid: 1127422096.9194 Server: asterisk1 Event: QueueMemberStatus Privilege: agent,all Queue: operator Location: SIP/9915004 Membership: static Penalty: 0 CallsTaken: 86 LastCall: 1127422074 Status: 0 Paused: 0 Server: asterisk1 Ahh magic again a wait to relate see the channelcalling and i bet what ever u want that when the agent finish u will find the same similarities. Event: AgentCalled Privilege: agent,all AgentCalled: SIP/9915004 ChannelCalling: SIP/s-f36c CallerID: . CallerIDName: ... Context: open Extension: 0606 Priority: 7 Server: asterisk1 so following the events for channel u will probably able to do the same even without the uniqueid. 2 concurrent calls will have diferent channels always u just have to be carefull to ensure u follow the call from the beginning to the end. regards saul > Basically, with the hack modified, here's what I do: > > Call comes in, enter the info into the database with uniqueid as the > key. When a call is answered, I update that record in the database and > so on. Without the uniqueid on the JOIN event I am stuck. > > Any suggestions on a way around this, or a better way of doing it? I > would also be curious if anyone would share their setup if the are > attempting the same. > > Thanks, > > > Josh > ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP DTMF problem
Hi! I have this setup: Analog phone <-> Audiocodes MP-114 <-> Asterisk 1<-> Aastra 480i | \/ Asterisk 2 The codec is alaw on all the calls. Asterisk is CVS-HEAD checked it a couple of hours ago. Asterisk 1 and 2 is connected with a SIP connection using INFO. The Aastra 480i does not support DTMF INFO mode as far as I can tell, so I configures it as rfc2833 in sip realtime. The MP-124 is confugured to use INFO. When I call the voicemail on Asterisk 2 from the analog phone it works fine. When I call from the 480i it does not work. Asterisk 1 and 2 does not "hear" the DTMF. When I call from the analog phone to the 480i or from the 480i to the analog phone Asterisk 1 "hears" the DTMF (* DTMF Received: '1') that is pressed on the analog phone and only some of the tones pressed on the 480i. Some of the DTMF sent from the 480i is registered multiple times on Asterisk 1 e.g. when I press 1 Asterisk register the 1 but thinks it is pressed 2 or 3 times, but I only pressed it once. Is there a problem when Asterisk needs to convert between INFO and rfc2833? Or am I missing something.-- Morten Isaksenhttp://www.misak.dk/blog/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] goiax.com
On Tue, Oct 18, 2005 at 09:14:31PM -0500, Kevin Scott exclaimed: >As a spin off of that, 10 or so numbers you can call anytime, and then 10 >more numbers after that in 24 hours for the random occurrences of 'ordering >pizza'. > > > >But you're right, there are really normally only 10 people I ever try and >call, the problem is, they have 3 phone numbers each. But the 10 standard >and 10 random, I would be content with that. > > > >Kevin I like this approach as well. People could still register multiple accounts, but each account would not be able to make more than 20 different calls per day. I'm also a fan of whitelisting the mailing list users ;-). It probably wouldn't hurt to put a max per-day cap as well (no more than 60 calls in a day). -Ryan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trying to clarify ideas about spands, libtiff & FC4
Carlos Alperin wrote: I spent more than 3 weeks, with some little help of people that belongs to this forum, and after try differents combinations of versions this is my conclusion: Please feel free to send every kind of disappointments opinions. That is going to feel me much better that no answers. (Even if you can show me how stupid I was doing all kind of mistakes) I've got good experience with Mandrake/Mandriva with a 2.6.13.4 kernel Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue Join Event
Tressler, Joshua A wrote: I did a quick Google search of the lists and I hope that I am not asking a question that has already been answered recently. I have been working on a interface to use with our CRM software. I am using the manager interface and mysql to store the changes. The only issue I am having is when a caller joins the queue. Currently, I can show the status of phones (ready, not ready, ringing, ringing ack, in call, etc). What I am wanting to do is to be able to track the status of the call in the database and do things with it accordingly. I am able to accomplish this and make it work exactly as I want, but it requires a modification to the source. For some reason, the JOIN event in the manager interface doesn’t seem to have the unique call id. Almost every other event does, but JOIN doesn’t for some reason. Can anyone explain why it doesn’t? My boss asked us to remove our hack to the source and find another way as it we want to be able to update versions of asterisk and not modify the source. I thought that I could get around this by using the NEWEXTEN event that happens just before the join, but I can’t tie the two events together. I think you boss was right.. that will allow 100% compatibility with all asterisk versions as far the events are there lets see the events for a moment HMMM the call enter go to the ivr so u will get 1 event like this for the entire IVR Event: Newexten Privilege: call,all Channel: SIP/s-f36c Context: open Extension: 0606 Priority: 7 Application: Queue AppData: operator Uniqueid: 1127422073.9183 Server: asterisk1 Ok this is the JOIN event still u can find things there that will allow u to relate the call see the channel parameter. Event: Join Privilege: call,all Channel: SIP/s-f36c CallerID: ... CallerIDName: ... Queue: operator Position: 1 Count: 1 Server: asterisk1 Now the asterisk is preparing himself for call an agent... Event: Newchannel Privilege: call,all Channel: SIP/9915004-e198 State: Down CallerID: CallerIDName: Uniqueid: 1127422096.9194 Server: asterisk1 Event: QueueMemberStatus Privilege: agent,all Queue: operator Location: SIP/9915004 Membership: static Penalty: 0 CallsTaken: 86 LastCall: 1127422074 Status: 0 Paused: 0 Server: asterisk1 Ahh magic again a wait to relate see the channelcalling and i bet what ever u want that when the agent finish u will find the same similarities. Event: AgentCalled Privilege: agent,all AgentCalled: SIP/9915004 ChannelCalling: SIP/s-f36c CallerID: . CallerIDName: ... Context: open Extension: 0606 Priority: 7 Server: asterisk1 so following the events for channel u will probably able to do the same even without the uniqueid. 2 concurrent calls will have diferent channels always u just have to be carefull to ensure u follow the call from the beginning to the end. regards saul Basically, with the hack modified, here’s what I do: Call comes in, enter the info into the database with uniqueid as the key. When a call is answered, I update that record in the database and so on. Without the uniqueid on the JOIN event I am stuck. Any suggestions on a way around this, or a better way of doing it? I would also be curious if anyone would share their setup if the are attempting the same. Thanks, Josh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDMoE and Badness in Kernel
2.6.12 > On Fri, 2005-10-21 at 09:27 -0700, [EMAIL PROTECTED] wrote: >> I received some postings back, as I was trying to do the same thing. >> >> it' is a problem with Kernel 2.6... 2.4 works fine .. this is the >> summary >> I got from reading the posts before. >> >> I hope that helps... I dont have the ability to go DOWn in kernel to >> 2.4.. >> > > the wiki suggested that it was a problem with softirq.c in the kernel > and that this was fixed at some point. What 2.6 version are you running > that you have this problem? > > -- > Trixter http://www.0xdecafbad.com Bret McDanel > UK +44 870 340 4605 Germany +49 801 777 555 3402 > US +1 360 207 0479 or +1 516 687 5200 > FreeWorldDialup: 635378 > ___ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trying to clarify ideas about spands, libtiff & FC4
Haven't had any issues here with slackware 10.x On Sunday 23 October 2005 09:23 am, Carlos Alperin wrote: > I spent more than 3 weeks, with some little help of people that belongs to > this forum, and after try differents combinations of versions this is my > conclusion: > > > > I tried RH9, FC4 & FC4 64 > > I tried with CVS 1.0.2, and Stable 1.0.9 > > I tried with spandsp 0.0.2pre18, 0.0.2pre20 & 0.0.2pre21 > > Libtiff 3.5.7 & libtiff devel 3.5.7 > > Libtiff 3.7.1 & libtiff devel 3.7.3 (I couldn't find 3.7.1) > > > > My conclusion is: > > > > If I need to be able to use fax with Spandsp, app_rxfax.c & app_txfax.c > with libtiff 3.5.7 (and libtiff devel 3.5.7) there is no way to do that on > FC4 (get conflict with GTK2+) > > So it looks like I have to go back to RH9 and at least upgrade to kernel > 2.4.31, and try again. > > > > This is under the presumption that Spandsp, & the rest are going to work. > (Looking at the forum, that is not a 100% fact). > > > > It should be a way to save us a lot of time, if somebody can unify all the > requeriments on each OS, so we can decide before to start which direction > to follow. > > > > The reason for RH9 & FC4 is because they're more familiar. But if someone > can show me a working configuration, I don't hesitate to move the platform. > > > > By the way, the 64 bits platform still looks to be very unstable and not so > fast to implement with Asterisk. > > > > To the digium support: I understand that your recommendation is to go to > 2.6 kernel, but if I need to run spandsp, how to do that without libtiff > 3.5.7. > > > > The general experience is libtiff 3.7.1 locks the asterisk when the machine > boots. > > > > Please feel free to send every kind of disappointments opinions. That is > going to feel me much better that no answers. > > (Even if you can show me how stupid I was doing all kind of mistakes) > > > > Regards, > > > > Carlos Alperin -- -BEGIN GEEK CODE BLOCK- Version: 3.1 GCS/GIT> d-@ s+:+ a? C+++ BLHIS$ U+++ P+> L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+> h r+++ y --END GEEK CODE BLOCK-- pgpwrfAhYS8eQ.pgp Description: PGP signature ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Trying to clarify ideas about spands, libtiff & FC4
I spent more than 3 weeks, with some little help of people that belongs to this forum, and after try differents combinations of versions this is my conclusion: I tried RH9, FC4 & FC4 64 I tried with CVS 1.0.2, and Stable 1.0.9 I tried with spandsp 0.0.2pre18, 0.0.2pre20 & 0.0.2pre21 Libtiff 3.5.7 & libtiff devel 3.5.7 Libtiff 3.7.1 & libtiff devel 3.7.3 (I couldn’t find 3.7.1) My conclusion is: If I need to be able to use fax with Spandsp, app_rxfax.c & app_txfax.c with libtiff 3.5.7 (and libtiff devel 3.5.7) there is no way to do that on FC4 (get conflict with GTK2+) So it looks like I have to go back to RH9 and at least upgrade to kernel 2.4.31, and try again. This is under the presumption that Spandsp, & the rest are going to work. (Looking at the forum, that is not a 100% fact). It should be a way to save us a lot of time, if somebody can unify all the requeriments on each OS, so we can decide before to start which direction to follow. The reason for RH9 & FC4 is because they’re more familiar. But if someone can show me a working configuration, I don’t hesitate to move the platform. By the way, the 64 bits platform still looks to be very unstable and not so fast to implement with Asterisk. To the digium support: I understand that your recommendation is to go to 2.6 kernel, but if I need to run spandsp, how to do that without libtiff 3.5.7. The general experience is libtiff 3.7.1 locks the asterisk when the machine boots. Please feel free to send every kind of disappointments opinions. That is going to feel me much better that no answers. (Even if you can show me how stupid I was doing all kind of mistakes) Regards, Carlos Alperin ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP DTMF problem
Hi! I have this setup: Analog phone <-> Audiocodes MP-114 <-> Asterisk 1<-> Aastra 480i | \/ Asterisk 2 The codec is alaw on all the calls. Asterisk is CVS-HEAD checked it a couple of hours ago. Asterisk 1 and 2 is connected with a SIP connection using INFO. The Aastra 480i does not support DTMF INFO mode as far as I can tell, so I configures it as rfc2833 in sip realtime. The MP-124 is confugured to use INFO. When I call the voicemail on Asterisk 2 from the analog phone it works fine. When I call from the 480i it does not work. Asterisk 1 and 2 does not "hear" the DTMF. When I call from the analog phone to the 480i or from the 480i to the analog phone Asterisk 1 "hears" the DTMF (* DTMF Received: '1') that is pressed on the analog phone and only some of the tones pressed on the 480i. Some of the DTMF sent from the 480i is registered multiple times on Asterisk 1 e.g. when I press 1 Asterisk register the 1 but thinks it is pressed 2 or 3 times, but I only pressed it once. Is there a problem when Asterisk needs to convert between INFO and rfc2833? Or am I missing something.-- Morten Isaksenhttp://www.misak.dk/blog/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Adit 3104 configuration
Has anyone been able to get the 3104 to register more than one line correctly? It seems to work OK for the first line, but as soon as I turn on more than one it appears that only the last one is actually registering corectly. The 3104 sometimes indicates the line is registered, but * says not. This looks like a very useful unit and would really like to get it to work. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem with asterisk
Venerable Sir, when i start the asterisk server a error message show that is chan_oss.c:287 sound thread read error on sound device resource temperorily unavailable and hmmming sounds comes sir what is the problem does my sound card inbuilt in motherboard does not support ? In malaysia digium retailers is available thank you for support neon _ Welcome Goddess Durga. Join the Navratri celebrations online. Be a part of the festivities http://server1.msn.co.in/SP05/Navratri/index.asp ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Testing AreskiCC
Not an answer to your questions, but just in case you don't know there is a lot of info on the wiki: http://www.voip-info.org/wiki/view/AreskiCC+CallingCard+Application We use Areskicc here, and it works great. However we do not use sip/iax friends, perhaps both of your problems lie there? Best of luck, G Rikunj wrote: Hello gurus, After successful installation of Areski. I am having few problem before I can do any test dial-outs. When I try to create sip/iax friend from web interface it says "Could not open buddy file '/etc/asterisk/additional_areskicc_sip.conf' I tried creating the file manually without luck. Second I am unable to dial any phone nos after card verification. As soon as the card is verified with the remaining balance it straight forward tells invalid-digit, without a prompt and hangs up. What could be missing? Please help. Regards, Rikunj ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Satellite receiver over IP
Jay Milk wrote: If that's dishnetwork and they keep charging you their $5 programming access fee or whatever they call it, just plug it in and confirm that you get a dial-tone. Then call tech-support and have them adjust billing -- all they check is that the receiver gets a dial-tone and they take your word for it. No, it's because they are auditing me because they have not called home. -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to play a voice file for " decline "
Asterisk guy wrote: When get sip respond 6xx ( such as 603 decline), I want asterisk to play a voice file to the caller, how to do this in extensions ? for example, when get 603 respond, play decline.gsm to caller when get 604 respond, play doesnot-exit.gsm to caller when get 606 respond , play not-acceptalbe.gsm to caller You can't. You can check the value of DIALSTATUS, but that does not directly map to SIP responses. chan_sip MAY set HANGUPCAUSE with more information, but it's still not a 1-to-1 mapping with SIP responses. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to play a voice file for " decline "
When get sip respond 6xx ( such as 603 decline), I want asterisk to play a voice file to the caller, how to do this in extensions ? for example, when get 603 respond, play decline.gsm to caller when get 604 respond, play doesnot-exit.gsm to caller when get 606 respond , play not-acceptalbe.gsm to caller ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with asterisk 1.0.9 and sip and dtmf
Check if you have Silence Suppression disabled on PSTN line of spa-3000 (admin/advansed/PSTN line). On Sat, 2005-10-22 at 18:05 -0400, Mike Bernson wrote: I have asterisk running with sipura 3000 connect to PSTN and sipura 2000 connected to phones. On inbound calls I am getting what sounds like DTMF tone when someone is talking on the PSTN side of the phone. It sound like someone is hitting key on the phone while talking. Is there any way to stop this from happing. Here is the PSTN and one ext from the sip.conf PSTN line [199] username= type=friend secret= record_out=Adhoc record_in=Adhoc qualify=yes port=5061 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=incomming canreinvite=no ext [206] username= type=friend secret= record_out=Adhoc record_in=Adhoc qualify=yes port=5061 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-sip canreinvite=no ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Admission Control in Asterisk
Hi Trixter, Yes i did try to make setgroup for the outbound but the problem is after you move it to the desired context or extension in the gotoif statement the group that you have set it in is back to zero so I really can't use it for the outbound, the group used for the outbound will not give the correct count of users dialling out. As u said I am using CVS-Head and used the group_count() with gotoif statements so I am clear of the checkgroup() bug. Thx MAG trixter aka Bret McDanel wrote: On Sun, 2005-10-23 at 11:42 +0200, Mohamed A. Gombolaty wrote: > Dear All, > > I was trying to limit the number of calls between different located sites in > order to avoid congestion of the bandwidth, but as I found from the mails and > testing that it is easy to do it for the incoming calls by the setgroup() and > group_count while it is the outgoing is hard to track or limit, So I was > wondering if we will see a Call Admission Control soon in Asterisk that can do > this job or not? setgroup() should work for outbound. Did you try it and have problems? In asterisk 1.0.x there is a bug about transfered calls, is that where you were running into problems? I find this unlikely since you referenced group_count, which is a 1.2 function (replacing the deprecated checkgroup()). http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 Name: signature.asc signature.asc Type: application/pgp-signature Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTBILL
On Sun, 2005-10-23 at 15:57 +0600, Kanishka Somaratne wrote: > hi > can we install astbill under mysql 4, or is mysql 5 a must it uses stored procedures which arent available under 4. if you can work around that you might be able to use 4.x -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTBILL
hi can we install astbill under mysql 4, or is mysql 5 a must regards kanishka ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Admission Control in Asterisk
On Sun, 2005-10-23 at 11:42 +0200, Mohamed A. Gombolaty wrote: > Dear All, > > I was trying to limit the number of calls between different located sites in > order to avoid congestion of the bandwidth, but as I found from the mails and > testing that it is easy to do it for the incoming calls by the setgroup() and > group_count while it is the outgoing is hard to track or limit, So I was > wondering if we will see a Call Admission Control soon in Asterisk that can do > this job or not? setgroup() should work for outbound. Did you try it and have problems? In asterisk 1.0.x there is a bug about transfered calls, is that where you were running into problems? I find this unlikely since you referenced group_count, which is a 1.2 function (replacing the deprecated checkgroup()). http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk dropping call file without *any* notice
I'm trying to debug the old call file redial bug I prepared a call file and trying to setup a call from my remote asterisk server to my home number. However whenever I dump a call file to /var/spool/asterisk/outgoing it is just deleted without *any* action Nothing in the logs, nothing on the console... GRRR I already increased the verbosity and enabled sip debug... nothing It would be nice if * would at least report that it found a call file and ditched it for whatever reason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Admission Control in Asterisk
Dear All, I was trying to limit the number of calls between different located sites in order to avoid congestion of the bandwidth, but as I found from the mails and testing that it is easy to do it for the incoming calls by the setgroup() and group_count while it is the outgoing is hard to track or limit, So I was wondering if we will see a Call Admission Control soon in Asterisk that can do this job or not? Thx MAG ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone using Java SIP communicator with Asterisk ?
Hi, this java video softphone claims it can operate with Windows messenger. It's also mentioned on this web page http://www.voip-info.org/wiki/view/SIP+COMMUNICATOR But I couldn't find any more info on how to set it up with Asterisk and how compatible is with other video softphones... Anyone with such experience or working installation ? Thanks in advance, regards, Rob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
