Re: [asterisk-users] 811
Quoting Mike Diehl mdiehlena...@gmail.com: Is there a list somewhere? There is a list by state here: http://www.call811.com/state-specific.aspx -- _ -- 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] MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am
Quoting Tim Nelson tnel...@rockbochs.com: Do you have any sort of site/mailing list/etc setup to facilitate this group? I'd be interested in attending such a meetup in the future. http://www.tcaug.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] Not Dialing 9
Quoting Thczv F. Thczv thczv.th...@gmail.com: When I set up my Asterisk box at home I didn't want to have to dial 9 to dial off premises, so I gave all my local phones three digit extensions with this format: 1[1,0]*. My thought is that there are no area codes that start with 0 or 1, so if I use those numbers, I can create 20 local extensions that can be dialed with 3 digits, and not have to use a timeout when dialing long distance. If I dial 1, then anything other than 0 or 1, Asterisk knows I am dialing long distance. If I start with any number other than 1, Asterisk knows I am dialing a local or local toll call. In North America: 0 is the intra-lata operator 00 is the inter-lata operator 0+ something else will be an operator assisted call 11xx is used for the rotary dial equivilant of *xx on many central office switches. Assuming you are not using rotary dial, I generally use 4 digit extensions with the 11xx format for the same reason you suggest. --Shane ___ -- 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] Voicepulse down
Quoting Fred Posner f...@teamforrest.com: Starting around 10:00 AM EST. All services from them whether I connect by IP or DNS (both east coast and west). Anyone else? Yes, I'm experiancing the same problem. Their www.voicepulse.com and connect.voicepulse.com seem to be offline as well. --Shane ___ -- 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] Seemingly easy question: NPA/NXX
International numbers are variable length, so the timeout applies for those. North American National numbers are a fixed length. Generally, the phone company will collect 7, 10 or 11 digits for North American numbers. For example, I live in Minneapolis, MN. My number is 612-xxx-. I have free calling to 612, 651, 952, 763 and a few numbers in 507 and 320. If I dial 1, the phone company will collect 10 more digits. (The call may or may not go through if I dial 1+ a 10-digit local number depending on the carrier. MN regulations prohibit charging for local calls dialed as toll) If I dial 612, 651, 952, 763, 507 or 320, the phone company will wait for the remaining 7 digts as there are no numbers within area code 612 that start with those digits. Anything else will only be collected as 7 digts and assumed to be 612. Because of that, I can't dial a california number (for example), without dialing it as 1+. I wouldn't call it fancy, the phone company just knows what is a valid local number for you. Making a digit map in an ATA isn't that hard, you just need to think about what you want it to do. If you want to permit 10 digit dialing without the 1+ for long distance *and* support 7 digit local dialing, you'll need a timeout. There are also the N11 numbers, which of course should stop collecting after the second 1. --Shane Quoting Karl Fife [EMAIL PROTECTED]: Question: How does the local Telco know you're done dialing a seven digit number? Easy you may say: If your dial string begins with 1, the parser expects 11 digits total, otherwise seven, 011 is international. The reason suspect it's more complex is that: 1) International numbers can vary widely in length and 2) Our local analog Telco will route a ten digit NANP numbers with no leading 1 and with no terminator--seemingly instantly Obviously this could be done with 'timeouts'--implicitly 'sending' after a delay. But it works so well I suspect there's more logic in there. For example I have dozens of ATA's provisioned with timeouts, and I find it difficult or impossible to replicate the Telco dialing experience (Either the delay is too long, or you have frequent 'reorder' tones because it 'sent' before you were finished). Therefore I assume that there is something more 'fancy' going on. Can someone validate, debunk or clarify this? Theory 1 Is it all done with timeouts, but they're CONDITIONAL timeouts. i.e. give a LONG timeout if the number: -did not start with a 1 and is still shorter than 7 digits, -started with a 1 and is still shorter than 11 digits -started with a 011 and is shorter than the theoretical international minimum lenght Theory 2 As you know, a few years ago the 2nd digit of the NPA was always 1 or 0. Therefore the switch could easily determine(without the leading 1) if your first three digits were an NPA or just an NXX (exchange). They were nationally unambiguous. Now that's no longer true. STILL, it could be possibleto consider all known valid NPA's and exchanges so they can determine via context what you're trying to do, and thereby optimize the dialing experience? Can anyone speak to this? I would very much appreciate any knowledgable input. -Karl ___ -- 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] Broadsoft Sip provider
Quoting Gustavo A Gonzalez [EMAIL PROTECTED]: I am looking for a sample sip configuration of a SIP provider that runs Broadsoft VoIP switch. This is what I use: register = 3115552368:abcdefghijklmnop:[EMAIL PROTECTED]/3115552368 [broadworks] type=peer host=1.2.3.5 dtmfmode=rfc2833 outboundproxy=1.2.3.4 fromdomain=1.2.3.5 fromuser=3115552368 username=3115552368 authname=3115552368 secret=abcdefghijklmnop canreinvite=no disallow=all allow=gsm allow=g726 allow=ulaw qualify=yes insecure=port,invite context=inbound ___ -- 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] Adit 600 password reset
Quoting Doug Lytle [EMAIL PROTECTED]: C F wrote: Then there is basicly no way to do this besides for cracking it? I Not that I am aware of, no. This subject went around several years back. They also talk about brute forcing the password as well. As far as I recall, nobody came back saying they were successful. have already figured out the username, now I just need to figure out the password. What is a good screen automation program that can bruteforce this for Windows? I had the same problem with one of mine. I smply forgot the password. I seem to recall that the adit had a flaw in it, where it was obvious by the error message returned if you had the correct length username and password, which should make your brute-force attempt much easier. --Shane ___ -- 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] Show Callee name on Display
It would be possible if Asterisk sent a remote-party-id back to the calling phone. Polycom and Sipura phones (possibly Cisco phones) Support this with SIP on Broadworks and it works great. --Shane Quoting Eric \ManxPower\ Wieling [EMAIL PROTECTED]: It is not possible to do this the way you want. Most phones will display the called name if that name/number is in the phone's directory. Peder @ NetworkOblivion wrote: We have users with Cisco 7900 phones running sip. When user A calls user B, we want user B's name to appear on user A's phone. It shows the extension they call, but not the internal name of the called user. Is this possible? We have some people that used to be on an MGCP based system and they would get the callee's name popup on their phone when they called someone. I can't figure out if it is possible or if it is just a limitation of the Cisco SIP firmware. Just to clarify with an example: 1 - Steve 2 - David David calls ext 1. Right now it says calling 1. We want it to say calling Steve 1. ___ 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 --Shane ___ 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] Show Callee name on Display
I don't think that Asterisk currently sends a remote-party-id to the called party. That would proably have to be added to the sip channel. It *does* work with Broadworks, another SIP based phone system. On a phone registered to Broadworks: Your phone invites the Broadworks system, Broadworks replies with a 180 (ringing) which includes a remote-party-id: field populated with the destination you are calling. That is what displays on the Polycom and Sipura 841 that I have tried. I had eneabled remote-party-id on a Cisco 7960, but something in the dialog caused the call to die. I never investigated further. --Shane Quoting Eric \ManxPower\ Wieling [EMAIL PROTECTED]: Have you ever actually done this with Asterisk? Shane Young wrote: It would be possible if Asterisk sent a remote-party-id back to the calling phone. Polycom and Sipura phones (possibly Cisco phones) Support this with SIP on Broadworks and it works great. --Shane Quoting Eric \ManxPower\ Wieling [EMAIL PROTECTED]: It is not possible to do this the way you want. Most phones will display the called name if that name/number is in the phone's directory. --Shane ___ 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] Show Callee name on Display
It's all priced by quantity of each feature you license, number of users, number of concurrent calls, things like that. Previously it only ran on Solaris. It now also runs on Linux. I wasn't involved with our initial purchase, but I couldn't imagine you could have a working system for less that 100k (not including hardware). Normal broadworks systems include: 2 Application Servers N+1 Network/Routing servers N+1 Media servers In addtion to the software, you'd need to purchase the hardware and OS to run it on. It will do SIP or MGCP on the user side and SIP on the back-end/PSTN side. It doesn't support any telephony hardware directly (nothing like zaptel). It just does SIP or MGCP. You'd need to connect to something that will get you back to the PSTN (either your own hardware or a provider) Quoting Seysan [EMAIL PROTECTED]: Hello, Does anyone Knows the price of the Broadworks? any idea? Seysan --Shane ___ 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] Nufone problems
Quoting C F [EMAIL PROTECTED]: Anybody here having any problems with nufone? Calls are not going thru, when trying to call their customer service number it doesn't go thru. When trying to resolve www.nufone.net I get (sourec: http://www.dnsstuff.com/tools/lookup.ch?name=nufone.nettype=A ): I received this when they had an outage on Wed: Date: Wed, 25 Jul 2007 17:35:07 -0400 [07/25/2007 04:35:07 PM CDT] From: NuFone Inc. [EMAIL PROTECTED] Subject: Hardware Failure In Washington, DC Data Center Headers: Show All Headers At 4:36PM Eastern time we experienced a hardware failure in our Washington DC Data center. As of 5:30PM Eastern, we have restored our services at limited capacity and are working to a complete restoration of services. We apologize for the outage and are currently adding additional redundancy in our network to avoid any future outages. --Shane ___ --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] RF to IP bridge
Quoting Curt Shaffer [EMAIL PROTECTED]: I wanted to see if there was anything reasonable in price out there yet that performed an RF to IP bridge via asterisk. What I mean by this is callers from PSTN can be patched to a UHF/VHF radio and vis-à-vis. I know there is an option available for the Avaya systems but its a little out of the price range Im looking for (~$200/channel). Has anyone out there found a stable way to do this? Asterisk does this quite well: wawnmnxast1*CLI core show application Rpt wawnmnxast1*CLI -= Info about application 'Rpt' =- [Synopsis] Radio Repeater/Remote Base Control System [Description] Rpt(nodename[|options]): Radio Remote Link or Remote Base Link Endpoint Process. Not specifying an option puts it in normal endpoint mode (where source IP and nodename are verified). Options are as follows: X - Normal endpoint mode WITHOUT security check. Only specify this if you have checked security already (like with an IAX2 user/password or something). Rannounce-string[|timeout[|timeout-destination]] - Amateur Radio Reverse Autopatch. Caller is put on hold, and announcement (as specified by the 'announce-string') is played on radio system. Users of radio system can access autopatch, dial specified code, and pick up call. Announce-string is list of names of recordings, or PARKED to substitute code for un-parking, or NODE to substitute node number. P - Phone Control mode. This allows a regular phone user to have full control and audio access to the radio system. For the user to have DTMF control, the 'phone_functions' parameter must be specified for the node in 'rpt.conf'. An additional function (cop,6) must be listed so that PTT control is available. D - Dumb Phone Control mode. This allows a regular phone user to have full control and audio access to the radio system. In this mode, the PTT is activated for the entire length of the call. For the user to have DTMF control (not generally recomended in this mode), the 'dphone_functions' parameter must be specified for the node in 'rpt.conf'. Otherwise no DTMF control will be available to the phone user. ___ --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] Display Caller ID of called party
Quoting Savoy, Kevin - Williston, ND [EMAIL PROTECTED]: Not sure if this can be done or not, but I can't seem to find it anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I would like to have the caller id of the person I am dialing displayed and not the number I just dialed. Is this possible? So, if extension 4023 is John Doe, and I dial 4023, my display should read John Doe and not 4023. I am using a Polycom 501 by the way in case that matters. The polycom will display the name of the person you are calling if it receives it in the remote-party-ID. This is how it works with Broadworks. Remote-Party-ID: Shane Youngsip:[EMAIL PROTECTED];user=phone;screen=yes;party=called;privacy=off;id-type=subscriber ___ --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] Open CallerID Database?
Our CNAM provider claims to have more than 196 million entries. I just don't think you could reliably maintain that in this format. Let's say I'm a CLEC and I have 40,000 numbers. I want to update that in one place (my SCP, probably). I wouldn't also want to update another database through another method. Quoting Robert Norton - SophMedia LLC [EMAIL PROTECTED]: Hey Shane, The basis of my idea was that it would be user-moderated/generated. A 'owner/operator' of a number, would submit verify their phone number, enter their caller id, and basically be done with it. The logistics of it I don't really think would be that complicated. If a listing needs to be updated they basically go through the same process. Right now, we're using a commonly available script (I can't remember the link off hand) that uses Google, 411.com, etc, to do a lookup and although it works pretty good, it is horribly inaccurate the majority of the time. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. -Original Message- From: Shane Young [mailto:[EMAIL PROTECTED] Sent: Monday, February 19, 2007 12:46 PM To: Robert Norton - SophMedia LLC Subject: Re: [asterisk-users] Open CallerID Database? Robert On the surface, I don't see how you could a db with a very good hit rate without paying for the data. There are thousands and thousdands of database updates every day. Perhaps I am missing your intent here. Quoting Robert Norton - SophMedia LLC [EMAIL PROTECTED]: Hey Guys, I'm curious if there's an interest in a free, CallerID database? For those of you in the same spot we are, our current provider only provides us with the CND, excluding CNAM. Would creating a public database, managed by users be worthwhile to anyone? Thanks - Any input is greatly appreciated. -- Robert Norton SophMedia LLC Operations Manager Cell: 480-234-4312 Office: 480-626-5449 (x300) P.O. Box 7755 Tempe, AZ 85281 http://www.XStreamHost.com - Web Hosting http://www.SophMedia.com - Consulting Web Development -- NOTICE: This e-mail (including all attachments) may contain confidential and privileged material for the sole use of the intended recipient(s). You, the recipient, are obligated to maintain it in the safe, secure, and confidential manner. Any review, use, distribution, disclosure, or copying by others is strictly prohibited. If you are not the intended recipient (or authorized to receive for the recipient), please notify the sender by reply e-mail and delete, or destroy all copies of this message immediately. --Shane ___ --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 --Shane ___ --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] 4-wire analogue interfaces?
Quoting Tony Mountifield [EMAIL PROTECTED]: Does anyone know of any 4-wire analogue interface cards that could be made to work with Asterisk? (I'm not averse to hacking channel drivers) A T1 card to a D4 bank with something like a 4WEM or 4WTO should do the trick. --Shane This message was sent using IMP, the Internet Messaging Program. ___ --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] Text to Speech
Quoting Kevin Savoy [EMAIL PROTECTED]: Can someone recommend a good text to speech engine that is usable by Asterisk? I have tried the Festival one and it just doesn't cut it for commercial applications. I like Cepstral. Using the information here: http://www.oldskoolphreak.com/tfiles/voip/installing_app_cepstral.txt You can install app_cepstral after you have installed the package and libs from Cepstral. Then you can do something like this: [macro-cepstral-demo] exten = s,1,Cepstral(voice name=DuchessHello and welcome to the world of text to speech using Cepstral. My name is Duchess./voice) exten = s,n,Cepstral(voice name=WalterHello and welcome to the world of text to speech using Cepstral. My name is Walter./voice) exten = s,n,Cepstral(voice name=ShoutyHello and welcome to the world of text to speech using Cepstral. My name is Shouty./voice) exten = s,n,Cepstral(voice name=WilliamHello and welcome to the world of text to speech using Cepstral. My name is William./voice) exten = s,n,Cepstral(voice name=WhisperyHello and welcome to the world of text to speech using Cepstral. My name is Whispery./voice) exten = s,n,Cepstral(voice name=RobinHello and welcome to the world of text to speech using Cepstral. My name is Robin./voice) exten = s,n,Cepstral(voice name=LindaHello and welcome to the world of text to speech using Cepstral. My name is Linda./voice) exten = s,n,Cepstral(voice name=EmilyHello and welcome to the world of text to speech using Cepstral. My name is Emily./voice) exten = s,n,Cepstral(voice name=DianeHello and welcome to the world of text to speech using Cepstral. My name is Diane./voice) exten = s,n,Cepstral(voice name=DavidHello and welcome to the world of text to speech using Cepstral. My name is David./voice) exten = s,n,Cepstral(voice name=DuncanHello and welcome to the world of text to speech using Cepstral. My name is Duncan./voice) exten = s,n,Cepstral(voice name=DamienHello and welcome to the world of text to speech using Cepstral. My name is Damien./voice) exten = s,n,Cepstral(voice name=CallieHello and welcome to the world of text to speech using Cepstral. My name is Callie./voice) exten = s,n,Cepstral(voice name=DogHello and welcome to the world of text to speech using Cepstral. My name is Dog./voice) exten = s,n,Cepstral(voice name=AmyHello and welcome to the world of text to speech using Cepstral. My name is Amy./voice) --Shane This message was sent using IMP, the Internet Messaging Program. ___ --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] 911 Testing
Quoting Leif Neland [EMAIL PROTECTED]: According to what I've read somewhere, at least our 911 (112) has an answering machine, saying Alarm central, one moment and a few seconds delay, before the call actually is signaled to the dispatcher, to filter out misdials and crank calls. So if you hang up quickly, they'll never know or be bothered. In Minnesota (probably most places in the US) Once you have dialed 911, even if it was in error, you should stay on the line until a dispatcher answers. If you don't they'll consider it a 911 hangup and attempt to call you back. If they can not reach you, they will dispatch a law enforcement officer (and in some areas, other emergency services). The usual call flow I've experianced is this: I Dial 911 They answer Minneapolis 911 I say This is Shane from company x making a 911 test call. They will either say ok or Please Hold if they have other calls waiting. Once they have said ok, I'll say I want to confirm you see my number as xxx-xxx- and my address is y They will almost always say Yes, that's what we have I'll say Thank you They will say Good Bye and hang up. I'll hang up. --Shane This message was sent using IMP, the Internet Messaging Program. ___ --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] Zip code, city and area codes
Quoting Ronald Wiplinger [EMAIL PROTECTED]: Is there a table available, which tells me if a zip code, city and area code matches? For now I did it with google, type each info in and found out if it matches, but it would be easier if there is a table available. If you subscribe to the LERG, you can build a table which might fit your needs. If you simply want to find a zip code for a NPA-NXX, you can lookup the switch for that NPA-NXX in table LERG6 then lookup the zipcode for the switch in table LERG (I think). This works good for finding a nearby zipcode to match a callers ANI. If you need something more than that, it will be difficult. A zip code can serve multiple NPA-NXX's and an NPA-NXX can be in multiple zip codes. --Shane This message was sent using IMP, the Internet Messaging Program. ___ --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] Alarmreceiver
Quoting Kevin Withnall [EMAIL PROTECTED]: Does someone have code to do this already ? Ie log alarm stats to a database and determine when to call out ? I have a fairly simple system setup that logs things to a database and will perform some type of action based on the account, zone, type and trip/restore status of the alarm. It's fairly ugly, but it get's the job done. Email me offline and I'll give you a tour. This message was sent using IMP, the Internet Messaging Program. ___ --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 to select Ceptral's Voice in Asterisk's Swift application??
Quoting Pimjai Wesnarat [EMAIL PROTECTED]: Hi, I'm using Cepstral as a TTS Engine for Asterisk with Swift application. It works fine when I have just 1 voice installed. Now I have 2 voices in the same language installed but I can't seem to find the way to select which voice to use in Swift's application in Asterisk. Does anyone know?? [cepstral-demo] exten = s,1,Answer exten = s,n,wait(1) exten = s,n,Cepstral(voice name=DuchessHello and welcome to the world of text to speech using Cepstral. My name is Duchess./voice) exten = s,n,Cepstral(voice name=WalterHello and welcome to the world of text to speech using Cepstral. My name is Walter./voice) exten = s,n,Cepstral(voice name=ShoutyHello and welcome to the world of text to speech using Cepstral. My name is Shouty./voice) exten = s,n,Cepstral(voice name=WilliamHello and welcome to the world of text to speech using Cepstral. My name is William./voice) exten = s,n,Cepstral(voice name=WhisperyHello and welcome to the world of text to speech using Cepstral. My name is Whispery./voice) exten = s,n,Cepstral(voice name=RobinHello and welcome to the world of text to speech using Cepstral. My name is Robin./voice) exten = s,n,Cepstral(voice name=LindaHello and welcome to the world of text to speech using Cepstral. My name is Linda./voice) exten = s,n,Cepstral(voice name=EmilyHello and welcome to the world of text to speech using Cepstral. My name is Emily./voice) exten = s,n,Cepstral(voice name=DianeHello and welcome to the world of text to speech using Cepstral. My name is Diane./voice) exten = s,n,Cepstral(voice name=DavidHello and welcome to the world of text to speech using Cepstral. My name is David./voice) exten = s,n,Cepstral(voice name=DuncanHello and welcome to the world of text to speech using Cepstral. My name is Duncan./voice) exten = s,n,Cepstral(voice name=DamienHello and welcome to the world of text to speech using Cepstral. My name is Damien./voice) exten = s,n,Cepstral(voice name=CallieHello and welcome to the world of text to speech using Cepstral. My name is Callie./voice) exten = s,n,Cepstral(voice name=DogHello and welcome to the world of text to speech using Cepstral. My name is Dog./voice) exten = s,n,Cepstral(voice name=AmyHello and welcome to the world of text to speech using Cepstral. My name is Amy./voice) This message was sent using IMP, the Internet Messaging Program. ___ --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] MINNESOTA: TwinCities Asterisk Users Group - Saturday 04/08/2006
SPONSORED THIS MONTH BY: SOUND CHOICE COMMUNICATIONS LLC Keep in touch with the World Hello, The next Asterisk Users Group meeting has been scheduled for this Saturday March 11th at 11:30am. Meetings are held monthly on the second Saturday of each month, excluding July and December. The Agenda is posted online http://www.voip-info.org/wiki/index.php?page=Twin+Cities+Asterisk+User+Group+Agenda Meetings are held at Sound Choice Communications LLC... http://maps.google.com/maps?oi=mapq=7839%2012th%20Ave%20S%2055425 Sound Choice Communications is located in Bloomington Minnesota, just 1/2 mile west of the Mall of America. The address is: 7839 12th Ave S, Bloomington Minnesota 55425. We are just south of Interstate 494 on 12th Ave. 12th Avenue is one exit west of Hwy 77 (Ceder Ave). This is the Semi-Annual New Asterisk Users meeting . If you want to learn how to install asterisk on your system, this is the meeting to attend. If you're having a problem with Asterisk, bring your questions to a meeting for free help. We love helping new users! Come to a meeting to meet other asterisk users, see asterisk solutions, win a door prize, eat food, or for the good company, to look for work, if your looking for employees, to go out for a drive, to get out of your house, whatever, JUST COME TO THE MEETING! New visitors can help themselves to FREE FXO Interface cards (So you can connect your phone line, and have a timing source for meetme and IAX protocols). Some members have been known to swap hardware at the meetings. Have extra VoIP gear, looking for VoIP gear? There's plenty of hardware to see. Have you been to a meeting recently? Please come and share your own ideas and learn from others. As always, free food. We are always looking for help with meeting topics. If you feel like taking the lead, please do and simply let me know if you need anything. Meeting starts at 11:30am and parking is available in the rear of the building. Runs about 2 hours or less, and we'll order Pizza to the meeting for lunch. Look forward to seeing you there. http://www.voip-info.org/tiki-index.php?page=Asterisk%20User%20Group%20TwinCities%20Minnesota%20USA If you have a product or service you'd like to introduce to our members, send a private message to ejo1(at)soundchoicecomm.com and we'll see if we can't get you listed as next month's sponsor. This message was sent using IMP, the Internet Messaging Program. ___ --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] Alarmreciver
Quoting Andrew Nowrot [EMAIL PROTECTED]: Hi, Did anyone try to set up alarmreceiver application over IP network? Which ATA can I use? I tried to set up it with Linksys PAP-2 but with no luck. Maybe I did something wrong with alarmreceiver.conf (I tried diverse settings, but nothing worked). Sometimes alarmreceiver is able to get some events but sometimes not. I think Linksys PAP-2 has a problem with recognizing digits in appropriate way. I've been using it for a couple of years and it works great. I've found that some SIP devices need to be set to In-Band for DTMF signalling. This will also force you to use G.711. It seems the contactID format is really picky about timing and some SIP devices seem to fiddle around with the timing when doing out of band DTMF. This message was sent using IMP, the Internet Messaging Program. ___ --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] ADT/Brinks alarm dialing through Asterisk
Quoting Chris Mason (Lists) [EMAIL PROTECTED]: Lists wrote: I am hoping the alarm companies adopt quicker to the internet. I don't see that happening. Internet reliability is not going to be sufficient for alarms. PSTN lines, for all their issues, don't fail, and alarm systems can sense the dial tone and alert if it is missing. I would not hold your breath waiting for a voip alarm protocol, but there are smtp and paging interfaces, would they suffice or are you looking for central station dialing over IP? I have seen it happen. Digital Monitoring Products http://www.dmp.com/ has an alarm panel called the XR500 which has an option for connecting to the central station over IP *and* is UL listed. It also has the ability to use a POTS line or cellular line as a backup. This message was sent using IMP, the Internet Messaging Program. ___ --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: RE: [Asterisk-Users] ADT/Brinks alarm dialing through Asterisk
Quoting [EMAIL PROTECTED]: A secondary issue may be insurance. In a domestic situation, if you receive a discount for having an alarm installed, you may find that the insurance discount is only valid if the alarm is installed over POTS, and usually by hardwiring. This is for actuarial reasons, that is to say the premiums are based on the reliability of POTS. Any rerouting of alarm over VOIP could invalidate your insurance leaving you uninsured! Generally, to receive the discount, there is some language surrounding a UL listed central station. Dialup POTS is one way alarms are connected to central stations. There are also systems which use a dedicated (non-dial) line, IP, Cellular and even a relatively new system which uses the skytel pager network to communicate. This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: how to show called name on calling polycom display
We use Polycom phones at work behind Broadworks. When we call from one phone to another phone, we see the called name on the display. This is because Broadworks sends a remote-party ID back to the calling phone when it invites the called phone. This also seems to work on the Sipura phone. So the Polycom phone already has this built in functionality. It just needs to be implemented in Asterisk. Quoting Jerry Jones [EMAIL PROTECTED]: Could always create named extensions and dial by name On Mar 15, 2006, at 11:17 AM, C F wrote: IIRC, it's something that is supported in the latest versions of SIP, which Asterisk doesn't support yet. On 3/15/06, Noah Miller [EMAIL PROTECTED] wrote: Hi Giorgio - we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to show the called name on the calling polycom display instead of his /her extensions as I do with the caller name on the called polycom. Is it possible? If yes, how? If this is possible, it would be quite complicated to do. This would take some tricky XML hacking on the Polycom side to read this info and display it on the phone's screen, and some even more clever way to send this info from the asterisk machine. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message was sent using IMP, the Internet Messaging Program. ___ --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] 7940/60 SIP 8.2
Quoting Mailing List [EMAIL PROTECTED]: I believe they've done that the entire time. I've never known them to be real supportive of competing third party solutions. They support third-party partners such as Broadsoft. This message was sent using IMP, the Internet Messaging Program. ___ --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] Alarmreceiver
Quoting andrutto [EMAIL PROTECTED]: I just want to ask if anyone has some experience with Alarmreceiver application in * 1.2? Is this application reliable (according to wiki it isn't)? I don't see anywhere in the wiki where it says this is unreliable. The wiki mentions that This application is NOT Underwriter's Laboratory (UL) approved. My experiance is that it is as reliable as anything else in Asterisk. I've been using it since August of 2004 and it's always worked fine for me. I use a DMP security system with card access. When somone opens a door to my house with their card, it reports the event to Asterisk which then annoucnes the name of the person who as come in (and what door)throughout the house. I also have a text message sent to my pager of the event. I've also used it with a GE panel to send alarms from a remote Central Office. I managed to communicate Asterisk (Alarmreceiver) with a burglar alarm, but it behaves very strange. Sometimes alarmreceiver is able to get some events but sometimes not. Can you be more specific? How is the alarm panel connected to the Asterisk system (ATA, ZAP Channel, etc) This message was sent using IMP, the Internet Messaging Program. ___ --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] Push to talk and asterisk
I have a repeater using app_rpt, it seems to work just fine. Quoting Mustafa N. Deeb [EMAIL PROTECTED]: Has anyone been able to compile app_rpt? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Saturday, August 13, 2005 4:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Push to talk and asterisk You may be duplicating work that has already been done. http://www.zapatatelephony.org/app_rpt_article.pdf Mark, KC2ENI Mustafa N. Deeb wrote: Hi We are putting some efforts on having asterisk work as a PTT server over GPRS. Anyone interested to part of it , Please email me privately Best Regards Mustafa N. Deeb ___ 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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] Panasonic KX-TD1232
I can help you I think. do you have the manuals for the Panasonic? Quoting Dan Morin [EMAIL PROTECTED]: If anyone has any experience with a Panasonic KX-TD1232 phone system, I would really like to talk to you for a few minutes. I have asterisk connected to a Panasonic system via FXS - CO ports. I'm trying to get the Panasonic configured so that if someone dials a number (9) while Intercom is selected, it will select a line in the correct trunk group (Asterisk lines, rather than PSTN lines), then the user can finish dialing the asterisk extension. Any ideas? Thanks in advance. ___ 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] CIsco 7960 SIP Image
http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/mgcp/frmwrup .pdf Quoting Preston Garrison [EMAIL PROTECTED]: www.voip-info.org has it Preston Garrison direct: 877-748-4142 fax: 310-774-3901 cell: 623-748-4140 -Original Message- From: Ryan Finnesey [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tue, 31 May 2005 11:18:47 -0400 Subject: [Asterisk-Users] CIsco 7960 SIP Image Does anyone have a document I can use as a guide on how to load a SIP image on a cisco 7960 phone? Ryan ___ 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 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 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] Any 24 (or 30) way FXS PCI cards?
Quoting Rich Adamson [EMAIL PROTECTED]: It seems to me silly to have a T1/E1 card to connect to a channel bank when you could just have a 24/30 way FXS card in the slot in the first place. Does such a thing exist? Wouldn't Digium have a lot of customers if they could produce one for say $1000 retail? Trouble is power. Unless there is more power made available, you may not be able to drive the ring voltage of several consecutive lines at once. Take for instance the Adit 600, it has a 130w power supply for just 25 ren capability. Think of the troubles that would cause trying to be regulated through your standard PC PSU of 300w. Won't you just love trying to diagnose random reboots right after a phone call comes in and over draws your PSU capacity and it goes into short protection where it begins pulsing power. The InterTel Axxess had a good solution to this. Each station card had it's own ringing generator which produced ringing voltage at about 70 volts. It worked for most things but we had problems with a few modems and double-gong ringers. If you needed more, you would move a jumper on the card which would disable the internal generator. Then, you would provide 90/20 on the 25th pair of the AMP connector. In my mind, I imagine something very similar. It would look like an old SCSI card from the back of the PC with the big 25 pair connector. Pairs 1-24 would be the station lines and pair 25 would be for the external ring generator. --Shane ___ 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] cepstral integration with * using AGI?
Quoting John Middleton [EMAIL PROTECTED]: Hi, I've looked at the Wiki for this, have seen the Swift.agi details, but has anyone got a current script for Cepstral and an example of integraton in * please? It's been a while since I've fiddled around with it, but it should work like this: exten= s,1,Answer exten= s,2,agi(swift.agi|Hello. This is shanes boat calling.) exten= s,3,agi(swift.agi|Shane will be going out on the boat soon.) exten= s,4,agi(swift.agi|Shane will be out on the lake in uproximatly 45 minutes.) exten= s,5,agi(swift.agi|If you would like to go for a ride you should be able to meet at sun sets in Wyzeta.) exten= s,6,agi(swift.agi|To hear this message again. Touch one.) ___ 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] Twin Cities Asterisk meeting still on for Saturday?
Yes. Quoting Roger Hanson [EMAIL PROTECTED]: Is the meeting still on for Saturday 1/8/05? 11:30am at 2375 University Av W STE120, Saint Paul. ___ 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 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] NPA NXX data
Quoting Jon Bebeau [EMAIL PROTECTED]: HI all - I know, slightly off list, but.. I'm looking for a NPA NXX database with City and State. The North American Numbering Plan Admistrator has some info at http://nanpa.com/nas/public/assigned_code_query_step1.do?method=resetCodeQueryModel You can download files which give you information like this: State NPA-NXX Ocn Company RateCenter Switch EffectiveDate Use AssignDate Initial/Growth File Updated 10/27/2004 IA 319-201 4822NEXTEL PARTNERS OPERATING CORP. MTPLEASANT MNPLIAXC5MD AS 03/19/2004 I IA 319-202 8577IOWA WIRELESS SERVICES, LP CEDAR RPDS CDRRIAAXCM0 AS 05/13/2004 G IA 319-208 8474MCLEODUSA TELECOMMUNICATIONS SERVICES, INC.- IA BURLINGTON DVNPIAEQDS0 AS 03/25/2003 I IA 319-209 4822NEXTEL PARTNERS OPERATING CORP. BURLINGTON DVNPIADT0MD AS 04/10/2001 I IA 319-210 8447SPRINT SPECTRUM L.P.- IA CEDAR RPDS CDRRIADT9MD AS 02/12/2002 G IA 319- 211 UA IA 319-212 6570CELLCO PARTNERSHIP DBA VERIZON WIRELESS - IA COLUMBSJCT CLJTIA01CM0 AS 04/10/2003 I IA 319-213 7229MCIMETRO, ATS, INC. CEDAR RPDS CDRRIADTDMD AS 07/14/2004 I IA 319-217 6570CELLCO PARTNERSHIP DBA VERIZON WIRELESS - IA MTPLEASANT MNPLIAXC3MD AS 02/06/2003 I ___ 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] Has anyone connected to 7960 with console cablefor setup?
If the phone has not been converted to SIP, the console may not work. I was never able to get the console to work on a skinny phone, but it does work on a SIP phone. Quoting Paul Brock [EMAIL PROTECTED]: Randy, Is it a new unit? The only reason I ask is that hitting the settings button should let you straight in. There is an Rs232 port on the bottom - however not oversure what it's used for on the 7960's. The reason I as wether it's new or not is that it might need firmware resetting as per the cisco information (not immediately to hand). If you can see the menu's and just chance change the setting, I think it's something like *# or **# to allow change. Sorry if that's suck egg territory - just trying to cover anything obvious which is easily missed!! Paul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Randy MacKay Sent: 16 December 2004 18:35 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Has anyone connected to 7960 with console cablefor setup? I have a Cisco 7960 phone. I cannot seem to use the settings button to get into the phone to change the TFTP server. I've set up a DHCP Server, TFTP Server with the same address, and the phone requests the address of 0.0.0.0, the server offers the address of 192.168.2.2, but the phone seems not to take it. I have no action on the TFTP side. So, since I can't seem to server the phone anything by TFTP, and I can't use the settings button, then I thought I might make a console cable (see below). I tried to use hyperTerminal, but got no response from the phone. Anyone have any ideas? Thanks, Randy I found a link to make a Cisco Console Cable for RJ-45. http://www.hardwarebook.net/cable/serial/ciscoconsole9.html DB9F RJ45 Receive Data 2 3 Transmit Data 3 6 Data Terminal Ready 4 7 Ground5 4 Ground5 5 Data Set Ready6 2 Request to Send 7 8 Clear to Send 8 1 The Console Access Manual, give the following cable information: Console Cable Requirements You use a serial cable with a connector to connect a PC and a phone. The cable uses an RJ-11 connector for the phone and an RJ-45 connector to an RJ-45-to-DB9 converter for the PC. Table D-1 shows the pinout requirements for the console cable. Table D-1 Console Cable Pinouts RJ-11 Connector RJ-45 Connector Pin 2 == Pin 6 Pin 3 == Pin 4 Pin 4 == Pin 3 So, I thought I would go right from DB9F to RJ-11 DB9F RJ-45 RJ-11 Pin 2 Pin 3 Pin 4 Pin 5 Pin 4 Pin 3 Pin 3 Pin 6 Pin 2 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.817 / Virus Database: 555 - Release Date: 12/15/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] XML to monitor queues on Cisco display ?
Quoting Henry Devito [EMAIL PROTECTED]: I attempted this but I got stuck on one issue. Cisco phones pull data so I couldn't get them to autoupdate. In other words push data to them. You can use an http Refresh to keep the screen updating once you've accessed your XML application. It's not the best solution, but it is a step closer. --Shane ___ 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] new ATA box for sale by Linksys
Quoting Matthew Boehm [EMAIL PROTECTED]: Does anyone have one of these models? Can they confirm that it works with any other SIP server? How is the PAP2-NA configured? Web? Phone? The pdf I downloaded from the pap2-na page on shopblt.com says Model: PAP2. The product manager for this devices sent us one and the first thing I did was configure it for my home Asterisk box. It works just as an SPA2000 would. The voice prompts are the same (except no mention of the word Sipura). The web interface looks like it has a different style sheet. ___ 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] Cepstral
Quoting Jerry Geis [EMAIL PROTECTED]: Cepstral offers Linux versions. Just contact them. http://www.cepstral.com/cgi-bin/downloads?page=voices Note that you can not download any Linux versions from that page. They changed something a while back. Released a new TTS engine for Windows and Windows CE, but have not as of yet released it for Linux. I have an old version of the program called theta and I have the Frank voice which works well. ___ 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] Zap channels stuck in 'Rsrvd' state
Good Evening I found your post about this problem. Did you ever find a fix for it? I'm experiancing the same problem. Thanks. Quoting Steve Creel [EMAIL PROTECTED]: I have two Adtran 750's connecting our analog phones to asterisk. On occasion, I get a channel that gets stuck off hook. 'show channels' shows: Zap/27-1 (longdistance s 1 ) Rsrvd (None) (None) And will just stay like that until the phone is manually picked up and hung up again (or asterisk is stopped/started). I guess this is a function of an unclean hangup (being read as a flash instead of a hangup?). A 'soft hangup zap/27-1' will not do anything (though it makes an attempt). Does shortening the rxflash time sound like it may help this? (Does anyone have a good explanation, or link to one, of the prewink, wink, preflash, flash, start, rxwink, rxflash, debounce timing functions?) Thanks, as always... Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lots of FXS ports / Channel Bank ?
Quoting Joel Vandal [EMAIL PROTECTED]: Hi, I have a client that have currently 400 analog phones (all wired w/ Cat3). I need multi-ports FXS interfaces but I only find 24 ports FXS (like Mediatrix 1124) but it's a little bit expensive to get 15-16 box (~408 FXS ports). You can get 40 stations out of an Adit-600 using MGCP as long as you don't need to use G.729. I beleive it only supports 24 g.279 calls at one time. Two of these mount side-by-side in a rack using two rack-units, giving you 80 ports in the space many other devices will give you 48 ports. Sadly, It seems that rack and stack is the cheapest way to go. ___ 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] 7960 Dynamic DNS?
Quoting Marty Mastera [EMAIL PROTECTED]: Hello everyone Searching the archives and google always comes up with entries regarding the dyn dns option in the 7960, but I can't find answers to my specific question It's a way to specify a DNS via config file which has priority over whatever is handed out from DHCP. (Optional) IP address of a new dynamic DNS server. If a new DNS server address is specified, it is used for any further DNS requests after the phone uses the initial DNS address upon bootup. The DNS addresses are used in the following order: 1. dyn_dns_addr_1 (if present) 2. dyn_dns_addr_2 (if present) 3. DNS Server 1 4. DNS Server 2 5. DNS Server 3 6. DNS Server 4 7. DNS Server 5 The dynamic DNS address is not stored in flash memory. Only dotted IP addresses are accepted. This value can be cleared by removing it from the configuration file or by changing its value to a null value or to UNPROVISIONED. Note The dynamic DNS address is not stored in flash memory. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940 Phones as paging system?
Quoting Rich Adamson [EMAIL PROTECTED]: The cisco v6.x sip releases also include the ability to auto-answer a call (required for phone paging), however some folks tend to suggest that is a security problem as anyone can call that autoanswer extn number and listen in on whatever is going on around the phone. There is no beep or other indication the phone/microphone is open. We are on Cisco's beta program where we get to try out the new sip software before it's released, report bugs, suggestions, etc. This was one of the things I pointed out to them. I suggested adding a beep or something just before it answers. At the very least, they could make the icon of that line different than the regular idle icon. They didn't add the beep, but they did add the icon so when the phone is idle, you can see that it could auto-answer. I'll try suggesting it again for 7.X of the SIP image. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 vs 7905
Quoting Eric Wieling [EMAIL PROTECTED]: Anything that says CallManager is NOT SIP. You want SIP. These are the the part numbers. Your pricing will vary. ++--+--+ | part | description | list | ++--+--+ | CP-7960G | Cisco IP Phone 7960G, Global | 415 | | CP-7940G | Cisco IP Phone 7940G, Global | 315 | | CP-7912G | Cisco IP Phone 7912G | 245 | | CP-7905G | Cisco IP Phone 7905G, Global | 165 | | CP-LCKNGWALLMOUNT= | Locking Wallmount Kit for the 7910, 7940, 7960 IP Phones | 31 | | CP-WALLMOUNTKIT= | Non-Locking Wall Mount Kit for 7910, 7940, 7960 IP Phones| 26 | | SW-SMH-UL-7912 | SIP license for single 7912 IP phone | 80 | | SW-SMH-UL-7905 | SIP or H.323 license for single 7905 IP phone| 80 | | SW-SM-UL-7960 | SIP and MGCP license for single 7960 IP phone| 150 | | SW-SM-UL-7940 | SIP and MGCP license for single 7940 IP phone| 150 | ++--+--+ ___ 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
