[asterisk-users] Any timeframe for the release of the Asterisk 11-Lumenvox connector bridge?
I'm starting to think about migrating from an old Asterisk box to a new one and want to use the Asterisk 11 long term support release, but need Lumenvox integration and I don't see the Asterisk 11 connector bridge for Lumenvox available yet. Lumenvox tech support says this is under Digiums control. Can anyone give an idea of how soon it'll be available? Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Corba interface
On 6/15/2010 9:57 AM, Muro, Sam wrote: Does anyone know how to configure asterisk to be able to query Corba interface directly from the dialplan I do it by implementing my dialplan as a Java server using the asterisk Java FastAGI interface, then from Java I can make all the CORBA calls I want (I personally use JacORB). Check http://www.lumenvox.com/partners/digium/applicationzone/projects/javaPizza.aspx for some sample code on the Java part - this happens to drive Lumenvox to handle a call, but you can easily hack it to stick CORBA calls in. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Here is Step by Step Example of Asterisk PBX System Install and configuration
Tilghman Lesher wrote: Emacs is a nice operating system, but it lacks a decent editor. :-P It's also a nice religion. Steve ___ -- 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] Open door automatically...
Carlos Chavez wrote: I have a new setup that uses a 2N Entrycom door phone that has a switch to open an electric lock. The way this works is that when you are speaking with someone at the door you dial a code and it releases the lock on the door. This part works great. My customer wants to be able to dial a certain number and have the door open automatically without having to wait on the phone. I can simulate this option by using the D option of the Dial command to send DTMF to the door phone once it answers. The only problem is that they do not want to wait until the door phone answers. They just want to dial a number and hangup immediately. How can I do this? Maybe something like the number the user dials runs an AGI script which creates a call file which dials the real device. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Panasonic Door phone monitor to Asterisk box?
C F wrote: the Panasonic model you mention will only work with a panasonic PBX. and no it does NOT plug into a station port on the Panasonic but onto the door card of any of the Panasonic systems. Having installed that Panasonic video doorboxes in the past I would suggest stay away from it. it is not as nice as the brouchures make it look, and extremly overpriced. Glad I asked. Here's what I REALLY want to do... I like the look of the Panasonic door unit - looks like a simple doorbell, but has a camera, mic, speaker in there as well as the button. I want the camera (NTSC) hooked to a video capture card full time so that system can always see outside. I'm happy with a plain old doorbell sounding when the button is pressed. I don't care about the inside wall mounted panel at all, and I'd be happy to have the mic/speaker connected to my asterisk box somehow. This means: - the doorbell works with a chime - my home automation/security system can always view out the camera - I can call the outside unit from my Asterisk PBX and talk to whoever is out there. It all starts with the outside hardware, do you (or anyone else) know of a nice looking unit preferably along the lines of the look of the Panasonic unit? Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Panasonic Door phone monitor to Asterisk box?
I'm considering getting a Panasonic video door phone system (VL-GM301A) which can interface with a PBX and would like to connect it to my Asterisk box with an analog FXS port. Of course the Panasonic documentation only talks about hooking it up to a Panasonic PBX which only talks about using Panasonic phones, so it's hard to tell whether the 2 wire connection from the door phone monitor is analog or digital. The video door phone itself hooks to the door phone monitor with a 2 conductor wire, so that part is clearly digital since video, audio, and button press all go through that wire, but since the door phone central station apparently plugs into a Panasonic PBX FXS port which possibly supports fax machines I'm thinking that this might be an ordinary analog connection. Does anyone know if the door phone interface is analog or digital? Anyone have experience with interfacing with one? Thanks Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Magnetic door locks
Paul Hales wrote: I remember driving past a building with a magnetic door lock (where a friend worked) late one night, and he noticed that their door was open and swinging in the breeze. Turns out that if you lost power, this particular door would open up as well. PaulH Heck, that was the whole premise behind the bad guys opening the vault in the first Die Hard movie - get the feds to cut the power... Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Poor audio quality with TDM400 card
Try adding the following to your voicemail.conf context: format=wav49|wav Steve Leotis buchanan wrote: Hey Guys, I have configured my first asterisk box. it works ok so apart, but the playback sound quality is terrible, its low and the output sounds distorted and its seems to have been clipped. Can anyone help. On Sun, Jul 13, 2008 at 11:00 AM, Chris Rowson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, this is my first post to the list, but I have tried to search elsewhere for a solution SNIP I'm using sipgate.co.uk http://sipgate.co.uk for incoming calls, but when I make a test call from the PSTN, the call just dies without connecting to my Astlinux box. (I'm monitoring asterisk console via 'asterisk -rv' and see nothing). SNIP Thanks for the suggestions. I ran tcpdump and it indicated that traffic on that port was being forwarded to the asterisk server. It looks like I basically wrote a load of nonsense in the extensions.conf file. I edited the file to input the extension the incoming call should be coming from and it now works. Working file --- [from-pots] exten = 277,1,Answer() exten = 277,n,Wait(3) exten = 277,n,Playback(tt-weasels) exten = 277,n,Hangup() So in summary it was basically me misconfiguring the box... Cheers Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Leotis Buchanan Manager/Electronic Design Systems Engineer Exterbox.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Database Handling
Tilghman Lesher wrote: Correct; it's actually a workaround for a bug in the MySQL drivers. It was discovered long after 1.2 was end-of-lifed. I got bit by MySQL reconnects on some other software I wrote I think when I jumped from MySQL 4.* to 5.*. If memory serves, here is the relevant info from the official MySQL documentation: From: http://dev.mysql.com/doc/refman/5.0/en/upgrading-from-4-1.html The reconnect flag in the MYSQL structure is set to 0 by mysql_real_connect(). Only those client programs which did not explicitly set this flag to 0 or 1 after mysql_real_connect() experience a change. Having automatic reconnection enabled by default was considered too dangerous (due to the fact that table locks, temporary tables, user variables, and session variables are lost after reconnection). This may explain why this is happening in Asterisk. In the case of my other code the answer was not to keep a long term connection through idle periods. Steve ___ -- 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] Which Cepstral Voice to license
The Allison voice is nice and matches with the built in recordings fairly well. Steve ___ -- 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] New generic sounds
Quoting Philipp Kempgen [EMAIL PROTECTED]: Thank you for calling 911. All of our representatives are currently busy. Your estimated hold time is 2 hours and 15 minutes. Thank you for your patience. ... MOH More like: Thank you for calling 911. All of our representatives are currently busy. To hear tips for for resuscitating yourself on the back of a chair press 1 or visit our website at www.911selfhelp.com. Steve This message was sent using IMP, the Internet Messaging Program. ___ -- 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] DUDE!!!!! was RE: Dialplan Visualization(Extensions.conf or Dialplan Show)
Quoting Steve Totaro [EMAIL PROTECTED]: You may have a point although I was more doing a favor or looking out rather than trying to push my wares on someone where they did not fit. Thanks, Steve Totaro I for one found your post about the cheap machines very useful and would appreciate them in the future. Because I'm generally interested in Asterisk itself and not the business I don't subscribe to the biz list. Steve T, I think the only reason you were called on this was because the other posters who were violating the spirit of the rules were trying to obfuscate their own issues and you should ignore this. Steve Prior This message was sent using IMP, the Internet Messaging Program. ___ -- 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] FYI about my Mona Vie business venture
BerkHolz, Steven wrote: Here is a list of what it is supposed to do: ... Improves mental clarity/focus Just wait until that benefit really kicks in and you realize you've joined a pyramid scheme... Steve ___ -- 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] Mail Server
Mike Hammett wrote: I need to setup a small mail server on a local network. It only needs SMTP ability as it's just so Asterisk can send out emails. The machine has sendmail installed. My primary mail server seems to be rejecting the messages. Some research says something isn't configured properly. What do I have to do so the outside world accepts emails from my Asterisk box? It is behind a NAT. On the mail server machine add the IP address or name of the asterisk box to /etc/mail/relay-domains and restart sendmail. Steve ___ -- 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] Best alternative for getting prompts recorded.
[EMAIL PROTECTED] wrote: Thanks everyone for the reply. Till now we had simple IVR so we recorded it ourself. Now I have a requirement where customer needs a customized message to be played to customer. I am basically looking for some Text to Speech software that would be cost effective (most probably a open source) and would convert Text to Speech. I tried Fetival, but the quality of the sound is not good. Can we improve the sound quality of Festival somehow. Cepstrel is good, cheap, and now has an Allison (yes, THAT Allison) voice! Steve ___ -- 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] TDM400P phone won't ring
Shane Wegner wrote: Hello all, I have two handsets connected to FXS ports on a TDM400P, both GE models but one rings and the other does not. The phone models are not identical. The phone which doesn't ring on the TDM does ring when connected to a regular POTS line and I tried connecting another phone to the port and it rings fine. Do you have the power connector on the TDM400P card hooked up? Steve ___ -- 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] Replacement for Allison
Matthew Rubenstein wrote: Is anyone else interested in creating new voices for Festival (the voice synth bundled with Asterisk) that might not be as good as Allison's recordings, but are better than the current Festival voices? If you try to do live voice synth for prompts you'll probably run into something I encountered - the background and speechbackground (I also use Lumenvox in addition to Cepstrel) only take a file. That means unless I'm missing something you can't have a TTS prompt that can be interrupted like a recorded file. The workaround of TTS to a file and then play the file sounds to me like it would introduce delays and besides it's ugly. That's why I posted a suggestion to the recently created asteriskideas.org that background and speechbackground be enhanced to take an app in addition to a simple file. If you agree, then please vote for it. Steve ___ -- 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] AsteriskIdeas.org :: Comment on submitted ideas
Johansson Olle E wrote: I can't say that ideas are pouring in to AsteriskIdeas.org, but we still have a few ideas worth a discussion. I entered one and submitted it, but then it seems it was caught in approval mode and never showed up by the time I gave up looking at the site. Now that you mention it I see it's there. Steve ___ -- 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] Interrupt the swift text
Naveen Palani wrote: Does anyone have any ideas i can work it out. How can i have the Asterisk cmd Background inside macro? or how to execute the GoTo command? I have really started to wish for 2 new standard commands - BackgroundApp and SpeechBackgroundApp to be added to Asterisk just for this sort of situation. Steve ___ -- 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] Mark Spencer and guest(s) LIVE today at 12 Noon EST - 11 Central - 17:00 UTC
randulo wrote: On Jan 6, 2008 12:45 AM, Steve Prior [EMAIL PROTECTED] wrote: Can you make a podcast out of these recordings by creating an XML file indexing them? That way it would be much easier for us to pick them up as they come out. http://feeds.feedburner.com/AstUser Got it, thanks! Steve ___ --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] Mark Spencer and guest(s) LIVE today at 12 Noon EST - 11 Central - 17:00 UTC
randulo wrote: After hours with Allison Smith, Voice of Asterisk Some of you may want to listen to the after hours of the conference where Allison called in. The first thing she said was I am not muted! Player at http://voipUsersConference.org Full Mp3 downloads list: http://food4wine.ning.com/conference Can you make a podcast out of these recordings by creating an XML file indexing them? That way it would be much easier for us to pick them up as they come out. Steve ___ --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] How are you using Asterisk at Home ?
GNUbie wrote: By the way, my Asterisk PBX server is also my wireless access point, web server, file server, music server, VPN server, database server, firewall and router. Repeat after me - NEVER NEVER NEVER run other servers on your router/firewall machine!!! That machine needs to be a maximum security low vulnerability box and running all sorts of stuff on it conflicts with that. Your web server is probably your weakest link in security, so I wouldn't put your file server, music server, or database server on that same box because if someone hacks through some webapp you've installed (it's happened to me with both the TWiki and awstats packages) then if they've got root on your web server box you don't want them messing with the other stuff. I know it sounds like overkill, but I see three boxes here: 1 - firewall/router 2 - web server and other public facing services (sendmail for example) 3 - internal facing services - database, asterisk, file/music server Some day when box #2 gets rooted (and it will eventually) you'll thank me... Steve ___ --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] Cepstral's Allison is having troublespeaking clearly
Kai-Uwe Jensen wrote: How are you playing the voice? Do you use something like app_swift or app_cepstral? Just fixed app_swift for my own installation by changing the framesize constant definition from 160*4 to 20, after googling for a similar issue. Works like a charm now. It only broke recently, i.e. not with the first 1.4.x releases, but maybe only a couple of months ago. Can you specify exactly where you made this change? I'm looking at the source for app_swift-0.9 right now and don't see a framesize constant. I'm getting some breakup when using app_swift over an IAX connection and thought I'd try this. Thanks Steve ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] phone as control interface (was 99 bottles of beer)
Steve Edwards wrote: Personally, I hate voice recognition systems. Voice prompts are great, but don't take away my keypad. I never proposed to take away your keypad, I just wanted to add the voice option as well. What I do want to get rid of is the step below where you press ** to get into remote mode. What I think would be smoother is to have the extensions organized in such a way that the first button press (or voice utterance) is enough to determine whether the session is a phone call or an automation request. Then as soon as the phone is picked up you'd get into the automation context and the voice menu you gave would start right away. Then if it turned out that first button meant that the user really wanted to make a call, then Asterisk would shift into the normal call dialplan, but reprocess that already pressed key as part of the phone number to dial. I think that we can provide something more intelligent when the phone is first picked up than your basic dialtone and not require extra button presses to get into the right mode. Your desire is for speed and so is mine. Steve Maybe I'm too far out on the edge of the bell curve, but I CAN remember what Alison was prattling about long enough to be told which key to press. Also, keys are much faster, especially once memorized as will happen quickly for something as frequently used as the TV. Think of a task like muting the TV: ) Off-hook ) Dialtone ) Press ** (change to remote mode) ) To control the... ) Press 1 ) To change the vol... ) Press 1 ) To mut... ) Press 0 ___ --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] phone as control interface (was 99 bottles of beer)
Steve Edwards wrote: Almost every room in my house has a phone -- if I could teach my kids to put them back where they belong. This could easily be extended to recognize which phone was used so it could control the Myth FE in that room. Also, it could/should be extended to control x10 devices as well... To control the tv in this room, press 1. To control a tv in another room, press 2. To control the outside lights, press 3. To control the sprinklers, press 4, ... A while back I was thinking along the lines of using a phone as a home automation interface, though I was thinking of it in combination with a voice recognitition system such as Lumenvox. It occured to me that when you want to turn the lights on, you don't really want to pick up a phone, dial a special extension, and then start using menus. What I was thinking about was what if instead of a dialtone you are brought directly to a home automation voice menu which works in parallel with your normal dial plan. If you wanted to make a call, just ignore the voice menu and dial normally. If you wanted to turn on the lights, just say lights on. or somesuch. Having a traditional dialtone seems unnecessary when you can get more function instead. The trick is doing this without giving up on the use of nice existing GUIs to manage the dialplan that we have now. I'd like some way of merging in the voice dialtone function with the existing dialplan such that initially both are active, but as soon as either a phrase is recognized or a button is pressed the system branches to one or the other, but that button or phrase is passed through to the rest of the processing and not just an extra prompt getting in the way. Does this spark anyone's imagination or ideas to implement? Steve ___ --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] phone as control interface (was 99 bottles of beer)
Steve Prior wrote: What I was thinking about was what if instead of a dialtone you are brought directly to a home automation voice menu which works in parallel with your normal dial plan. If you wanted to make a call, just ignore the voice menu and dial normally. If you wanted to turn on the lights, just say lights on. or somesuch. Having a traditional dialtone seems unnecessary when you can get more function instead. The trick is doing this without giving up on the use of nice existing GUIs to manage the dialplan that we have now. I'd like some way of merging in the voice dialtone function with the existing dialplan such that initially both are active, but as soon as either a phrase is recognized or a button is pressed the system branches to one or the other, but that button or phrase is passed through to the rest of the processing and not just an extra prompt getting in the way. Now that the idea is coming back to me a bit, here's a possiblity. When the phone is picked up it is auto-dialed into the voice driven/home control application AGI. At this point there are three options: 1. User utters a voice command. 2. User presses a touch tone which is meant for home control. 3. User presses a touch tone meant for the dial plan. option 2 vs 3 would be determined by internal extensions starting with a given number and dial 9 to reach an outside line, so other digits could be used for home control. As soon as option 3 is detected the voice AGI stuffs the touch tone back into the processing buffer, transfers to the normal diaplan, and exits. From there the normal dialplan handles the call normally. So, does anyone know if it is possible to stuff a touch tone event back into the processing stream so it can be handled by the new dialplan context? Steve ___ --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] LumenVox Speech Recognition
randulo wrote: Nitesh, I've messed with the Lumenvox starter kit. If you are serious about this field, I think it's a must see. It was easy to set up and there are demos available. Their support is excellent. There is a quiet mailing list where questions are never ignored and most problems are solved AFAIK. I'll agree that the Lumevox starter kit is a great starting point. I started out with the dialplan version of their pizza demo and quickly realized that using a dialplan would degenerate into spaghetti code very quickly. So using the Asterisk-java library I ported their pizza demo to Java (my code is now one of the demos on that page). I found this combined with Cepstral to be a great platform that could be used to write data driven voice recognition applications. If you are doing something interesting, the Lumenvox folks are very helpful in getting you going. Steve Take a look here for demos, etc. http://lumenvox.com/partners/integrator/digium/applicationzone/i /r ___ --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] asterisk setup for church / conference call / speaker system integration
Tim Litwiller wrote: Oh and we will also want to record the services so that if someone wants a copy or to listen later they can call in to listen or we can burn then a copy. So I'll need to program a recording menu system that is somewhat automated and lists the last few weeks of services by date or something. This part sounds like a podcast to me. Steve ___ --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] asterisk setup for church / conference call/ speaker system integration
Dean Collins wrote: Worth looking at using Talkshoe for this application maybe? My personal podcast authoring tool is vi (and you thought it was just a website authoring tool...) so I don't have any user friendly recommendations, but there's got to be a lot out there. Steve ___ --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] Microsoft's Move Into IP PBX Market
I'm sure everyone loves the idea of patch Tuesday for their phone system. Phones were too boring before, this will make them nice and exciting again... Steve George Pajari wrote: From c|net News: On Monday,Microsoft and nine leading phone manufacturers--Asustek Computer, GN, LG-Nortel, NEC, Plantronics, Plycom, Samsung, Tatung, and Vitelix--announced the public beta program for Microsoft Office Communications Server 2007 and Microsoft Office Communicator 2007. http://news.com.com/8301-10784_3-9719931-7.html?part=rsssubj=newstag=2547-1_3-0-20 ___ --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
[asterisk-users] Is STP wire decent for analog phones?
I've got a run of Shielded Twisted Pair (4 conductors) which used to be a Token Ring Network drop and I'm not using it anymore. Would it be decent to replace the ends with normal analog phone connectors and use it for a phone extension, or is STP unsuitable for that? Thanks Steve ___ --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
[asterisk-users] How well does a celldock work with Asterisk?
I've seen in the wiki that it is possible to use a celldock device to use a cell phone as a PSTN line to Asterisk, but I haven't seen any comments as to how well this actually works. I was thinking about hooking a celldock to a FXO input of my Digium TDM400P card and use it to connect via bluetooth to my RAZR V3C. I am aware of the software solution (chan_bluetooth), but my Asterisk box is a bit far away from where I want to keep the phone so the celldock seems to be the more convenient solution for me. Any comments about the sound quality or issues in making it work? Thanks Steve ___ --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] How well does a celldock work with Asterisk?
Steve Prior wrote: I've seen in the wiki that it is possible to use a celldock device to use a cell phone as a PSTN line to Asterisk, but I haven't seen any comments as to how well this actually works. I was thinking about hooking a celldock to a FXO input of my Digium TDM400P card and use it to connect via bluetooth to my RAZR V3C. I am aware of the software solution (chan_bluetooth), but my Asterisk box is a bit far away from where I want to keep the phone so the celldock seems to be the more convenient solution for me. Any comments about the sound quality or issues in making it work? I just found out that the celldock I'm talking about is also called the Dock-N-Talk. I look forward to hearing about experiences in using it with Asterisk. Steve ___ --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] Cepstral and numbers
Julian Lyndon-Smith wrote: Does anyone have any idea on how to force cepstral to convert a number to speech ? I have noticed that sometimes it speaks the number correctly, and at others it doesn't. 1) 787 is pronounced 7-8-7 2) 123 is pronounced one-hundred and twenty-three. 1) is wrong for what i need, 2) is perfect. Is there anyway of forcing numbers to be pronounced as 2) ? I've tried looking at the ssml tags .. TIA Julian Total guess on my part, but by any chance does it use 1) if the number isn't separated by whitespace from something preceeding it, but 2) if it is? This would make a traffic report for example read correctly: The traffic on I-84 is moving at 60 MPH. Where you'd want I-84 read using 1) and 60 using 2) Steve ___ --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] Cepstral voices
Lee Jenkins wrote:
Funny you should mention FastAGI. I am implementing a variation of my
DTSwift app through an Object Pascal based FastAGI scripting server now.
http://www.datatrakpos.com/pos/datatalk/images/asterpas.htm
The newer version just uses the System() AGI command to build the file
to play through the shell. I'd be open to any suggestions for a more
efficient way of doing it.
Sorry for not answering faster - was busy shoveling snow.
Once you've got app_swift installed, saying something
should be pretty close to:
AGI.Exec('Swift','This is something to say.');
And since it doesn't render to a file first you'll probably experience
less of a delay to say something.
Steve
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Re: [asterisk-users] Cepstral voices
Julian Lyndon-Smith wrote: what's the easiest way of using cepstral voices with asterisk ? On their website, in the ssml page (http://www.cepstral.com/cgi-bin/support?page=ssml), they say Asterisk PBX SSML can be used with Cepstral voices in Asterisk by simply embedding the markup into the input text. what input text ? To what application ? I agree completely with the app_swift suggestion from loopfree as Kai suggested. It provides the app_Swift which you can use from within a dialplan. In fact, if you're getting fancy by using a fastAGI bound language(as I'm doing with asterisk-java), app_swift becomes the only good option. slight rantI think Cepstral should be providing an app_swift like binding themselves because if you're writing an application which is going to use information from a back end business model in creating the speech (and this is something they seem to think is their future and I agree), then a high level language through fastAGI seems by far the best way to control the call. /slight rant Stay away from app_cepstral, if you're Googling it appears to be an option, but it didn't work for me. Steve ___ --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
[asterisk-users] Is Allison going to be banned from foreign travel over polar bears?
I read this story and thought of Allison's prompt to try not to think about blue eyed polar bears. Will she be banned from foreign travel now? Steve Prior -- snip -- WASHINGTON (Reuters) - Polar bears, sea ice and global warming are taboo subjects, at least in public, for some U.S. scientists attending meetings abroad, environmental groups and a top federal wildlife official said on Thursday. http://today.reuters.com/news/articlenews.aspx?type=topNewsstoryid=2007-03-08T222736Z_01_N08259521_RTRUKOC_0_US-POLARBEARS-SCIENTISTS.xmlsrc=rss ___ --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] Which Java FastAGI implementation has the most market share?
Matthew Rubenstein wrote: The real advantage in choosing an AGI (or CGI or ...) platform/language is *reusing* the existing code that already runs on that platform, with Well of course you should pick whatever AGI implementation matches the rest of your environment best. minimal porting to the platform in that language. How much does a Java application, net/bean, or modern (1.4-6.x) class have to be revised to make it work with asterisk-java as FastAGI instead of, say, AGI, CGI, commandline, browser JVM, or other execution environment/UI? I'm not totally sure you're asking the right question here. Asterisk-java in combination with Asterisk and in my case Lumenvox is just a user interface for whatever application I am developing. In my case it's not even the only user interface I've created for my system (which happens to be in Home Automation which uses CORBA to connect the pieces together) - I've also got a web interface as well as other standalone front ends and even the light switches can be considered part of the UI (and therefore non reusable). Asterisk-java provides you with an ordinary JRE environment where you might not be in direct control of main() (though you can be if you really want to), but that's similar to the other server environments you mentioned (browser JVM is a different animal). So the real question isn't so much how a class needs to be revised for asterisk-java, it is does your back end system provide a robust API such that you can be dropped naked in the middle of a JRE woods and without anything more than some additions to the CLASSPATH be able to interact with your back end system. Steve ___ --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] Which Java FastAGI implementation has the most market share?
Kate Kretz wrote: Steve, keep me in touch please ? We are also looking for moving all our activities to java platform. Let me know if You'll find/test something like asterisk2billing written in java ? I haven't received any feedback at all on the relative use of the java options, but I'm pretty happy with the way a little project turned out in asterisk-java. My project was to see how well asterisk-java would work in combination with Lumenvox to create a speech enabled AGI, so just for kicks I've ported their Pizza ordering demo to Java using it. In the process I've been working with Lumenvox to fix the couple of problems which turned up as a result of this experiment, and use an asterisk-java code change which is available in their latest svn. Sometime soon my code will be made available most likely through the Lumenvox site so others can use it as a starting point. Overall I'll say that I really like using Java to control such a dial plan. In this particular case the output is a simple pizza order which I've modeled as a plain old Java object (POJO), so once the dial plan has built up the object it can simply be passed to whatever back end (possibly J2EE) code which processes the transaction without regard for the user interface that created it. Sounds very maintainable to me. I did the development/test right in the Eclipse IDE and could use the debugger when necessary - I've got to believe that's better than trying to trace through a regular dial plan. I also really like the fact that aside from sound files and just a couple of lines of dial plan code to call the Java, all the actual Java code is running in a different server box so I'm keeping the load down on my Asterisk box and have flexibility in where I deploy things. Steve Cheers, Kate ___ --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
[asterisk-users] Which Java FastAGI implementation has the most market share?
When I was looking for a Java FastAGI interface for Asterisk I came across asterisk-java first and didn't realize there was more than one out there. It seems to work fine and I've got my first project working with it, but I was wondering which Java FastAGI implementation is the most popular and how they compare against each other. So I'm aware of: asterisk-java JastAGI OrderlyCalls Any comments on who the front runner is and why? Steve ___ --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] Nufone
Wiley Siler wrote: Are these guys still around? I cannot get to _www.nufone.net_ file://www.nufone.net or nufone.com Not only can I get to their website, but yesterday I called their customer service and for the first time ever it was actually answered by a live person. Steve ___ --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] FYI Panasonic Wireless Phone MWI
Noah Miller wrote: Last week I asked about MWI indicators on wireless phones that would work with Asterisk. I sent a message off to Panasonic asking them about it because in their ads they specifically stated that the indicator works with and requires phone company voicemail subscription. That indicator will not work for your voicemail. We do not have any phone system that has a message alert indicator that will work both for your voicemail and your answering machine. How exactly did you phrase the question to their tech support? If you described Asterisk as an answering machine then you'd get the wrong answer. If you described Asterisk as a PBX which provides a signal just like a telco voicemail would, then the answer would make sense. Steve ___ --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] TDM400P won't ring GM phone of mere 0.1B
Yuan LIU wrote: The feature request # is 4542, but I don't know any associated bug number, nor with what phones other people had to tweak. My phone is a GE 27935GE3-B. (Don't know what possessed me to say GM:-) Yuan Liu Just gotta ask - you did plug in the power supply connection on the board, right? Steve ___ --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] 5.8gig phone MWI
Tom Lynn wrote: You're trying to teach a pig to sing. The uniden items you refer to probably have their own internal answering machine, mine does. It's designed to light the lamp only when it's own machine has a message. You're giving out totally incorrect information. The TRU-8866 unit I mentioned is a 2 line unit (which I wanted), but does NOT have a built in answering machine (which I didn't want). Uniden seems to offer models with and without answering machine function. However, despite the fact that it does not have a built in answering machine, the handsets and base unit both support MWI. I believe that Uniden does make a single line version of this phone, and I bet it also supports MWI - especially since they've standardized their handsets to be universal across their latest line. Steve ___ --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] 5.8gig phone MWI
Doug Crompton wrote: Does anyone have personal experience with a 5.8gig wireless phone (system) that has an MWI that WORKS with asterisk via fxs (in my case spa3k) generated MWI. I know the spa3k does stuttered dialtone but not sure if it generates FSK MWI. I see some that state they do but I also see reviews that say they don't. Doug I've tested the MWI with the Uniden TRU-8866 phone and it works for me. I've tested it with the Digium TDM400P FXS. Steve ___ --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] Is there any Asterisk controllable thermostat?
Doug Crompton wrote: and it works great. Now I have one more way to control X10 devices. I can even call my VM on the way home and turn on my lights or whatever before I get home. Doug I've started to play with writing some code using the Java FastAGI interface to connect to my home automation system. The code is working and I could now write whatever I wanted, but I haven't figured out what would be a reasonable menu interface that wouldn't be very annoying to use. I'd be very interested to hear what menu structures and what actual capabilities people have found useful and nice to use. For example, has anyone come up with something less annoying than the following dialog: Press 1 for living room, press 2 for outside, press 3 for bedroom (I press 2) Press 1 for porch light, press 2 for garage light (I press 1) Press 1 to turn on, Press 2 to turn off, Press 3 to say current status (I press 1) congratulations, you just spent several minutes just to turn on a light! Steve ___ --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] Found GSM version, but any better WAV or ULAW recordings of Steve or Ian out there?
Lacy Moore - Aspendora wrote: Can anyone point me in the direction of a WAV or ULAW recording of those names? http://www.digium.com/en/products/voice/allisonsmith/ Thanks, but I meant a recording that already exists - I thought since both Steve and Ian already exist in GSM there might be ones around somewhere in WAV or ULAW. Steve ___ --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
[asterisk-users] Found GSM version, but any better WAV or ULAW recordings of Steve or Ian out there?
I'm looking for the best recording I can get of Allison saying Steve or Ian. I found gsm recordings of both out there but was looking for something higher quality. Can anyone point me in the direction of a WAV or ULAW recording of those names? Thanks in advance Steve ___ --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] FW: NuFone Update: DIDs
Rich Adamson wrote: At least outbound calls still work, even though they changed IP addresses (and probably colo locations). Maybe not so much now. I just got a disconnect notice from nufone which states that I have a positive balance in my account, but still need to add money to bring it up above zero for my account to be re-enabled. Somebody just broke the billing code I think... Steve ___ --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] [Asterisk-Dev] Peter Nixon to Speak at Cluecon
Brian West wrote: Peter Nixon will be making the trip to Chicago to speak at Cluecon, he'll be speaking on the topic of Real world deployment of Open Is there an echo in here? Steve ___ 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] TDM card and voicemail volume
Andrew Kohlsmith wrote: On Monday 27 June 2005 14:26, David Brodbeck wrote: There was a bug filed about this. It was marked as a wontfix, as I recall, because no one would step forward with code or $$$. Were either of those the actual reason given in the bug? Honestly, is it asshole day today or something? -A. Here is the text of the last 2 bug comments by MikeJ (who I would assume closed the bug). --- snip 1 --- Ok, somone put up a bounty for this or get a patch on here or there is really no point it keeping this open. Adding the code to raise the record volume on the other couple formats should be easy enough. Are we also seeing another issue here than the difference between the different formats still? --- snip 2 --- Ok, it seems no one is interested enough in this to fund the work. If anyone wants this to get resolved, please create a bounty on the wiki or contact an asterisk consultant to perform the work. This may be re-opened when there is a patch to review. end snip --- David's explanation was exactly right. Andrew, when you checked the bug comments and saw this, did you come to a different conclusion? Do you agree that a bug isn't a bug anymore just because nobody puts up a bounty to fix it? It seems a little more unreasonable to me that this would be the case as it was mostly reported as occuring with the supported product we paid for from Digium (the TDM400P), in part to support their continued development of Asterisk. Steve ___ 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] LOOKING TO HIRE
Iqbal wrote: and my $0.02 when u want something done quick u write a script...its amazing how these small things eventually grow into something big, in other words alot of what we have out here/there/everywhere started of in a small script on the back of a napkinif u wanna see what scripts can do...see the human genome project... A programming language is one of the languages you wish a program had been written in when the perl code you've been given to look at got too big and unmanageable. Steve ___ 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] Setting DID info for analog Zap channels
Is there any way to fill in the DID information for analog Zap channels? I've got a TDM422P and since I know the phone numbers associated with each of the 2 FXO channels I'd like to set that so that future extensions contexts can use it and the caller-id info in the form mydidnum/callerid like I can with VOIP DIDs. I haven't found if there is a variable that can be set with this info. If there is an easy way to set the DID information, then I believe I can provide a solution for some [EMAIL PROTECTED] users who are trying to make use of incoming call routing for Zap channels. Thanks Steve ___ 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] Can the originator of a call transfer it?
I've been getting familiar with call parking between a SIP extension and a Zap extension. I noticed that if either extension calls the other, then on the receiving phone I can press #70 to transfer the call to the parking extension. However, the phone that originated the call cannot do so. This is important to me because in a home setting I want to be able to make a call from one phone and then switch to using a different one to continue the call. Am I missing some bit of configuration that would allow this? Thanks Steve ___ 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] SIPURA SPA-2000 webserver dead after firmware upgrade
I just got a refurb Sipura SPA-2000 and was able to assign it an IP address with DHCP and ping the device, but then I ran the firmware upgrade utility to bring it up to spa2k-2.0.13g which seemed to work just fine, but after it rebooted I cannot connect to its webserver for configuration. I can still ping the unit. When I use the built in voice menu it reads back the right IP address, webserver port, and claims the webserver is enabled, but I can't connect to port 80 on the device and running the firmware upgrade utility says that it cannot connect to the unit either. Has anyone seen something like that and is there a fix? A google search didn't turn up any apparent hits. Thanks Steve ___ 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] SIPURA SPA-2000 webserver dead after firmware upgrade
David Boyd wrote: Run nmap against the ip address and see what ports are active for tcp service. Maybe you can connect via a different port (I know it should be 80), and see if the configuration is different between voice and web dave Good suggestion, unfortunatly I didn't like the results: root:~# nmap -p 1- sipura2 Starting nmap 3.50 ( http://www.insecure.org/nmap/ ) at 2005-05-10 22:24 EDT All scanned ports on sipura2 (192.168.0.32) are: closed Nmap run completed -- 1 IP address (1 host up) scanned in 3.427 seconds root:~# ping sipura2 PING sipura2 (192.168.0.32): 56 octets data 64 octets from 192.168.0.32: icmp_seq=0 ttl=250 time=0.8 ms 64 octets from 192.168.0.32: icmp_seq=1 ttl=250 time=0.8 ms --- sipura2 ping statistics --- 2 packets transmitted, 2 packets received, 0% packet loss round-trip min/avg/max = 0.8/0.8/0.8 ms Steve ___ 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] Sipura SPA2000 dialplan vs Asterisk dialplan
Trevor Peirce wrote: Steve Prior wrote: the SPA2000 does for me over the one in Asterisk. Is there a way to disable the use of the SPA2000 dialplan so I don't have to keep it in synch? Or is there some reason why it would be a bad idea for me to do so? Sure just put x. as your dial plan and any number will be accepted. The catch is you'll have to wait for the Short (Long?) Digit Timeout to pass before the call goes to asterisk for processing. If the SPA has an idea of what digit combinations are accepted it will wait until it has a match and send the call along at just the right time. No delays waiting the digit timer to expire. Thanks for the info. I also have an IAXy and it doesn't have anything like the concept of a dial plan in the ATA, but I've never noticed any kind of digit delay either - how long is this timeout on the Sipura? So does the IAXy just send digits as they come in whereas the Sipura tries to collect them and send them all at once? It sounds like a strategy might be to use the X. dialplan while I'm tinkering with the Asterisk side of things, and once I've got something I want to keep stable I'd make the Sipura dial plan more specific. Steve ___ 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] voicemail volume with sipura 3000
snacktime wrote: The problem is that voicemail messages are way too quiet. You can't hear enough to understand what is being said. If someone answers the phone before the 20 seconds the volume is fine. If I dial from the sipura line 1 into voicemail and leave a message the volume it just fine also. Chris Welcome to bug #2023... http://bugs.digium.com/bug_view_page.php?bug_id=0002023 Looks like a codec volume loss issue. Steve ___ 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] voicemail volume with sipura 3000
Tim Connolly wrote: Use wav, not gsm or wav49. /etc/asterisk/voicemail.conf ; ; Voicemail Configuration ; [general] format=wav While that workaround does fix the volume, it means the message as an email attachment feature is effectively useless because of attachment size issues. Same for retrieving the messages through the web interface unless you have lots of bandwidth between the user and webserver. A real fix to the bug I mentioned would cause all of these issues to be fixed. Steve ___ 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] Sipura SPA2000 dialplan vs Asterisk dialplan
I've got a Sipura SPA2000 ATA basically working (I can place calls between the extensions plugged into each of its ports) and part of that was setting up the dial plan on the SPA2000 to match the one in Asterisk. This seems like a pain to deal with long term and I don't know what exactly the dial plan built into the SPA2000 does for me over the one in Asterisk. Is there a way to disable the use of the SPA2000 dialplan so I don't have to keep it in synch? Or is there some reason why it would be a bad idea for me to do so? Thanks Steve ___ 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] Asterisk@Home bug
I disabled serial port login by commenting out the ttys0 line in /etc/inittab. Then after I rebooted the machine the message stopped. There is a way to make this take effect without rebooting the machine, but I was too lazy at the time... I don't know what the real cause is though. If you actually did need to login through the serial port you couldn't use my fix. Steve Manny A. Wise wrote: After installation of [EMAIL PROTECTED] v1, I have an annoying message in the screen, anyone know how to fix it INIT: Id s0 respawning too fast: disable for 5 minutes ___ 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] using * for Internet call waiting
Gary Carr wrote: Wondering if it is possible or if something already exist to setup * to offer Internet Call Waiting. For those that do not know what it is, it's a small application that runs on a users computer that will pop up a window letting them know they have a incoming call and who it is from then they can choose to take the call which will disconnect their dialup modem and ring their phone or send the call to voice mail. That doesn't really make sense if the * box is in your house because if the phone line is tied up for a dialup call, then the * box doesn't have a phone line to receive the call either (unless you had call hunting in which case you wouldn't need the feature in the first place). This sounds like the sort of feature that can only be offered on the central office side which can know your line is tied up and then know to email/alert you. The other scenario is having an * box in a call center that is forwarding calls to agents and notifies them by TCP/IP if when it tries their extension and gets a busy signal. This sounds possible, but I don't think it's what you meant. Steve ___ 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] Database lookups?
Jan Johansson wrote: Is it possible (How complicated is it?) to do this; IVR plays the usual please type your order number, finish with pound Then I would like to query a MSSQL database server, looking up the Status column from a row where ordernr = the entered order number. Depending on the result of the lookup, play one of two messages (Yes, ready for pickup or No, your order is not ready). Can someone clue me in on which docs I should start with? Or is there an example of this somewhere? If Perl is your thing then take a look at: http://ruk.ca/code/amazon.pl This is a script which plays a prompt message, accepts a number (in this case an ISBN), does a lookup based on that data, then plays a response. If you can hack the database lookup part yourself, then this script should give you a framework for what you need. Steve ___ 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] Backup for linux/asterisk
Jeff Glassman wrote: My question is as follows. Is there a backup program that will save to a tape drive or a USB CD Writer so if I mess up an install I dont have to go through a complete reinstall? I saw a few programs out there but they required X windows and from what I read it is suggested that X windows not be installed on an Asterisk box. I recently used G4U from: http://www.feyrer.de/g4u/ See if it does what you need. Steve ___ 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] IAXY Polarity
Yesterday I was using one of the cheap Radio Shack phone polarity on various phone outlets in my house and ended up plugging it into my IAXY. While the regular phone jacks tested OK, the IAXY tested as being reverse polarity. The tester was plugged directly into the IAXY so there is no chance of this being caused by crossed wires. Is this something that is misconfigured on the IAXY, a hardware issue with the IAXY, or just not really a problem anyway? Steve ___ 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] About the weather..
Matt Riddell wrote: dean collins wrote: There is a script on the [EMAIL PROTECTED] sourceforge list that reads the weather for you. Basically ftp's a text file from the BOM and then uses festival to read it out to you :) Yeah but he doesn't want to use festival. He wants to use the recorded prompts by Allison. I guess you could parse the text to look for cloudy/sunny/raining etc and then use that to form the playback statement. The festival approach is required when you use the National Weather Service text forecasts which are less formally written and meant to be read by a human - I did a survey of the kind of language used in them a while back and they contain all sorts of fuzzy language which is hard to interpret by computer. The recorded prompts by Allison are more in line with the very language structured text forecasts typically seen by pilots - I'm not sure what feeds for this type are available. Also the National Weather Service now has highly structured XML forecasts available which while being more work to deal with the XML should be much easier to automatically interpret and use the appropriate recordings. I believe they use a SOAP API for the data and there was a bit of a stir when they first started making this data available from other companies who charged for this kind of information and didn't like the idea of the government freely distributing what the citizens had already paid for (no bias here...). I looked at working with this data feed a while back, but didn't have time then and haven't gotten back to it yet. Steve ___ 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] About the weather..
Wolfgang S. Rupprecht wrote: [EMAIL PROTECTED] (Steve Prior) writes: The recorded prompts by Allison are more in line with the very language structured text forecasts typically seen by pilots There are home weather stations with computer interfaces that simply tell you the current stats (temp, pressure, humidity, wind direction, wind speed, rainfall rate). Converting this information to something that could use the asterisk sound snippets wouldn't be that hard. -wolfgang You're right (I even have a weather station on my roof), but I thought the discussion was more about the forecast than the current stats. Steve ___ 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] Wildcard X100P or TDM400P?
Steven Critchfield wrote: TDM400P with FXO daughter card includes 1 hour of Digium support. It is supposed to support other line types. If you have trouble, it is likely you will get direct support from Digium and from the community here. It should be noted that several people including myself are having voicemail volume problems with the TDM400P (is anyone having it with the X100P?) which for us makes the card unusable for what was intended (a basic home PBX/answering machine). This is documented in bug #2023: http://bugs.digium.com/bug_view_page.php?bug_id=0002023 You'll also notice that it has been dormant for quite some time. Steve ___ 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] festival text for weather report
dean collins wrote: Hi Ernie, Man I hope you didn't write all of that for me, I feel really bad now, someone posted to the list about 15 mins after I posted with the solution lol- I've already been playing with it for hours working out what other sites I can get it to read from as well. Thanks anyway - good practive I guess. I'm modifying the festival wikki page once I work out how. Please add what you ended up with sooner rather than later. I'm interested in the same weather forecast as you (in fact I'm in the same forecast region as you), but I've also been wondering what's around in terms of traffic information. http://www.hudsonvalleytraveler.com/perl/IncidentCongestionRpt.pl Might be of interest to you as well, but unfortunatly they only provide the data in HTML format - I've already emailed them about how easy and useful it would be to provide XML as well... My future hope is that I'd set up a DID that I could call from my cell phone headset using the voice dialing built into the phone, and then get a traffic report read to me without pressing any phone buttons. That either means using a dedicated phone number or getting even a very primitive speech recognition going to chose between a very limited number of options. The speech recognition part seems like it's going to be a while (I'd love it if someone else figures this part out before I get there...) Steve ___ 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] ANNOUNCEMENT : NEW CallingCard Application forAsterisk
Sergey Kuznetsov wrote: Robert, It is better to stay with Postgres. If you don't want to loose your business stay away from MySQL. regarding MySQL and Postgres. I would say Postgres is a Open Source Oracle. It's very stable, very scalable and it's perfectly works under serious workload. MySQL is dying at the same configuration. I have client of mine who having issue with MySQL. Under some workload ( 10 users inserting at the same time ) it corrupts the index. Even MySQL 4.0.X is still corrupts the indexes under heavy load. All the Best! Sergey. There might be reasons to prefer Postgress over MySQL, but I find it hard to believe that scalability is one of them - I mean we're talking about the database that runs Slashdot which is so scalable that users reading it routinely take down other websites with the load. Steve ___ 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] Your Acerbic Tyrant will be off line for about 10 days
Race Vanderdecken wrote: Greetings List, I know many of you are looking for advice from me but I am moving from the 28th until about the 4th of February. Race Vanderdecken Hmm, let me check my schedule... Sorry, no that doesn't work for me - you'll have to reschedule. Steve ___ 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] UPS for Asterisk
One word of caution in case you have X10 equipment. I recently found out the hard way that some of APC's newest UPS models will cause interference with X10 signals going over the powerline. I'm not talking about the X10 signal not going through the UPC - that would be expected. I'm saying that in my case it interfered with X10 signals elsewhere in the circuit the UPC was on. Plugging the UPC into an X10 noise filter solved the problem. Steve steve szmidt wrote: The days of shoddy UPS's are long gone, unless you always buy the cheapest stuff you can find all the time. In which case you might be able to find something crappy. APC gives good support and make decent UPS's at a decent price. ___ 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] UPS for Asterisk
Andrew Thompson wrote: Steve Prior wrote: I have an X10 dimmer switch in my bedroom. Initially, it operated fine, no troulbe to speak of. Then, all of a sudden, it started randomly turning on the main room light in the middle of the night. I didn't notice this for a while, mainly because it doesn't bother me unless I'm already awake. But my wife mentioned that it wakes her up and she has to get up to turn it off. (The remote switch seems to have give out, but that could be the battery.) I am remembering now, that one day I got mad at the power blinking out so I brought in a heavy duty (well, at least heavy, two part, average geek would only want to move one piece at a time) UPS for my asterisk box. Could this be a symptom of the interference you spoke of? What filters have you used? Thanks. My problem was of the modules aren't receiving signals anymore variety which would indicate that something (in this case the UPS) was putting out noise on the power lines, not what you experienced. You might have a module that's getting flaky or something else is generating signals. The filter I used had been siting in my closet for a few years, but I probably bought it from smarthome.com and they have a couple of them. Steve ___ 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] UPS for Asterisk
steve szmidt wrote: On Monday 24 January 2005 12:12, Steve Prior wrote: One word of caution in case you have X10 equipment. I recently found out the hard way that some of APC's newest UPS models will cause interference with X10 signals going over the powerline. I'm not talking about the X10 signal not going through the UPC - that would be expected. I'm saying that in my case it interfered with X10 signals elsewhere in the circuit the UPC was on. Plugging the UPC into an X10 noise filter solved the problem. Steve That's interesting. Good fix too! I suspect that might not be all too uncommon as they all generate tones for the frequence. Have you tried it with a few different UPS's? I do have other UPS's in the house that didn't cause a problem, but there are lots of factors (X10 is a pretty hard thing to make reliable on a good day). I've had all sorts of stuff cause noise on the line including a breadmaker and some old IBM displays. If you're serious about X10 for home automation it's a good idea to have a few noise filters around for things that come up. I only mentioned this on this list because I thought the same sort of person who might put a PBX in their house probably has some X10 there too unless they did something else very high end. Steve ___ 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] Help! - Unintelligible prompts and music
AHBLWEB wrote: Aha! Remove the Digium card and everything sounds fine. Leaves me with a SIP-only server though. Looks like I'd better RMA that sucker. I had a similar problem with a TDM11B - even VOIP calls to the Digium demo server were broken up when the card was in and the FXO and FXS ports didn't work, but VOIP was fine with only the ztdummy driver. I drove myself NUTS for over a weekend trying to get it to work. Once I got a replacement TDM11B card everything started working normally (except for the low volume voicemail problem). BTW, the modules on the original card were fine (I still have them), it was just the backplane card that was bad. Steve ___ 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] Is asterisk that unstable ????
[EMAIL PROTECTED] wrote: I like to joke that Microsoft uptime is measured in hours Unix/Novell is always in years,months, and days. It's not just you. A while back Microsoft was running a TV ad where a server was bragging that it was so reliable that it hadn't even seen the sysadmin for DAYS. Can you imagine what would have happened to a Unix company who ran the same ad? Everyone would be laughing their butts off... Steve ___ 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] Dialtone for Software phone?
Lane wrote: Hi, Is it possible with asterisk to deliver a dialtone to a software phone, such as kphone? I'm able to dial, but the silence seems to confuse my users :) Tell it's a software cell phone. NEXT!!! Steve ___ 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] caller id NUMBER in addition to or in place of NAME
Eric Wieling aka ManxPower wrote: I seem to recall a bug regarding this. Are you using 1.0.3, 1.0.x CVS STABLE, or CVS-HEAD. The problem is that what was listed in the voicemail.conf.sample as the default e-mail message was, in fact, not the default e-mail message. Uncommenting out the example message fixed the problem. Yup, you got it right. I chased that issue once. Steve ___ 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] caller id NUMBER in addition to or in place of NAME
I just checked the code and didn't see any way to change the pager email body or subject like you can the regular email notification - only the from string. If you wanted to hack it, the messages are in apps/app_voicemail.c on or about line 1033. Steve Dorn Hetzel wrote: Now the regular emails have the name and number ... However, the pager emails still have name only... Is there a string variable to redefine the subject/body sent to the pageremail? Regards, -Dorn ___ 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] Predictive dialer
mattf wrote: Hello, The asGUIclient suite has a predictive dialer component to it(VICIDIAL) and it can function well on multiple Asterisk servers at once using a single MySQL server backend. It performs on par with several mid-level commercial dialers that we have compared it to(Nobel, TripleP, DataTel, etc...) and it is free as in GPL. We released a new version of the suite last week with many enhancements. Before I found Asterisk I never heard the term predictive dialer before. Is it the feature used to make annoying telemarketing calls? Does it have any popular non-annoying uses? Steve ___ 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] Asterisk in parallel with PSTN
Andrei (MPI) wrote: richard wrote: Asterisk also ring (so that all three of them are ringing), and then someone can then choose which phone they want to answer? Hi Richard, Absolutely, you can do that with Asterisk. Though VoiP telephone (Asterisk) may start ringing a second later than analog phones connected to the line directly. I do not think that would be a problem. I believe you should note that if Asterisk is configured for callerid that the SIP phone won't ring until your non-Asterisk phones have rung twice. This is because Asterisk gets the CID before it passes the call through. I'd like to be wrong about this, but I believe that's what I've been told. Steve ___ 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] Another Asterisk Certification
[EMAIL PROTECTED] wrote: Man, this is sick! :-))) Isn't there a law against unclearly-marked jokes (except on April 1st, of course)? Some people could even take you seriously! :-)) I'm rolling on the floor here :- Regards, Telmo. Then I guess you haven't seen this one: http://www.j-walk.com/other/conf/ Steve ___ 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 demo the Power of Asterisk
Adam Goryachev wrote: Hi all, I have the opportunity to demo asterisk to a large group of people, and was just wandering *how* to do that? ie, I can put a couple of phones on a desk, which looks nice, but doesn't really look exciting, because they are just phones so, how do you 'demo' the true power of asterisk?? Personally I thought the AGI script on the Wiki that allows you to key in the ISBN of a book and have Asterisk read back it's current price on amazon.com to be pretty cool. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bluetooth with *
Jon Radon wrote: That's useful for covering a larger area, but not really for pinpointing a users location. BTP is intended to be used with multiple bluetooth sensors/adapters. It determines where to send a call by which sensor has a greater signal strength to your bluetooth device. My cell phone (Mot V710) is discoverable for only 90 seconds when I actually tell it to be. Will BTP work by pairing the BT adapter to the phone and then polling to see if the paired device is there? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax pass-throught.
Alessandro Ren wrote: I've found the fax extention setting, but this is not what I want to do. I'd like to dial from the line on the other side of the IAX channel to a fax, to cut long distance costs, and send a FAX from the source IAX channel. Like bellow: source destination FAX --- Asterisk --- internet --- Asterisk --- external line - PSTN -- FAX I haven't done this, but I've heard that faxing through a voip connection is problematic. Have you considered the possibility (if you control both Asterisk installations in your diagram) that you could fax to a virtual fax on the source Asterisk system which would capture to a file, email or file transfer the image to the other Asterisk box which would then dial out and send to the final destination? This assumes your source and destination actually have to be real fax machines, otherwise you have even more options. Check out: http://scottstuff.net/scott/archives/000152.html and see if it gives you any ideas. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Low Volume WAV Files in Email Attachments
Philippe Daoust wrote: I have read several posts regarding this problem but can't find one with a solution... I see the same issue: Voicemails picked up by handset have normal volume, but voicemail sent as a wav attachments in email are so low they are barely usable... Is there a way to fix the volume before they are emailed out? Thanks for any tips. Sign up and join the fun in bug #2023... http://bugs.digium.com/bug_view_page.php?bug_id=0002023 I'd also recommend emailing support at digium - not because you'll get anywhere, but to keep them aware that people are interested in getting this fixed. I've got the same problem and emailed support last week and got very little to encourage me that it's any kind of priority. What are you using for telephone hardware? Your situation is a little different in that you say the voicemails play back at decent volume over the phone - are you testing with voicemails left through a POTS line connection in all cases? I'm pretty new myself, but I've got C skills and am planning to get back to looking over the code myself soon. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] vm notification no longer contains calling party
I noticed that the commented out mail message in voicemail.conf isn't actually what is the default. I fell into the trap that because it was there as a sample and commented out that I assumed it was actually the default and it's close, but not exactly. Maybe the default behavior actually changed and should be changed back, but actually putting it in your voicemail.conf from the sample file is the fix. Hope this helps Steve Prior Jason Peck wrote: Hello all, I recently updated * to 1.0.x from a CVS version downloaded in July, at the request of BroadVoice. Every since the upgrade, the email notifications that a voicemail has been left only contains CIDName and not CIDNum. Here's an example: I've installed 1.0.0, 1.0.2, and the latest CVS, and they all seem to be having this problem. I've searched through a couple months of the mailing list, and briefly in the bugdb, but I haven't been able to find anyone else that seems to be having this problem as well. Has anyone seen this before? Is it a known bug? Might I have something misconfigured? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] vm notification no longer contains calling party
Jason Peck wrote: I take it that you are talking about the commented out ;emailbody= line? I guess that could be. I just figured that since it was: Yup, that's the one. Looking at app_voicemail.c there is a line that says: pbx_builtin_setvar_helper(ast, VM_CALLERID, (callerid ? callerid : an unknown caller)); The pbx_builtin_setvar_helper function, as well as how it's called in app_voicemail.c, is the same in my old and new versions. I can't find exactly where the CDRs are written, but one would assume that it uses a similar function to get CIDName and CIDNum. --jason At first I wasn't that curious once I got it fixed by uncommenting the emailbody line in voicemain.conf. I still think it is a little weird that the commented line in the sample file doesn't match what the default really is, but I'll get over it. Since you seem to be more interested in the details I took a look through the CVS histories and found where the change really happened (it wasn't caused by the line you mentioned). Between revisions 1.151 and 1.152 of app_voicemail.c the lines: fprintf(p, Dear %s:\n\n\tJust wanted to let you know you were just left a %s long message (number %d)\n in mailbox %s from %s, on %s so you might\n want to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n\n, vmu-fullname, dur, msgnum + 1, mailbox, (callerid ? callerid : an unknown caller), date); were changed to: fprintf(p, Dear %s:\n\n\tJust wanted to let you know you were just left a %s long message (number %d)\n in mailbox %s from %s, on %s so you might\n want to check it when you get a chance. Thanks!\n\n\t\t\t\t--Asterisk\n\n, vmu-fullname, dur, msgnum + 1, mailbox, (cidname ? cidname : (cidnum ? cidnum : an unknown caller)), date); And this is what caused the new behavior. I happen to think that the sample file should also have been changed to reflect the new default to avoid confusion. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Patching asterisk for spandsp
I just ran into this last weekend. I believe that you are using a version of spandsp which is for an older version of Asterisk. First I had to use the most recent version of spandsp from the source (if there is anything later by the time you get there use it): ftp://ftp.opencall.org/pub/spandsp/spandsp-0.0.2pre6/ Instructions on how to add it to extensions.conf are also a bit changed from what you probably saw in the wiki. I found the following site to do the trick: http://scottstuff.net/scott/archives/000152.html All in all his approach works well. I had to find and install mime-construct, and had to remember to create the directory under /var/spool where the faxes would go (he uses /var/spool/asterisk-fax but I prefer /var/spool/asterisk/fax) but once I did that I started getting pdf emails. The thing that he doesn't seem to do is to clean up those fax files after a while. Good luck Steve Eric Rees wrote: When I try to patch the Makefile for asterisk with the Apps_makefile.patch from Spandsp I get the following error. patching file Makefile Hunk #1 FAILED at 47. Hunk #2 FAILED at 76. 2 out of 2 hunks FAILED Has anybody seen this. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Patching asterisk for spandsp
Steve Prior wrote: I just ran into this last weekend. I believe that you are using a version of spandsp which is for an older version of Asterisk. ... Good luck Steve I forgot to mention that I didn't yet get fax tone detection working, my success so far has been setting up an extension for spandsp which I am calling from my fax machine on another extension. The fax machine is hooked up to a fxs port on my TDM22B. I haven't tried it yet, but I believe that the fix is in adding one of the following to zapata.conf: ;faxdetect=both ;faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no I also think the following are needed: echocancel=yes echocancelwhenbridged=yes Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Patching asterisk for spandsp
Gregory Junker wrote: Just for sanity's sake, I went back and read the README on the site again, and it does say: Add the files rxfax.c, txfax.c and dtmftotext.c (the last one has nothing to do with the fax machine, but my makefile patch expects it to be present) You have to grab the dtmftotext.c file as well, which also is not part of the tarball. That could be the problem. Greg I found the dtmftotext.c wasn't needed for the most recent version of spandsp on the ftp site, and I also didn't have to modify the Makfile with the path where to put the fax files - it is now set in extensions.conf instead. Examples of this are at scott's site in my original note on this. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Patching asterisk for spandsp
Michael Welter wrote: echocancel=yes echocancelwhenbridged=yes Steve Underwood says not to use echo cancel on a fax line. Mike Oops, you're right. I knew I was not supposed to use echocancel, but somehow got these two lines backwards. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recording from AGI playback is LOW
Jerry Geis wrote: I am doing a record in my AGI. When I play it back at that moment on the phone it sounds just fine. When I do a play file.gsm it is REAL LOW on the soundcard When I send dial using IAX2 to another asterisk box and play it on Zap/2 it is low also. When I play demo-congrats.gsm on the soundcard it is just fine. Any ideas why or what to do? I'm chasing a similar issue (normal sound levels for calls and playback of canned recordings, but very low voicemail volume) and noticed that when I leave a voicemail the uncompressed WAV file is at normal volume but the 2 compressed versions are very low volume. Apparently under the covers the system uses sox to create these compressed files. My current theory is that my version of sox is old (12.17.3) and that maybe that level of the software has volume problems when converting files. Tonight I plan to upgrade to a new version and see if the problem goes away. As an interesting data point can you do a sox -V and report your version number? If I'm right this may solve both of our problems. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recording from AGI playback is LOW
Steven Critchfield wrote: but the 2 compressed versions are very low volume. Apparently under the covers the system uses sox to create these compressed files. No it doesn't. The uncompressed version is also getting a double amplification by shifting the bits up before saving and all the compressed versions aren't doing the same thing. I know this as I have done work in the formats directory. OK, that's good information in the right direction, but then what should be done to fix it? Am I the only person having problems with low volume voicemail recordings? Is there a code patch to apply somewhere? Thanks Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recording from AGI playback is LOW
Steven Critchfield wrote: OK, that's good information in the right direction, but then what should be done to fix it? Am I the only person having problems with low volume voicemail recordings? Is there a code patch to apply somewhere? As far as I know there is no code patch to be applied. The question comes basically when the recordings go off the PSTN. In the uncompressed WAV format, the audio is downshifted for playback. This means it is played at the same volume as any of the compressed formats. The question then I guess is maybe we need to be able to specify a gain setting and deal with it before we route the audio to the audio writers. But then again, my memory about the internals is fuzzy and not up to date right now as to whether that is doable. Maybe the gain setting in the uncompressed wav format should be removed. It seems to me that the uncompressed file has it right here - if I playback the uncompressed file from winamp it is just a little low, but in the ballpark. If I do the same for the compressed files it is very low. I would think the goal would be for the volume in all of the files to be playable without cranking up the volume. But something about all of this seems a little odd to me since I'm new to Asterisk and I would assume if this is the cause that it would have been brought up and solved a long time ago. So I ask again - when live conversation is generally at a decent volume is it common for voicemail recorded messages to be low(er) volume? When other users have Asterisk email the recording do they generally have to raise the volume controls on their computers to hear it? Thanks Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Avoiding 2 ring callerid delay for calls that don't go to voicemail
I have my dialplan configured for an incoming call on the FXO to connect right through to a FXS on the TDM100P. Because of the callerid the calling party gets 2 rings before asterisk picks up and then it's another 2 before the caller id shows up on the analog phone connected to the FXS module. I understand that the 2 rings are needed to collect CID, but is there any way to tell asterisk that since it's passing right through to the phone to streamline the process to 2 rings total instead of 4? People will almost be at the point of hanging up when I've just seen the CID at the phone for the first time. The only time the callerid would be used by Asterisk itself would be if the call goes to voicemail. Any hints? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is IAXTEL working?
As a test I'm trying to call the Asterisk shipping dept at (700) 428-6004 and have had no success. Is IAXTEL working? I have had no problems with the Inter Asterisk connection part of the Asterisk demo. Is there anything wrong with what I've got below? Here is the log (password substituted of course): -- Starting simple switch on 'Zap/2-1' -- Executing Dial(Zap/2-1, IAX2/sprior:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Called sprior:[EMAIL PROTECTED]/[EMAIL PROTECTED] Nov 15 22:18:45 NOTICE[24211]: chan_iax2.c:2199 auto_congest: Auto-congesting call due to slow response -- IAX2/69.73.19.178:4569/2 is circuit-busy -- Hungup 'IAX2/69.73.19.178:4569/2' == Everyone is busy/congested at this time == Auto fallthrough, channel 'Zap/2-1' status is 'CHANUNAVAIL' -- Hungup 'Zap/2-1' I've got the following in my iax.conf: register = sprior:[EMAIL PROTECTED] And the following in my extensions.conf: [globals] IAXINFO=sprior:password [iaxtel] exten = _1700NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten = _1888NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten = _1877NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten = _1866NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) exten = _1800NXX,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) Thanks Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3 - TDM31B Card Installation Difficulty
Begumisa Gerald M wrote: o I can call the analog phone from X-Lite however on receiving, I cannot hear much voice. What I hear is choppy sound corresponding to whatever I say from the analog side. When someone speaks from the X-lite side, nothing is heard from the analog phone. If you call from X-Lite to the demo menus can you hear them clearly (no choppy sound)? Given the problems you are having this might point to a bad TDM100P card. I recently had to swap out a TDM100P card (rev H) for a replacement (rev G) card because the card apparently wasn't supplying correct timing for Asterisk. This even affected X-Lite to demo calls where the FXO and FXS modules weren't even being used. The way to check is to switch to the ztdummy driver instead of the TDM100P driver and see if the X-Lite - demo calls become clear. I don't know if was a defective card or a REV H issue, but now at least working. The other symptoms in my case were no dialtone through the FXS card and the FXO card not answering incoming calls. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 3 - TDM31B Card Installation Difficulty
Begumisa Gerald M wrote: Mmm. I have a spare one. I'll replace the one that doesn't give dialtone and see what happens. I'm not jumping to any conclusions, but please note the revision of the card you're taking out and the rev of the card you're putting in. In my case the modules on the card were perfectly fine - it's the backbone card itself that supplies the timing and had the problem. Thanks alot Steve. I'll fix the card and let you know what happens. I'm very much a newbie myself, but I seem to be (only) a couple of days ahead of you :-) Rgds, Gerald. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WAV file volume in voicemail - anyone actually solve this?
I have seen several posts asking about how to increase the WAV file volume for voicemail recordings, but haven't really seen a definitive answer. A while back Mark Spencer mentioned a #define GAIN as a possible solution, but I haven't seen if that has been made the official solution (and does anyone have a suggested value - I see the default is 2). Has this problem been solved or have an official workaround? Thanks Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
