Re: [asterisk-users] howto debug bad iax voice quality?
John Beaman Telecom Specialist II Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 4/4/2008 3:23:49 PM I'm set up to call 3 digit extensions at the office ( running 1.4.13) from home ( 1.6.0 beta7) over iax. 1 out of 3 times the call breaks up, but only in the home - office direction. office - home always sounds good. If it were a poor internet connection, I'd expect both sides of the conversation to be poor. Not surprisingly, each side can ping the other in the same time - 25-30ms. Both servers have the iax jitterbuffer on. I could always use a lower bit-rate codec ( now using mu-law ), but I don't see how it could be a one way bit-rate issue. Any suggestions appreciated. sean ___ Sean, Broadband connections are almost always asynchronous, which means your download speed is considerably higher than upload speed. With some or our remote workers they were getting 1.5 Mbps download but only 125 Kbps upload speed! We ended up having to upgrade their connection to a business class connection, but upload speed was still only ½ of the download speed. You can check your speed both directions with a speed test from a site such as: http://www.speakeasy.net/speedtest/ HTH ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Fax and anti-spam
John Beaman Telecom Specialist II Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 12/11/2007 11:23:29 AM Hi, One of Asterisk features is fax2mail. As a good share of incoming faxes can be considered as advertising spam, does it make sense to use email anti-spam features to filter them ? I can't foresee any practical way to do so but I would be very curious to discuss about it. Regards I do not believe this is possible. Email spam filters use rules to filter on the text of the email message, while a fax is scanned, transmitted and received as a graphic image. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need T1 crossover cable?
For pinout info, check out: http://www.asteriskdocs.org/cables/ John Beaman Telecom Specialist II Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 10/26/2007 4:01:29 PM Michelle Dupuis wrote: I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My Sangoma A102D shipped with 2 T1 cables - which I assume are straight through. Do I need to make crossover cables for this scenario? Thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yes, use a T1 crossover(not an ethernet crossover). Lyle - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ABE 1.4 release date
Anyone know when Asterisk Business Edition 1.4 will be released? We are looking to purchase, but with all the changes between 1.2 and 1.4 think it may be best to wait if the new version is just around the corner. Thanks, John Beaman Telecom Specialist II Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
Lynn, If I understand you question correctly, you would need: A computer (preferably a server) to run Asterisk An analog interface card such as the Digium TDM400P An analog phone line (POTS) An analog (real) phone Calls would come in on the POTS line, get answered by Asterisk. Callers would hear your voice menu, and input their choice. If they opted for a live person, asterisk would then send the call to your analog (real) phone. John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 8/1/2007 8:48:47 AM Hello, I am a small business owner in need for a solution that automatically answers an incoming call, prompts the caller via touch-tone menu (press 1 to leave a message, press 0 to speak to a representative) and will ring my (real) phone ONLY if requested by caller. I know that Asterisk is capable of all the logic behind what I described above. However, I couldn't find a hardware product that will allow me to accomplish the above (preferrable using Asterisk software). Does such thing exists? Thanks, Lynn Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. http://tv.yahoo.com/collections/222 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware that can ring my phone?
Lynn, I am unfamiliar with soho-pbx, so I cannot comment on quality, service, configurability, etc. They are based out of Hong Kong, and their box is probably already running some flavor of Asterisk, so you would need nothing additional except for the phone line coming in and the telephone. I got quite a kick out of their description for the SP-104 box as referenced by your link: The photos below are model SP-104, a model that costs only tens of US dollars Not sure how much that comes to, but sounds pretty cheap... John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 8/1/2007 10:29:51 AM Yes, you understood correctly. Thank you - and all others who replied so quickly - for your precise and guiding answers. The Digium TDM11B looks looks like the perfect match for me: http://www.telephonyware.com/telephonyware/tw00068.html But one thing that I forgot to mention is that my business is only in its beginning stage and I need to be as thrifty as possible. While $216 is a reasonable price, I was wondering whether my (currently very modest) goal can be achieved by spending much less (under $100). For example, what if I buy one of those el-cheapo PBX boxes and connect it to an Asterisk server? http://www.soho-pbx.com/sp-104.htm Do you think this could work for me or did I expose a gross misconception on my part? Thanks, Lynn --- john beaman [EMAIL PROTECTED] wrote: Lynn, If I understand you question correctly, you would need: A computer (preferably a server) to run Asterisk An analog interface card such as the Digium TDM400P An analog phone line (POTS) An analog (real) phone Calls would come in on the POTS line, get answered by Asterisk. Callers would hear your voice menu, and input their choice. If they opted for a live person, asterisk would then send the call to your analog (real) phone. John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 8/1/2007 8:48:47 AM Hello, I am a small business owner in need for a solution that automatically answers an incoming call, prompts the caller via touch-tone menu (press 1 to leave a message, press 0 to speak to a representative) and will ring my (real) phone ONLY if requested by caller. I know that Asterisk is capable of all the logic behind what I described above. However, I couldn't find a hardware product that will allow me to accomplish the above (preferrable using Asterisk software). Does such thing exists? Thanks, Lynn Sick sense of humor? Visit Yahoo! TV's Comedy with an Edge to see what's on, when. http://tv.yahoo.com/collections/222 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Need a vacation? Get great deals to amazing places on Yahoo! Travel. http://travel.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission
Re: [asterisk-users] Royalty for On Hold Music ?
Just Google for: royalty free music, and will find plenty of sites that will serve your needs. John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 7/31/2007 12:49:45 PM So is there a simple way to license decent, up to date music? Can I just go to a website, click a buy button, pay my money and download the song? It seems idiotic that you need 15 lawyers and a million bucks use decent on hold music. Maybe I just don't know the procedure. I am all for paying the license fees and doing it right but they sure don't make it easy to give them money. Any help would be appreciated. On 7/31/07, Deepak Naidu [EMAIL PROTECTED] wrote: Thanks Jared, Yes I am using with Asterisk only. So I am using the inbuilt music from Asterisk for onhold. -- Deepak Jared Smith [EMAIL PROTECTED] wrote: On Tue, 2007-07-31 at 06:36 +0100, Deepak Naidu wrote: I think we need to pay for the later, but I am not sure if we need to pay for the inbuilt asterisk(freepbx) on hold music. I'm no lawyer, but here's what I understand. (Please consult with an attorney in your area, and don't consider this legal advice.) The hold music that comes with Asterisk is provided by Digium under license from Freeplay Music Corporation for use in conjunction with the Asterisk software only. It's my understanding that you don't have to pay any kind of royalties to use it, as long as you're using it with Asterisk. You *do* have to pay royalties on music (or MP3 files) by commercial artists. These royalties vary by country. Using commercial music as hold music is considered broadcasting the music, which requires different licensing arrangements with the copyright holder. In the United States, you can buy a license from ASCAP (the American Society of Composers, Authors, and Publishers) to be able to broadcast music from the major record labels. There are also several other places you can get royalty-free music for hold music. I've had good luck looking online, especially at sites like MagnaTune. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your free account today. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Royalty for On Hold Music ?
No, you will not. According to the music industry those artists are all are entitled to compensation for every time their song is broadcast, which includes MoH. AFAIK, there are no popular songs by popular artists that are royalty-free. John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 7/31/2007 1:37:00 PM I have done this in the past and I don't recall ever finding any popular music by popular artist. For example, if I wanted to play oh I don't know an original song performed by the original artist such as Nora Jones or The Beatles will I find this sort of thing at a Royalty Free Site? On 7/31/07, john beaman [EMAIL PROTECTED] wrote: Just Google for: royalty free music, and will find plenty of sites that will serve your needs. John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 7/31/2007 12:49:45 PM So is there a simple way to license decent, up to date music? Can I just go to a website, click a buy button, pay my money and download the song? It seems idiotic that you need 15 lawyers and a million bucks use decent on hold music. Maybe I just don't know the procedure. I am all for paying the license fees and doing it right but they sure don't make it easy to give them money. Any help would be appreciated. On 7/31/07, Deepak Naidu [EMAIL PROTECTED] wrote: Thanks Jared, Yes I am using with Asterisk only. So I am using the inbuilt music from Asterisk for onhold. -- Deepak Jared Smith [EMAIL PROTECTED] wrote: On Tue, 2007-07-31 at 06:36 +0100, Deepak Naidu wrote: I think we need to pay for the later, but I am not sure if we need to pay for the inbuilt asterisk(freepbx) on hold music. I'm no lawyer, but here's what I understand. (Please consult with an attorney in your area, and don't consider this legal advice.) The hold music that comes with Asterisk is provided by Digium under license from Freeplay Music Corporation for use in conjunction with the Asterisk software only. It's my understanding that you don't have to pay any kind of royalties to use it, as long as you're using it with Asterisk. You *do* have to pay royalties on music (or MP3 files) by commercial artists. These royalties vary by country. Using commercial music as hold music is considered broadcasting the music, which requires different licensing arrangements with the copyright holder. In the United States, you can buy a license from ASCAP (the American Society of Composers, Authors, and Publishers) to be able to broadcast music from the major record labels. There are also several other places you can get royalty-free music for hold music. I've had good luck looking online, especially at sites like MagnaTune. -- Jared Smith Community Relations Manager Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your free account today. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com
Re: [asterisk-users] Newbie Advice on Asterisk and Linux
Mark, Welcome to the club. Learning Linux can be a daunting task. After working with it for the last decade, I am still learning. My best recommendation is to play with it on a test box, and post questions to a related community forum if you get stuck on something. If you are looking for something more intense and less time-consuming, check your local colleges. The colleges in my area offer several classes on Linux as part of a degree in Network Administration. HTH, John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 7/26/2007 3:08:36 PM HI All, I'm new to Asterisk and also to Linux. I have a large IVR project that I'm about to embark on. I'm new to programming; new to Linux and new to Asterisk. I think I'm about to climb a steep learning curve. I have an existing IVR which is getting on for nine years old and is no longer supported by my vendor. I intend to replicate the system almost as is and then add additional features and functions. I have been looking for a developer to put together my project and while doing so have done lots of research and spoken to many people. The people who seem to understand my needs have recommended Asterisk. For the last couple of days I've been trying to look into Asterisk and learn as much as I can; this has got me excited, motivated and a little confused. Asterisk sounds like a great project and a great community. I think I have as much of an overview as I can. Now I need to set up a Linux system and get Asterisk running on it. I've started to read the book Asterisk: The Future Of Telephony and would like to now setup up a hobby computer to do some hands on learning. The book covers Red Hat Linux so I thought I'd look for a 'Red Had for Dummies' book. Even that got confusing. There's Linux Fedora, Enterprise Linux 4 and others. Can someone suggest a starting point on learning Linux? Thanks in advance, Mark No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.476 / Virus Database: 269.10.20/919 - Release Date: 26/07/2007 9:56 AM - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 12
I too was curious about this, so I copied the text into Babel Fish, and this is the result: I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my return. For any urgency, to contact Emmanuelle Parache Moga or Cédric Buzay. If this guy is really going to be out until November these messages will get rather tiresome... John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 4/3/2007 2:29:11 PM Ok, I'll bite. This is the 4th message like this I've gotten today. I don't speak French but it looks like an autoresponder. If so, why is it replying back to the list, why not on every message sent, and why is it incrementing the issue number? Or am I missing something? Jay [EMAIL PROTECTED] wrote: Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 12
Ah, yes. One of the many differences between the US and the rest of the world. [EMAIL PROTECTED] 4/3/2007 2:52:16 PM john beaman wrote: I too was curious about this, so I copied the text into Babel Fish, and this is the result: I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my return. For any urgency, to contact Emmanuelle Parache Moga or Cédric Buzay. If this guy is really going to be out until November these messages will get rather tiresome... This is from April 2nd to April 11th. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No sound with Playback() or Background()
Jake,
Check to make sure you have the sound files for whatever audio format
(gsm.wav, etc) that you are using. I don't remember the details, but Asterisk
quit including the sound files in the base distribution to minimize the size of
the download. Then, in a later version, they have a script that will prompt
you for some info, then will download and install the sound files that you want
to use.
John Beaman
Telecom Specialist
Voice Telecommunications Services Department.
Good Samaritan National Campus
605-362-3331
[EMAIL PROTECTED] 2/28/2007 7:30:03 AM
Hi Jake,
Perhaps you can add
NoOp(${PLAYBACKSTATUS}) after each Playback, it should return either
FAILED or SUCCESS in the CLI
Hope that helps.
Best Regards,
Joanna
On 2/28/07, Kuba [EMAIL PROTECTED] wrote:
After switching to Asterisk 1.2.14 from 1.0.x, I encountered a very
strange problem. There is no sound with Playback() or Background()
commands.
Even though, Asterisk console shows the file is being played when I call
the extension ( i.e. echo test), I can't hear anything.
My echo test extension looks like this:
exten = 600,1,Answer
exten = 600,2,Playback(demo-echotest)
exten = 600,3,Echo
exten = 600,4,Playback(demo-echodone)
exten = 600,5,Hangup
Console shows something like that when I call:
-- Executing Answer(SIP/206-081a7160, ) in new stack
-- Executing Playback(SIP/206-081a7160, demo-echotest) in new stack
-- Playing 'demo-echotest' (language 'en')
So it looks like Asterisk is playing the file, but I can't hear anything.
The files demo-echotest.gsm and demo-echodone.gsm are present in
/var/lib/asterisk/sounds, so this is not the matter of missing files.
The same problem occurs with every file I try to play with Playback() or
Background() commands.
Any ideas ?
Thanks
Jake
--
--- Domeny w ULTRA NISKICH cenach: ---
.pl - 29 zl, .com.pl - 22,50 zl, reg - 7,50 zl
http://www.domeny.alpha.pl
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Re: [asterisk-users] Asterisk dual contexts stupidity
Yes, there are duplicate 3 lines, which can cause havoc. For this reason it is
recommend you use 'n' in your contexts. Such as:
[main-night-aa]
exten = s,1,Answer
exten = s,n,Background(/etc/asterisk/night)
exten = s,n,Directory(foobarcorp,internal,l)
'n' allows for the addition and deletions of lines in the contexts without
getting the numbering all messed up.
[EMAIL PROTECTED] 1/30/2007 10:06:09 AM
You seem to have to many s,3's
[main-night-aa]
exten = s,1,Answer
exten = s,2,Background(/etc/asterisk/night)
exten = s,3,Directory(foobarcorp,internal,l)
exten = s,3,Wait(3)
exten = s,4,Voicemail([EMAIL PROTECTED])
exten = s,5,Hangup
exten = 1,1,Directory(foobarcorp,internal,l)
exten = 00,1,VoicemailMain([EMAIL PROTECTED])
include = retardonightfix
Bails
J. Oquendo wrote:
So I have my extensions.conf
(http://www.infiltrated.net/exten.stupidity.conf) shortened
in case someone wants to look. Has someone encountered the following?
I've racked my
brain on this for too long...
I have two contexts, day and night...
Caller (Daytime) -- Dials an extension -- Caller hears extension ring
on receiver -- Call goes through
Caller (Night) -- Dials an extension -- Caller hears silence until vm
picks up -- Leaves a voicemail...
Significant albeit insanely stupid Asstricks message:
2007-01-30 09:22:57 DEBUG[9946]: pbx.c:2300 __ast_pbx_run: Oooh, got
something to jump out with ('2')!
(Oooh how about creating errors we can figure out Digium!)
Any thoughts
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Re: [asterisk-users] Simple question
The first include references another context within extensions.conf. Contexts are defined by words in brackets. In your example, there would be a context in extensions.conf that would look like: [inbound] Contexts allow for setting up difference services and difference user capabilities all within the extensions.conf file. The second include is including the contents of multiple *.conf files located in a directory called inbound. JB [EMAIL PROTECTED] 1/27/2007 6:50 AM Whats the difference between the following statements in extensions.conf include=inbound AND #include inbound/*.conf -- Regards Rizwan Hisham Software Engineer - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] php agi - first phrase truncated, all others fine
Greetings,
I have never done any agi programming, but my first thought is maybe you need
a wait statement after answering?
John Beaman
Telecom Specialist
Voice Telecommunications Services Department.
Good Samaritan National Campus
605-362-3331
[EMAIL PROTECTED] 1/15/2007 10:53:51 AM
I have the following code. When I call the extension, it either ignores the
first Hello there everyone, or says hello and moves on sometime stoping
before it finishes hello. The rest of the text reads fine. Anyone else
have this issue??
Thanks!
require('/var/lib/asterisk/agi-bin/phpagi.php');
$agi = new AGI();
$agi-answer();
$agi-swift(Hello there everyone );
$agi-swift(Please press 1 for a search .);
$result= $agi-get_data('beep',3, 1);
$zip= $result['result'];
$agi-swift(That concludes your call. Thank you, Good bye .);
$agi-hangup();
-
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please immediately notify the sender by telephone or return
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The Evangelical Lutheran Good Samaritan Society.
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Re: [asterisk-users] Asterisk 1.4.0 B4 Sounds Directory
You're right. I just untarred asterisk-1.4.0-beta4.tar.gz. The sounds folder is there, but it is empty except for Makefile and sounds.xml. I am not expert, but when I looked at the Makefile, it appears that it prompts the user to pick a format for the sounds files (ulaw, wav, etc), and then it downloads the appropriate sound files. John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 12/16/2006 2:04 PM Hi, I noticed that the sound directory is missing from asterisk-1.4.0-beta4.tar.gz. This directory (7 M) witch existed in asterisk-1.4.0-beta3.tar.gz has GSM Core Sounds and some MOH. Does anyone know why it has been removed from the latest beta? Regards. Sponsored Link Mortgage rates near historic lows: $150,000 loan as low as $579/mo. Intro-*Terms https://www2.nextag.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] StripXXX apps missing from asterisk-1.2.13?
StripLSD is obsolete: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+StripLSD StripMSD is being phased out: http://bugs.digium.com/view.php?id=5673 John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 12/14/2006 3:55:59 PM All of StripMSD, StripLSD, etc., are missing when I downloaded asterisk-1.2-current.tar.gz, which explodes into 1.2.13. Are the strip club deprecated? What replacement functions should I use? Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
