Re: [Asterisk-Users] where is voice conduits
ross jones wrote: Does any one know what happened with voice conduits? I have been trying to reach them for nearly three weeks now. Their voice mail boxes are full and writing email to them does not get any returns. Thoughts or sightings are appreciated. There was a thread a month or two ago on here about voiceconduits. The general gist was they are not yet open for public business. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Monitoring calls through a transfer
Asterisk wrote: We have the following scenario: Incoming call to a queue, Agent A answers. Agent A determines after about 20 seconds that agent B needs to deal with this call. A puts call on hold, calls and speaks to B, and then transfers the call to B. B speaks to the incomming caller for 5 minutes. That's all fine. However, the CDR records the call as incomming to agent A for 5 minutes, and the agent monitoring recording is also determined as belonging to A. Trouble is that we need to find all calls that B received (both directly and through a transfer) and look at them. How can we do this ? How are you performing the transfer? Have you tried the following? show application ResetCDR -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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 with @home
Kurt Fankhauser wrote: just reinstalled @home and i have a one of those 100 cards, anyways when i call from the pstn the box picks up but i hear nothing, then it clicks a couple times, then nothing again, i am trying to get the digital receptionist to work but it won't save my wav file to the @home box and all the radio buttons under incoming calls are greyed out. the greyed out thing seems to be my biggest problem right now, also do you have to use a ip phone to record your greeting because this wav file stuff isn't working. Are you logged into the console while your testing the dialing in? What messages are you seeing? If asterisk is already running in the background, do a asterisk -r before you start to dial in. If there is some other interface in the @home distribution for monitoring asterisk, you'll have to say what app you're using and what you're seeing. At any rate, without log, error, or console messages there's not alot we can do for you. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Trying to install X100p
Mike Wright wrote: Hmm - I actually installed asterisk FIRST - was playing with it then I decided I wanted to try the X100p. So I got the card, installed zaptel and libpri. SO Do I now have to go back and rebuild asterisk? If you pulled all three from a cvs, then you should pull a new copy of asterisk just to be in sync with the libraries. (then rebuild) -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Anyone having trouble with VoicePulse Connect?
beonice wrote: But we still have the issue of what happens when calls come in from DIDs in other countries. How are our colleagues in Europe and Asia handling this? Are you all creating handlers that special-case your incoming DID pattern and then map it to the handler for 's' as Robert demonstrated above? I guess the fundamental question is why is a call coming in from a DID any different? And, of course, does a call coming in _not_ from a DID (maybe via an SIP device? I don't know what the options are!) get automagically handled by the 's' handler without special mappings? I think you've confused your DID with inbound callerid. Unless you have a international DID, the exten=_NXXNXX pattern should always accept any call bound for you from that context. Even if someone calls you from an international location, voicepulse *should* always present a unique DID to you the same way every time. Now, if you want to do processing of an inbound call diferently based on it's origination number(it's callerid), you handle that afterwards. exten = _NXXNXX/NXXNXX,1,Goto(fromUS,1) exten = _NXXNXX/XX.,1,(Goto(fromSomewhereElse,1) ; see note! Note: I'm guessing on the pattern matching for the international number. I've not had to handle this yet, so I am just guessing based on what I've read. You would also probably want to deal with numbers that didn't present a callerid. Check the wiki for examples, I don't know the syntax. (search for ex-girlfriend blocking) HTH! -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] need info
Michael Di Martino wrote: What is the unsubscribed address? You can't unsubscribe, you're here for life. Welcome to our family. We'll be stopping by to see you from time to time and invading your house whenever we feel like it. Or, you could read very carefully this message from beginning to end. Not found it yet? Read it again. All of it. Still looking? OK, I'll give you a hint, all that stuff below my signature. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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 asterisk startup errors
Edward Banfa wrote: [EMAIL PROTECTED] asterisk-1.0.5]# ./asterisk -vvvc == Parsing '/etc/asterisk/asterisk.conf': Not found (No such file or directory) == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) It looks like you haven't created any of the config files. Is there a folder named /etc/asterisk? If not, make one(I'm not sure if the next command would do it itself), and then from your asterisk source directory, do: make samples That should get rid of most of the errors. From there you should read everything you can on these sites: http://www.voip-info.org/ http://www.asteriskdocs.org/ -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] VoIP Service Provider
William Cruz wrote: Hi everyone in the asterisk community. Am new to asterisk, while doing the installation I notice that sip.conf examples were not clear for beginners like me so I would like to share my current working configuration with everyone. Swifttel.net is a new VoIP service Provider out of Georgia. Their web site is www.swifttel.net. Currently we have service with them and it has been a pleasant experience. The asterisk SIP setup is very simple and it works great. Included is my current SIP configuration with this service provider. Am using the Grand streams AT286 with the Aastra 390 phone.Running on FC3. no problems so far. I can make out side calls and receive calls from any where. No delay is experience and the call quality is great. If you're going to promote your own company, that is fine, but you should do it in the -biz list. If you're going to try hard to sound like you are an objective third party, next time try to remember to set your email address properly. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Sending DTMF after a call is set up
Bill Hamlin wrote: I'm using Dial to place a call to a PBX. But then I want to wait a few seconds and dial an extension. Dial doesn't return until the call is disconnected though. Try this posting: http://www.voip-info.org/wiki-Asterisk+cmd+dial?page=Asterisk%20cmd%20dialcomments_threshold=0comments_offset=0comments_sort_mode=commentDate_desccomments_maxComments=10comments_parentId=931#threadId1168 It might be channel specific, I do not know. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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 functions without voIP
Pablo Fernandes wrote: Can i use the Asterisk functions (call recognition for example), using conventional telephony (in Brazil) ? Generally, yes. (VOIP is just a cool thing to be into these days.) Can you define call recognition for me? Do you mean CallerID(determining the phone number that is calling you)? You will need hardware that is compatible with your areas telephone network. (Stating that as it is likely different from the US network.) If digital voice circuits(in any form) are available in your area, you'll likely be happier using them than POTS lines. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem : undefined symbol.
Kim Daeyong wrote: I downloaded asterisk to use cvs to checkout the release version. After installing, I would like to load module chan_h323.so but there is some error : *CLI load chan_h323.so Feb 17 15:22:38 WARNING[2865]: loader.c:258 ast_load_resource: /usr/lib/asterisk/m odules/chan_h323.so: undefined symbol: __use_ast_pthread_create_instead__ Unable to load module chan_h323.so *CLI How can I solve that problem? Exactly which version did you download? (What did you type into your CVS statement?) If asterisk compiled and is runnable other than this error, just log in with asterisk -r and give us the Connected to... line. Using make, there is an option to do a make update, which should download any changes that were tagged to the version of CVS you downloaded. If the problem has been fixed since you downloaded originally, issuing a make update should help. I don't think update recompiles anything, so you will probably have to make install again. I am also not sure if you need to a make clean or anything like that. If that doesn't fix your problem, look in mantis on bugs.digium.com and see if anyone else has reported it. For other readers: I've not had an error like this, so I'm not sure exactly what the protocol is. Should the original poster post next to asterisk-dev, or to mantis? -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Having trouble with extensions in an include file and retrieve_extensions_from_mysql.pl
beonice wrote: The resulting extensions_from_mysql.conf file looks something like this: [vp_context] exten = 1000,1,Record(/tmp/rec:gsm); exten = 1000,2,Playback(/tmp/rec) ; exten = 1000,3,Background(goodbye) ; exten = 1000,4,Hangup(); I decided to #include this in my main extensions.conf, like so: [main_vp_context] exten = s,1,Answer #include extensions_from_mysql.conf exten = #,1,Background(goodbye) ; Notify caller exten = #,2,Hangup() ; Hang up exten = t,1,Hangup() ; Hang up if timeout exten = i,1,Playback(invalid) ; Play invalid ; extension if caller ; misdials an extension snip Anyone see a possible reason for the problem? Do you have any ideas how to use an include file which contains multiple contexts? Or will I have to generate multiple include files, one per included context, without the context lines in these files? The only thing that seems out of place to me is your #include in [main_vp_context]. It looks to me like you intend for the s, #, t, and i extensions to be in [main_vp_context]. The way you layed out this example, that's not what is happenning. I think you wanted this: Your extensions_from_mysql.conf should still look like: [vp_context] exten = 1000,1,Record(/tmp/rec:gsm); exten = 1000,2,Playback(/tmp/rec) ; exten = 1000,3,Background(goodbye) ; exten = 1000,4,Hangup(); Then, in extensions.conf: #include extensions_from_mysql.conf [main_vp_context] exten = s,1,Answer exten = #,1,Background(goodbye) ; Notify caller exten = #,2,Hangup() ; Hang up exten = t,1,Hangup() ; Hang up if timeout exten = i,1,Playback(invalid) ; Play invalid ; extension if caller ; misdials an extension include = vp_context This way, you define both contexts, and include the extensions that were defined in [vp_context] into [main_vp_context]. I don't know if this will resolve your other problem, but I believe this is the dialplan you were trying to build. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] The 'sipfriends' table is obsolete - ????
[EMAIL PROTECTED] wrote: IS Anything changed?? Missed something? You're running head and not watching -dev? How should the iaxpeers and sippeers tables look like then? This message was posted to asterisk-dev recently: http://lists.digium.com/pipermail/asterisk-dev/2005-February/009445.html -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Having trouble with extensions in an include file and retrieve_extensions_from_mysql.pl
beonice wrote: Yes, I see what you are saying. This sounds backwards, but it's actually doing what I _want_ it to do. :) From what I see in the dialplan, what asterisk does is, it loads the handlers for '#', 't' and 'i' as part of vp_context, not as part of main_vp_context. That actually happens to be as I wanted it. main_vp_context is simply a place-holder for when I am testing without the include file, and in those cases, I simply comment out my include file and voila, those handlers now handle the main_vp_context incoming cases. I know, I'm weird. :) Not necessarily... I'm thinking other words... ;) Back to your original post... As of yesterday, though, when I have this format, asterisk won't accept incoming calls. It barfs with the message: Feb 16 21:53:14 NOTICE[4330]: chan_iax2.c:5757 socket_read: Rejected connect attempt from 66.234.228.170, request '[EMAIL PROTECTED]' does not exist So, where is this voicepulse_connect_context context? -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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 to dial extension automatically
Oswaldo Arratia wrote: Has anyone figured out how to make a Sipura to dial an extension automatically as soon as you pick the the handset? Go to google and type: sipura hotline Read the first three links. Test. Send us a note telling what worked for you. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Anyone having trouble with VoicePulse Connect?
beonice wrote: The culprit? Me. I'd commented out the line: exten = _NXXNXX,1,Background(welcome) ; which is apparently a critical one. I was under the impression that exten = s,1,Answer Will s be traveled if a call arrives at it with a DID? The pattern you have above matches any US did that arrives into that context. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Dialplan + Registrar DB
Matt Riddell wrote: mohammad wrote: As you probably know, SER style of handling an incoming call is : 1) try to look-up it from registrar DB 2) if not found there, try to do some thing else Is there any possibility of doing the above at Asterisk Dial-plan? Just forward the call to Asterisk if it has a certain URI. I.E. sip address starts with 7,8 or 9 then send to Asterisk. Then you can do whatever you like with the call in Asterisk. I.E. I have features on 7 (i.e. 700=voicemail), iax extensions etc on 8, and 9 for outgoing calls. Is that what you were looking for? If not, can you explain this registrar db concept you're talking about? -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Sixtel.net / IAX.CC - Vanity Toll-Free Number
Rob Risner wrote: I'm just wondering, how long should a vanity number transfer really take? Were you requesting a new vanity number, or a transfer of an existing number? If it's new, have you checked to see if the number is still listed as available? google for vanity toll free number search -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Sixtel.net / IAX.CC - Vanity Toll-Free Number
BJ Weschke wrote: I've had the same experience. I've been waiting 7+ business days for their unlimited incoming minutes DIDs which were supposed to be provisioned within 1-4 hours. Did you get any notice from them on the DID? The dropdown for unlimited use DIDs only gives a choice for Area Code. Have you had any communication with them on the actual prefix you wanted? After clicking the button myself, I eventually found out that they couldn't give me a DID that was local to me. I had one other ticket open that now seems to be MIA. I'll not be using them for anything real important. There doesn't seem to be a business in the market that one could rely on. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] IAX2 bugs...
Mohit Muthanna wrote: 2) Placed a phone call. Pause. Busy tone. Asterisk never gets the call. iax2 show registry shows the connection (with the service provider) as Registered. Both times, restarting Asterisk has solved the problem. Of course, I'm not happy with this solution as I'm trying to provide a 24hr service here. Could this be a service provider problem? Mohit. I experienced something similar to this a week or so ago with an IAX provider. Nothing appeared in the console when a call was trying to dial in. I regret that I did not do enough research to see what exactly happened, but stopping and restarting asterisk resolved it immediately. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] MarkK: Auto Announce - Not auto answer
Mark Kidd wrote: if i pick my line up and the system plays back my voice saying hi how may i help you automaticaly. You're looking for something along the lines of an immediate or hotline mode. Basically, whether or not you can do that depends on the equipment you are using. I believe Zaptel devices can be set in immediate mode in your zaptel.conf file. For IP hardphones, look through their dialplans for a hotline mode. For example, I am fairly certain that my SPA2000 could be configured this way. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Strange error in debug file
Asterisk wrote: Has anyone seen this before ? Feb 15 18:37:34 DEBUG[866]: That's odd... Got a response on a call we dont know about. I'm guessing that happens when asterisk has hung up on some device but that device hasn't figured it out yet(therefore it's still trying to talk back to asterisk). Do you know if any calls had recently completed? It would be nice if the debug info gave more information about what device sent us this phantom data. I've got a whole load of them (328 in the last 5 minutes ...) I have seen this message, when dealing with my SPA2000. I am still testing VOIP providers for home use and have not turned my box up live, so I don't monitor the logs unless I am hacking on it and see something interesting go by. You could crank your verbosity up (3 minimum), turn on SIP and/or IAX debugs and wait a little more prepared to see if it happens again. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Native vs Intl calls
mohammad wrote: Is there any way that I can tell asterisk: 1) lookup among registered numbers 2) if not found in registrar , make an international call Review: http://www.voip-info.org/wiki-Asterisk+config+extensions.conf http://www.voip-info.org/wiki-Asterisk+dial+plan+-+working+example Since international numbers vary in length, you'll want to build a pattern that will match a rule(that you'll specify) for dialing international numbers of any length. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax.cc and/or Sixtel.net ,, IS IT A SCAM???
Andrejus Stavickis wrote: I'm in the same boat with DID! And even worse, I don't think there will be a way to recover the money you paid for the service you never received. I'm thinking to take legal actions on them. You paid with a credit card, didn't you? Give your credit card company a call and have them lean on sixtel. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: How do I match a D? (Was: RE: [Asterisk-Users] In-band disconn ect problem (legacy PBX) - asterisk doesn't hear the touchtone?)
Florian Overkamp wrote: Hi, On Wed, 2005-02-09 at 09:53 -0700, Kevin P. Fleming wrote: They don't. Most phones (99.9%) don't have any way to generate DTMF A through D. There are test sets that do, and of course softphones could easily do so. These tones could also be generated by automated applications, although I don't know why one of them would be talking to Background() on an Asterisk server... Actually, A-D digits would be excellent to use instead of # and whatever for transfer. Having a separate code to do '#'-transfer without having to use any regular key on the phone would take away my objections to the entire process of '#'-transfers :) I just have one problem with that. I have _never_ (that I know of) seen a phone with A-D on it! -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] looking for responsible iax provider, aftermath
Greetings, I'd like to thank everyone that has responded to my original email. I have received information from several companies, and will be testing several of them. I also would like to update a statement from my original message to clarify it: My strikelist: nufone, voicepulse, iax/sixtel The strikelist is just a list of carriers that didn't meet the needs a resonable person would ask of them. nufone - no local dids, horrible support, tollfree did's didn't work * voicepulse - no local did(for me), no tollfree dids iax/sixtel - no local did(for me), slow support * Nufone actually took their contact number off their website for a while, I guess hoping I wouldn't be able to contact them. They never once responded to any of the telephone calls I made to them. I just signed in and see that they didn't ever refund my payment and close my account, either. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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 for responsible iax provider, aftermath
Andrew Thompson wrote: Greetings, Sorry, wrong list :-D -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] More complicated huntgroups / delayed ringing
Stefan Gofferje wrote: [private_huntgroup_day] exten = s,1,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED],15,rt) exten = s,2,Wait(1) exten = s,3,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED],20,rt) exten = s,4,Voicemail(u810920) exten = s,5,Hangup exten = s,104,Voicemail(b810920) exten = s,105,Hangup I know this is OT from your posting, but I'm curious... Do the extensions in your internal context have voicemail failovers attached to them? How do you keep some random voicemail from picking up instead of falling down to your 810920 voicemail? -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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 for FXS device - CISCO ATA 186
Mike Wright wrote: I was looking for something to connect a couple of POTS handsets to my asterisk server and found this on ebay http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItemcategory=162item=5162868118 rd=1 The documentation says that it does SIP - therefore will it work in an asterisk environment. Yes, the ATA186 _will_ do SIP, but it may not with the firmware that's on it. (No, I didn't look, and couldn't tell you yea/nay if I did.) You have to get a license from CISCO for the SIP firmware. Depending on what country you're in, and the alignment of the stars, that can be a quick and painless, or an insurmountable task. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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 someone tell me why I'm getting these? (mailing list probe message)
Twice in the last week or so, I've received a message similar to the attached. A portion of the attachment that's attached is not in English. Is this my mail server failing, or someones who's on the list? -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ---BeginMessage--- This is a probe message. You can ignore this message. The Asterisk-Users mailing list has received a number of bounces from you, indicating that there may be a problem delivering messages to [EMAIL PROTECTED] A bounce sample is attached below. Please examine this message to make sure there are no problems with your email address. You may want to check with your mail administrator for more help. If you are reading this, you don't need to do anything to remain an enabled member of the mailing list. If this message had bounced, you would not be reading it, and your membership would have been disabled. Normally when you are disabled, you receive occasional messages asking you to re-enable your subscription. You can also visit your membership page at http://lists.digium.com/mailman/options/asterisk-users/asteriskuser%40aktzero.com On your membership page, you can change various delivery options such as your email address and whether you get digests or not. As a reminder, your membership password is asteriskuser If you have any questions or problems, you can contact the list owner at [EMAIL PROTECTED] ---BeginMessage--- This is the machine generated message from mail service. Unfortunately, we were not able to deliver your message to the following address(es): mail-. , : [EMAIL PROTECTED]: Unable to open maildir. --- Below the next line is a header of the message. --- . Return-Path: [EMAIL PROTECTED] Received: (qmail 26594 invoked from network); 8 Feb 2005 18:37:47 - Received: from digium-69-16-138-164.phx1.puregig.net (HELO lists.digium.com) (69.16.138.164) by grif.newmail.ru with SMTP; 8 Feb 2005 18:37:47 - Received: from [69.16.138.164] (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id 6815F2FE366; Tue, 8 Feb 2005 12:34:42 -0600 (CST) X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com Received: from psmtp.com (exprod5mx123.postini.com [64.18.0.37]) by lists.digium.com (Postfix) with SMTP id 579902FC423 for asterisk-users@lists.digium.com; Tue, 8 Feb 2005 12:34:37 -0600 (CST) Received: from source ([209.125.153.138]) by exprod5mx123.postini.com ([64.18.4.10]) with SMTP; Tue, 08 Feb 2005 12:34:42 CST Received: from [192.168.0.38] [192.168.0.38] by netresultsinc.com with ESMTP (SMTPD32-8.05) id A63FB9007E; Tue, 08 Feb 2005 13:34:39 -0500 Message-ID: [EMAIL PROTECTED] Date: Tue, 08 Feb 2005 13:32:09 -0500 From: Andrew Thompson [EMAIL PROTECTED] User-Agent: Mozilla Thunderbird 1.0 (Windows/20041206) X-Accept-Language: en-us, en MIME-Version: 1.0 To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] More complicated huntgroups / delayed ringing References: [EMAIL PROTECTED] [EMAIL PROTECTED] In-Reply-To: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-15; format=flowed Content-Transfer-Encoding: 7bit X-pstn-levels: (S:99.9/99.9 ) X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from [EMAIL PROTECTED] [65/3] Cc: X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com List-Id: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users, mailto:[EMAIL PROTECTED] Sender: [EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] ---End Message--- ---End Message--- ___ 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] More complicated huntgroups / delayed ringing
Stefan Gofferje wrote: Andrew Thompson schrieb: Stefan Gofferje wrote: [private_huntgroup_day] exten = s,1,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED],15,rt) exten = s,2,Wait(1) exten = s,3,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED],20,rt) exten = s,4,Voicemail(u810920) exten = s,5,Hangup exten = s,104,Voicemail(b810920) exten = s,105,Hangup I'll be playing around with Local/ some in the next few days(now that I more understand what it's for). I had a thought about your problem. Given the dialplan above, have you tried adding a Wait(15) to extension [EMAIL PROTECTED], so it doesn't start processing right away? You could encapsulate it using something like: [delayedinternal] exten = _.,1,Wait(15) exten = _.,2,Dial(Local/fill with appropriate variable@internal,20,rt) Note: I haven't tried this, and it might be utter malarky, but it seems logical, to me anyway. Also, please excuse any linebreaking that may have occured in my example. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] More complicated huntgroups / delayed ringing - SOLVED
Stefan Gofferje wrote: That's brilliant! And so easy... Works exactly as supposed. You should put it onto the wiki under section tips tricks. Regards, Stefan PS: Chris, your boss might like this also! Wow, is it too late to put in for royalties? ;) I'm glad it worked for you. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] breaking friends into users peers
I am about to start a program that will be generaging sip device configurations for sip.conf. My current sip.conf contains friend entries for each SIP device connected to asterisk. Should I even be attempting to split these in to seperate user/peer devices? Is there(should there be) a convention for referring to the two distinct modes of a device? Can two entries with the same name exist at the same time, if one is a friend and the other is a user? Mainly, I'm asking, are the results here undefined, or is this a recognized use? -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Can someone tell me why I'm getting these? (mailing list probe message)
Adrian Chapman wrote: It's bizarre, isn't it? Andrew sends mail to the list which sends to Boris whose mail server returns it to the list manager, which interprets it as... a problem of Andrew's - and sends Andrew his password automagically in the bounce so anybody in the middle can frig around? Not great, Digium, not great at all. I would think that mailman, having been around quite a while would be intelligent enough to figure out what happened. Following that line of thought, it would be logical to conclude it is a configuration error. Could it have to do with the ReplyTo munging that is done on lists.digium.com lists? -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Can someone tell me why I'm gettingthese? (mailing list probe message)
jbebeau wrote: OK - I should know this... How does someone call in and pick up there messages remotely? Jon Hint: Post a new thread properly, and those people who had stopped reading this thread might see it. Hint2: GIYF, asterisk check voicemail -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] no sound playing vm greetings and options
Asterisk wrote: Hi all, 2 days ago, managed to install rhat and asterisk, starting with 2 sip phones, simple config. CLI reports : playing 'vm-theperson' (language 'no')when transferred to voicemail after timeout, or playing 'vm-password' (language 'no') when dialing into voicemail extension. But, no sound, nothing heard in phones. First time, first problem, what am I missing, please help Have you loaded zaptel ? (Requires digium hardware) Checking/leaving voicemail doesn't require zaptel. MeetMe requires a timing device(zaprtc/ztdummy/zaptel). You have a SIP problem. Please post us your [general] section and the section for one of your sip phones. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] OT: How to own a telephone number?
David Brodbeck wrote: Is providing the ability to assign numbers to people instead of to locations really that hard? Is it really so much easier for Internet domains to do it? Or is this just an oligarchy at work? :) A phone number is more analogous to an IP address than a domain name. If you move, you'll have a different ISP, and you won't get to keep your old IP address. For most end users, this is a correct statement. Hosting companies, and other businesses with significant Internet presences can request a block of IPs directly from ARIN or their regional equivalent. Once assigned, the company can buy Internet access from any and multiple carriers who all point the inbound traffic to the companies assigned IPs. Among other things, this allows for semi-intelligent recovery if a carrier's network goes offline, by routing packets through another carrier's network. snipBilling is based on this, too. If people could move numbers around willy-nilly, you'd never know if you were making a long-distance call or not. This is very true, of our current area coding schema. I don't know how it could be made fair without perhaps a nationwide maximum charge for dialing 700 numbers or something like that. But it still ought to be a local call(very, very cheap or free) next door to my neighbors. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] New Asterisk user with a goal
Ryan Coates wrote: at present I am trying to test with some software phones and an asterisk box on a virtual network (no nat, no firewalls) to see if we can get anything working (client 1 calling client 2), before we splash out a fair ammount on some decent IP Phones, but am not having much joy if anyone could give me some help/advice on the matter I would be greatful if you need any more details please do not hesitate to ask, ill try to answer whatever I can Why don't you tell us exactly what you're trying, and what is and is not working? Are you getting error messages, or does your machine fall over and vomit all over itself? -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call forwarding
Adam Robins wrote: Works beautifully! Thanks. Could you post here, or to the wiki, or just back to myself the configuration you're using to implement this? Thanks. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] External Callforward (Vanity CLI)
magnus wrote: Hello all, We have been asked if we can forward (for vanity reasons) one number to another number whilst retaining the original callers Caller ID. For example caller (ie 02027 xxx ) comes in on ISDN pri, and is then auto forwarded to 0208 xxx can the original caller id e.g. 02027 xxx be presented to remote site? We have tried a variety of options, but can not achieve this, checked the wiki, but we are arriving at the conclusion that this is not possible unless the carrier allows complete control on setting CLI, currently they only seem to allow the CLI to be set as one of the DDI number on the PRI. Yet this can be done when you divert on a GSM handset and if I remember correctly on my old office definity pabx. Have I missed something? Can asterisk send Qsig call divert information? Thanks for any and all thoughts - Magnus Pick up an account with one of the many SIP/IAX providers that are asterisk compatible. They generally allow setting of the outbound CallerID. I have personally tested that this feature works on connect.voicepulse.com. I just tested iax.cc/sixtel a moment a go and it worked there as well. Before you Dial, do a SetCallerID(). -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] callback on busy
Bartosz Jozwiak wrote: Hello everybody, I would like to implement callback function. When I call a person and his extension is busy I can press, for example, 5 and get a callback when his phone is not busy anymore. When I create a call file and copy it to spool call folder asterisk makes a call. One problem is that when extension is still busy my phone rings and I get busy tone of the person who I am trying to reach. Is there another way to do it ? Or do I need to check first if channel is free or still busy ? Can anybody give me some hints ? Are you passing the Dial line in your .call file? Try building a context that has your logic in it, and directing the call file to it, after setting some appropriate parameters. You should be able to discern from Dial() why you can't get ahold of the other party. Once you do, test for that case as a part of your logic. If the case still exists, write a new .call file and exit. If the case no longer exists, connect to the original caller and be done. (Assuming you are not on the phone now, as well!) Seems like I read that you could date a .call file a little bit of time in the future and asterisk would wait to run it until then. This would be a way for you to buy some time between tries. If I'm just making this up, use cron to straighten it out. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call forwarding
Ryan Courtnage wrote: One question - let's say someone specifies their home phone number and their cell number. How do you take into the account if the cell VM picks up (ie. if cell is out of coverage and VM greeting is played)? AFAIK, there isn't much you can do in this scenario - other than ringing your house for a few rings before ringing your house AND the cell. Even then, the cell provider's 'out of the service area' message would answer the call. That's where you move on to building/adding the logic for: Press 1 to accept this call -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] toll-free anonymous
Hi, I'm Andrew. (Hi Andrew) I'm a toll-free number junkie. I've had an account with iax.cc/sixtel for about a week, and every few days, I find myself sitting at the DID menu clicking the link that reads Click here to get a random toll free number. I have three toll-free numbers now, and I don't know if it will stop... Is there any hope for me? -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Individual contexts pending on Caller-ID?
Daniel Nyström wrote: Hi! Is it possible to handle incoming calls with different contexts pending on the callerid ? E.g. like you are able to define different contexts on each Zap-channel. Just dump all the calls to a sorter context, and build your rules there. Either type in all the relavent telephone numbers, or use a database lookup tool. The last command ran here would be: Goto(VARIABLE_HOLDING_CONTEXT_NAME, 1) -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] OT: How to own a telephone number?
[EMAIL PROTECTED] wrote: Also, we're currently looking into toll-free service, but the alternatives seem to be much the same. At least nobody is telling us if there is a way to lock in a certain number even if we change providers. They've all told us that the number we receive is theirs, and if we change providers we lose the number. I'm sure 1-800-Flowers, et. al. are not being held hostage like that... They're not. Their number belongs to them, and is serviced by some LD carrier. If you call up a traditional carrier, and ask for a toll-free number, they will assign you one, and it _will_ be yours, and be portable(search for resporg). What you are seeing with these bargain providers is they have a clause in their contract that says they own the number, not you. It is a lock, and it ought to be illegal, but sadly, it's probably not. If you choose one of these companies that doesn't allow you to port or resporg your number out, that's your decision. Just ask when you get the toll-free if they do allow resporg's out, and have them show you the wording in their contract that confirms it. I would love to know what ideas you might have for getting a telephone number with the ability to stay with us even as the underlying infrastructure changes. Is this even possible? A normal (not tollfree) number, if assigned to you by a RBOC, or most CLECs belongs to you, and you can port it to any other carrier who services your area(assuming they allow port-in's). I doubt you'll find a LEC that will want to do you any better than what you've already seen with the call-forwarding, unless you have a significant amount of traffic and want to set up a point-to-point, frame, or other method of trunking the traffic. A company I used to work for advertised in newspapers and yellowpages in hundreds of cities across the country. In most areas, they had a Remote Call Forwarding(RCF) that they advertised locally with, pointed to their toll-free number. I remember looking over some of their phone bills, but I can't recall if I saw usage charges on the RCFs. Hope this helps. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] OT: How to own a telephone number?
[EMAIL PROTECTED] wrote: OK, then. If a $30/month for a virtual circuit forwarded is as good as it gets, then that pays for 600 minutes of toll-free number time at $0.05/minute. On top of the fact that we would like a toll-free number anyway, it looks like there is almost no reason to keep a permanent local number. We'll just have a permanent toll-free number instead. Is providing the ability to assign numbers to people instead of to locations really that hard? Is it really so much easier for Internet domains to do it? Or is this just an oligarchy at work? :) I don't think the general populace is ready for that. I know that some could be, knowing there are like 150+ million cell users in the US, but it's just too much for some people to process. Plus, it would screw with all the billing info and rate plans that are already in existance, contract rates, etc. I tried convincing my wife that we could cut the cord and go wireless using our cellphones mainly. I haven't been able to find a provider that will offer me a DID into asterisk in Rockingham, NC for a decent rate. I mentioned that I've got a toll-free DID set up already, and that it was really cheep, she just kind of shrugged, said she didn't think people we knew would call it. PS: If you can offer me a DID off of Level3's 910-817 block for a reasonable price, contact me offlist. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Record inbound and outbound calls to and from one number.
Tim Mattison wrote: Good call. For our American readers... does anyone know where I can obtain a list of states/counties and their regulations in regards to call recording? I would think the Public Utilities Commission for each state, but that's just a guess. A quick tickle of google came up with: http://www.rcfp.org/taping/ * I take no responsibility for their content. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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 to caller when called party has picked up - without initial Answer()?
Peter Svensson wrote: Dial() application will answer the incoming line once it is ready to bridge the two calls together. If nothing else then one can always modify the Dial() application to play a specific sound just prior to sending the answer. I have not checked if there already is a generic way to hook into Dial that early. I looked at show application dial when I read the question earlier today. There is a hook for playing a message to the recipient of the Dial before patching them together, but I didn't see anything for the other way around. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Vocera Badges
John Middleton wrote: Anyone got any experiences of these with *, and also costings? Someone mentioned them on the list several months ago, but I don't think anyone mentioned actually using it. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] redirect different phone number to different IP phone
Video Dery / Internet du Royaume wrote: Hi I have a simple question but I cannot find the answer. I have a line with 2 different phone numbers What kind of line? There has been some questions in the last day or so about DNIS, so I'm not sure that it can be done on inbound analog lines. I want to redirect each phone number called to a different IP phone Example Someone calls 5551234 and the call is redirected to IP phone 192.168.0.2 Someone calls 5551235 and the call is redirected to IP phone 192.168.0.3 exten=5551234,1,Dial(SIP/phone1) exten=5551235,1,Dial(SIP/phone2) Customize accordingly... -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Authentication against voicemail password database
Adam Robins wrote: I would like to allow my remote users to dial in from their homes, cells, etc., and instruct Asterisk to forward calls made to their office extension to a number of their choosing. The wiki entry on Asterisk call forwarding shows how to do this. For security purposes, I would like to front-end this by asking the user to supply a password for their extension. Ideally, this would be their voicemail password. Is there a cmd I can use in extensions.conf to check extension and password against the voicemail database? The only thing that comes to mind for me is loading the voicemail configuration from a database, and using an AGI that can read that database to authenticate and process your call forwarding. An upside to this might be the ability to allow users to change their own password(which I'm not sure they can do with voicemail.conf). -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Re: phone rings when I'm using it over VOIP - WHY?
Joseph wrote:
On Thu, 2005-01-27 at 08:40 +, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Joseph [EMAIL PROTECTED] wrote:
When I use my phone to make VOIP call and another calls comes from POTS
my phone rings to POTS caller. Why?
Shouldn't it generate busy signal!
Yes, but there are all sorts of configuration errors that could result
in the behaviour described. Without knowing your particular setup, it
is impossible to know what the cause could be. Perhaps you could
describe in more detail.
My setup is really simple.
I have Sipura-3000 connected to * with phone1 and another SIP phone2.
Here is my context:
exten = 1,1,Dial(${phone1},20,tr)
exten = 1,102,Dial(${phone2},20,tr)
I have setup two phones and have VOIP, when I make call over VOIP I
think channel return status -1 (the call is bridged). So when a call
comes from POTS my phone1 keeps ringing and I want to ring phone2 not
mine.
If the channel return status 0 the call is transfered to priority n+1
and that is what I want.
Why priority is 0 when I pickup the phone and hear dial tone (without
calling out); and priority is -1 when call is connected bridged with
another party?
To my understanding in both cases the phone1 is busy so why return
different priority code???
Take a look at DIALSTATUS at:
http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
Also,
Do you get a call-waiting beep when you're on the phone with the
original party?
--
Andrew Thompson
http://aktzero.com/
http://dev.asteriskdocs.org/
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Re: [Asterisk-Users] Trouble with Quicknet Linejack
Marcelo Echeverria wrote: I have a Quicknet Linejack in /dev/phone0. My phone.conf is: [interfaces] mode=dialtone format=slinear echocancel=medium context=mayores device = /dev/phone0 Only I can mark 7 digits, soon asterisk tries dial automatically. I cannot mark 8 or more digits. 6 or less digits work ok. It sounds like you don't have a pattern to match dialing more than 7 digits. Post your mayores context from extensions.conf. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] OT: iax.cc/sixTel local DID question
When you choose to add an unlimited local DID to your account from their control panel, do you get to pick the prefix/NXX, or just the area code? Their isn't any indication of whether or not clicking the Add button will immediately add a number to my account or take me to another screen to pick a NXX. (I don't want to click the 910 area code to find out they giving me a Wilmington or Raleigh local DID that's just useless to me.) Thanks! -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] OT: iax.cc/sixTel local DID question
David Mallwitz wrote: Their isn't any indication of whether or not clicking the Add button will immediately add a number to my account or take me to another screen to pick a NXX. snip The form lets you choose the NXX. Actually, it didn't. I asked in a ticket what happens and the response came back that I would have gotten an email about it. I've sent the request back, so we'll see what happens. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialing Delay
David Shaw wrote: Hello, When I dial out there is a long delay in dialing. What are you dialing out to? What are you dialing out from? What does your config look like for the answer to the above questions? Is this normal? It varies. The dialing sequence from Stargate Command takes longer than the dialing sequence from Atlantis. I believe this is due to the true DHD at Atlantis, and notably upgraded Stargate. I am open for discussion. ;) As you can see, your question was rather vague, and was nowhere near specific enough to get the answer you were seeking. Unless, you just happenned to be wondering about the Stargate... -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Mobile Callings
Germán Micale wrote: Hi, Does someone knows what kind of device I need to call from my pc to the mobile network? In Spain VoIP prices are very similar to call to a mobile than do it from an other mobile. So, I want to plug some device to the PC and get out the call throught it, but I dn't know what kind of device I need. Thanks in advance look up: cellsocket There are other similar devices, but the names now slip my mind. What type of network is your cell phone on? (cdma, gsm, tdma, etc) -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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 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. 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. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] First configuration
Germán Micale wrote: Hi everybody, I'had install asterisk, but I can't configure it to validate with my VOIP provider. Perhaps you could tell us who your provider is? Also, my telepathy is not working this week, so you'll actually need to send us the relevant sections of your config files. What I need is recieve our costumer's calls and redirect it using allways a unique user and password. Receive calls from where? Redirect them to where? Could some one help me? It's very likely that someone here can, but right now, we know nothing about your specific configuration, or your problem. Error messages are helpful, too! -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] H323 on Asterisk@Home
Walid Azab wrote: Guys, I am about to install H323 [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] (Asterisk CVS-v1-0-12/22/04-05:48:41). I noticed that the default h323h.conf file is not set up. I also noticed that many of you here say that it is better to use Oh323. What is the best scenario here for me? Well, that depends, what does your scenario look like? What hardware, how will it be accessed(over the net)? Should I go with the already existing h323 located on (channels/h323)? or go for oh323? Choose based on features implemented, hardware known to work with each version, ability do identify, find, and download appropriate versions of software, etc. This would be a good place to start: http://www.google.com/search?q=site%3Avoip-info.org+h323+oh323 BTW, how can I get PWLib! You could type PWLib into your favorite search engine... -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Any Notices from voiceconduit?
Michael Lyszczek wrote: Anyone have any issues like thisI am fwding broadvoice to zaptel,1 with my t100p and the t1 goes to a zhone zplex10b.. I can ring extension 1, which is pair 1 of the channel bank, but it doesnt recognize offhook and it keeps ringing the phone after I pick up. Also, its like each ring is like a seperate call as far as the callerid history goes. Anyone have any ideas? Michael Lyszczek What does that have to do with voiceconduit? -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM adapter for analog telephone - connect with fxo or fxs to Asterisk
Robert Rozman wrote: Hi, I have Siemens combiset - it can gateway GSM phone to normal analog phone. It has output where I can connect regulat analog phone. How can I connect to combiset with Asterisk - via fxo or fxs ? It is my understanding that an FXS device generates dial tone, and a FXO device expects dial tone to be provided to it. According to www.sipura.com, the SPA2000 I have has two FXS ports on it, and I know that they generate dial tone. If your gateway produces dial tone(which I expect it does), you need an FXO device to connect it to asterisk. This is the same type of device you would need to plug a standard phone line from your local LEC into asterisk. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Confrences..kinda
Chris wrote: Hey all, Is there any software or something out there that anyone knows of that will allow me to have a conference in asterisk (or possibly not if you know another solution) where I can see who is talking at the time? Kinda like teamspeak or ventrillo. I'm not getting my hopes up, but any help would be much appreciated thanks everyone! -Chris MeetMe? http://www.voip-info.org/ -- see the wiki -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Moderator on vacation?
Eric wrote: Seriously, what gives. Can we make some changes here? I'd like to post my findings and get some help. I can't get google to show me any, but there are sites that allow you to drop off large files and give you a url for retreiving them. Perhaps someone can come up with the name of one. Find a site, upload it there, post your message with info and point us at the link. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Moderator on vacation?
Andrew Kohlsmith wrote: If you got that message it means you posted to the list from an address that is not subscribed. It's a little misleading -- I've *never* had a moderator post or deny a message I've posted from a nonsubscriber address, on vacation or not. That may not be the only reason for the awaiting moderator approval, but it is the one I often get when I forget to hit the dropdown and change the From address to asteriskuser (list email). Post to the list from an address that is subscribed, like you just did here. No human intervention required. :-) He said he was posting a large file, it may have been larger than what the list allows. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] mantis password reset link
Greetings, Does someone have the link to reset your password on bugs.digium.com? I can't seem to find one. Thanks. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Comdial PBX -- can use Asterisk as VM box?
steve szmidt wrote: If you terminate the T's in the Asterisk box and then put patch cables between the Asterisk box and your Comdial, you can probably accomplish these things. You might need to detect what your Comdial does to talk to a VM system and then configure Asterisk to answer properly. What's the best way to figure this out? I'm looking to replace a VM that talks to a phone system over analog lines and a Dialogic card. I am guessing the phone system rings the voicemail(phone system provides dial tone), but I'm not sure how extensions and digits are being used to make the rest of the features work. Is there an application I can use to listen on a line for flashes, digits, callerid, and did-type info? -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to connect two Asterisks as secure as possible without too much additional bandwidth ?
Robert Rozman wrote: Hi, I plan to connect to remote Asterisk that will terminate calls to ISDN primary channel. I'd certainly like to secure this type of service, so would kindly ask for any advice on how to secure this authentication as much as reasonably possible. What are you trying to secure? The entire datastream, the authentication(username/passwords), and/or the voice traffic itself? -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime sipbuddies table structure why?????
Matthew Boehm wrote: How so? How would you change it? Are you aware that they have written code into app_voicemail.c that allows you to store the actual soundfiles for voicemail in the database itself? You want to talk about poor database design...sheesh.. So, providing a uniform access model that doesn't depend on file paths, capitalization, or numbering is a bad thing? Read it, makes no difference, it's broken :) Whats broken? The table structure, did you miss it? Also, it doesn't say why the table structure is the way it is. It most certainly does. Seems you didn't read the REAME after all. snip kindergarden database structure lesson from readme.extconfig Seems pretty simple and easy to use. This way if new config options are ever added, all you have to do to support them is to add a new column. And if all you are storing in most columns is 1 byte, it can't take up that much space. are ever added That's entertaining. Developers working on special tasks add fields that may not ever make it into CVS stable. I think I've read of two or more that are in CVS-HEAD right now. Flexible table structure isn't we'll just add a field. Especially when the likelihood of new pieces of data needing to be tracked is high. I don't claim to know everything about database structure, but that is my job, and I don't believe a flat table is the way to go. Trust me, I just added six fields to a flat table that I wish now that I had never created. If I can't get the time to rewrite it, I am sure I'll need to add six more fields within the next few months. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unsubscribe
Kevin Walsh wrote: Bart de Wild [EMAIL PROTECTED] wrote: unsubscribe No. LOL... You are trapped on this list forever! mwaa ha ha... I know, lame and useless discussion of a useless post, but I just couldn't stand it. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unknown number CID on SIP phone
Brian McCrary wrote: Hello, I'm a new Asterisk user and I hope I haven't missed something, but I can't seem to find an answer to this issue. I have a Cisco SIP gateway terminating calls into a 7960 phone. The issue I would like to fix is if I have an incoming call without an ANI, such as directly from my TDM phone switch, Asterisk says the call is coming from the IP address of the Cisco gateway, withough the dots, so if my gateway is at 10.0.0.1, Asterisk reports a call from 10001 instead of reporting Unknown, or simply not reproting anything at all. You should be able to set the inbound callerid from the switch/gateway to a specific unknown in sip.conf file with just a callerid= line. The place I looked on the wiki didn't show a specific description for the callerid= line, but that's what I thought I read for it somewhere. http://www.voip-info.org/wiki-Asterisk+config+sip.conf (currently hosed) http://64.233.179.104/search?q=cache:IIOmLeG89KwJ:www.voip-info.org/wiki-Asterisk%2Bconfig%2Bsip.conf+site:voip-info.org+sip.conf (google cache) -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100p and 6 second delay
[EMAIL PROTECTED] wrote: I am getting a 6 second delay whenever i dial 9 to call someone using PSTN, What could be causing this?? Pattern matching, perhaps? What's your dialplan look like for the station you're calling from? -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lock the phone when no using it
Denis Galvão wrote: Is there a way(or some AGI app) to lock out the phone when user are not using it!? Thanks. You can set up an extension that does a dbput to a variable which locks the phone. When someone tries to dial anything but 911 (or local equivalent), your context should test for if the phone is locked and if so, fail it. Same thing for calls going toward the phone if that's required. You would also need an extension and possibly a password to unlock the phone, by setting the variable to another value. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] timeout
Altus Snyman wrote: Good day all I have my extensions.conf configured so that it waits 8s the answers with a message saying press 1 for... and 2 for.. How do I tell it then that if the did not press anything to should go to the operator. And/Or if they did not press something it will play the message again And/Or if they typed a wrong extension ti will read the menu again please let me know Thaks Altus Go to the wiki: http://voip-info.org/ The answer is one or both of: t, i -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SysMaster and GPL Violation -- what about IPEYA
Mark Spencer wrote: There seems to be some confusion here so I would like to make a few brief comments and will likely not add much to this thread other than these few things: 1) Digium *does* license Asterisk (as we distribute it, no additional features) outside of GPL and we *do* have commercial licensees already. 2) Digium appreciates the community keeping a watchful eye on other products in the marketplace which may be in violation of Asterisk's licensing terms. Please feel free to contact us directly if you have any concerns or questions. 3) I do not wish to comment specifically about Sysmaster's relationship with Digium at this time other than to say we are in contact with them. Thank you again for all of your support in the community. Mark There, you have an answer. Digium is in contact with Sysmaster. It will be resolved one way or another. This is where, on a web-based forum, the Admin would lock the thread... Please, can we call this thread closed? -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limit DTMF tones
Andrew Kohlsmith wrote: On November 4, 2004 12:30 pm, Henry Devito wrote: The issue is the inmates have figured out a way to dial long distance numbers by calling different private phone numbers and using that companies DISA to complete calls. So in order to stop that I have to suppress dtmf after so many digits are dialed. Any idea's? If you configured a SIP phone to not transmit inband DTMF, would asterisk translate that to inband DTMF when bridged to an inband DTMF only connection, ie your POTS line? Note: Just talking out of my head here, I've not actually tested this... In any case, chan_sip would be much more likely to be hackable to make DTMF quit working. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Limit DTMF tones
Flynn wrote: Possibly, but his working configuration most likely doesn't use SIP (I would presume): It has the Digium 2 FXO/ 2 FXS card in it. I have two Lines brought in to the fxo ports and 2 standard 2500 analog sets for the prisoners to use to dial out. Yeah, I saw that, but the replies I'd seen so far were not looking real promising, so I thought I'd throw out another idea. Even if the handsets were ruggedized, a Sipura could sit in between them and asterisk. Critchfield's response about the bridge code seems the place to look, but that's going to require coding and testing. If a SIP adapter could be dropped in and as a side effect of the configuration it broke sending DTMF out, only a few changes to the dialplan would be required to get things back in order. Anyway, it was just an idea, and he did say he was looking for ideas. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dynamic DNS causes problems
Christoph Rothe wrote: Unfortunately not. I have the same problem and the same solution here. You really have to do a restart of asterisk. I think the reason that asterisk does not always lookup the IP Adress for the DDNS-Hostname is performance. But it would be nice if a sip reload could do so. Has anybody contacts to the developers ;-) ? Since not everyone uses DHCP, I expect not everyone will be interested in this, but what about a small app specifically for checking for changes in the DNS to IP lookup of specific(or any!?) hostname fields. You could configure the app to run on your specified interval. You could possibly tag somehow or name specific instances of the hostname= fields you wanted to test. Note: I am behind a DHCP'ed connection and use dyndns myself. Fortunately, my IP stays the same for very long periods of time, so it's not been an issue, but I can definately see its use. Further Note: Although I have self-studied BASIC, Pascal, C, Java, and PHP, I don't really consider myself a programmer. I would attempt this myself, but I know there are people who could have written an app in the time it took me to write this message. If no one comes up with an easy solution for the original poster, I'll take a look at it. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] test telephone numbers
Richard Bennett wrote: On Thursday 28 October 2004 04:15, Steve Totaro wrote: i think he meant numbers that would not be billed for completing a call. No, any number is just fine. Preferably a mix of mobile and fixed numbers for as many countries/regions as possible. So often a customer will say something like I've been trying to get a call through to Uzbekistan all day and nothing works, so i have to try to route Uzbekistan through a carrier who will be able to terminate it properly. Being able to test with a number that won't wake someone up at 3am would be much easier... Finding hotels or companies using an IVR system on the internet will help for landlines, but if anyone has any out of use mobile numbers that will still play a message, this would help a lot to... Thanks for the numbers and suggestions so far, Richard. When you get a decent size list, you will post it on the wiki(if you're not doing it now), or at least mail it to any other interested parties, right? -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] test telephone numbers
Paul Rodan wrote: Yeah, I'd definitely be interested in that list too. I thought I was one of the only find the right carrier game. Sometimes customers can't place calls or have quality issues to certain countries, sometimes even certain Area codes within the U.S, so I then route those calls through NuFone/LookieLoo/VoicePulse/1 of our 6 Voice T1/PRI's, whichever works best. A place I used to work was an inbound call center. The owner would get up every morning at 5 or so and dial every one of the toll free numbers to make sure they were all still working. If any of them failed, the lady that maintained the phone system would be the next person called. That was never a happy phone call... -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Funny thing with LinkSys / IAX2
Andrew Edmond wrote: *Community, I am a new VoicePulse customer, using their EXCELLENT connect.voicepulse.com services. I have asterisk CVS head 10/20/04 running quite successfully, with 10 SIP phones, voicemail, and 2 zaptel lines. The box is running Gentoo Linux, and is running on an Internet connection which is a ComCast cable modem, with a LinkSys BEFSR41 router running the LAN. I have IAX2 configured in IAX.conf registering my IAX2 line with their gateway. My toll outgoing calls are routed through their IAX2 gateway (my local calls go out through my Zap lines). However, I'm constantly having to restart now on the CLI (about every ten-thirty minutes) to keep my connection to VoicePulse online (incoming AND outgoing calls). I have port forwarding on for ports 5060 and 4569 to my asterisk box. After talking with user VoicePulse in #asterisk, we think that it has to be something with the router or Internet connection we have. Something about routing or NAT tables being unreliable in LinkSys routers. Anybody have any IAX2 weirdness with ComCast or LinkSys routers or a combination of the two? Help! Thanks! More likely than their being something broken, the router could just be timing out the connections for non-use. Take a look at the qualify directive. The link below is for SIP, but the meanings should pass over to IAX the same. http://www.voip-info.org/wiki-Asterisk+sip+qualify Also see if just an iax2 reload (or whatever) is sufficient, or if you must actually restart all of asterisk. (This could help narrow down the issue.) -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.72[69]
Kevin Walsh wrote: Kristian Kielhofner [EMAIL PROTECTED] wrote: P.S. - Asterisk rocks! That's not been my experience. I've found Asterisk to be reasonably stable. :-) Although I only go check on bebop (my dual processor linux/samba/asterisk box) once or twice a week, I also have not noticed any swaying. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] test telephone numbers
Richard Bennett wrote: Hi, I was wondering if there already exists a list of worldwide test telephone numbers for us to use to test if we can terminate that destination? If not i'd like to suggest setting this up on the wiki. I was thinking of a list of numbers sorted per country/type/operator, of: Companies that have 'press one for English' type messages. Free-phone info message numbers. Local rate info message numbers. Defunct mobile numbers that will still play a message (from lost or stolen mobiles etc) Designated telco test numbers. VOIP test numbers. etc. NOT premium-rate numbers. For instance, if you want to test US termination you can dial the movie-phone on 12127773456, or if you need to test Belgium mobile Mobistar you can use 32494413965... Any ideas? I have this (US) number in my cell just for fun: 3034997111 It's the time equivalent of the weather channel. It plays the time from a nuclear clock. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.2
[EMAIL PROTECTED] wrote: Hello everyone! Version 1.0.2 is now available for Asterisk, Zaptel, and libpri. Would it be possible to get official changelogs for these releases? (If they're out there, please point me toward them.) Thanks. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Conferencing Server
Smarty wrote: Hi, Repeating my request again. I need to: 1. Make a Group containing some agents (SIP User Agents) as members. 2. Start a conference between the members of the Group. (The asterisk server should do the conferencing between these SIP User Agents. So the asterisk server should be able to understand the request from one member to be redirected to all the group members). I'll be really grateful if I could have some suggestions as to how can I create the Group and start the conference? There probably isn't functionality for this built in to MeetMe right now. Here's a suggestion: 1)Build a simple database to contain your groups and members. 2)Build a webpage to allow selection of the group you want to start the conference with. 3)On submit from step2, dump a .call file for each agent that should be in the conference that dials the user, and patches them into MeetMe. 4)Click said link/button. 5)Wait for users to all beep in. Additionally... 6) Release said code to sourceforge or similar so that others can use/enhance. If you're interested in building this suggestion, or having someone build it for you, please let me know. I would be interested in working on the script, but have no use for it personally at this time. If someone else is interested, I'd be inclined to devote some compute cycles to it. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0.2
Brian West wrote: Does anyone know if just installing the new asterisk will work fine or do I have to remove the 'old' install first? greets no I generally tar and gzip the three folders to a backup folder before I do upgrades. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] divert if not here
Dave Cotton wrote:
On Tue, 2004-10-12 at 11:20 +0200, Altus Syman wrote:
Good day all
We have a pbx system running sip and sipphone(Bughtone)
My question is.If a user is not at their desk,how do I tell it if a call
comes in it should direct it to someone else
Do I need a different phone for this?The only other way is that they
have to switch it off and in my dialplan on stem 2 I will have to say go
to that user?
Please give advice on this
You could set up something like:-
;Call Forwarding
;
;on
;
exten = ${CFIM_ON},1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4})
exten = ${CFIM_ON},2,Playback(call-forwarding)
exten = ${CFIM_ON},3,Playback(has-been-set-to)
exten = ${CFIM_ON},4,Playback(extension)
exten = ${CFIM_ON},5,SayDigits(${EXTEN:4})
exten = ${CFIM_ON},6,Wait(2)
exten = ${CFIM_ON},7,Hangup
;
;status
;
exten = ${CFIM_STATUS},1,DBget(TEMP=CFIM/${CALLERIDNUM})
exten = ${CFIM_STATUS},2,Playback(call-forwarding)
exten = ${CFIM_STATUS},3,Playback(is-currently)
exten = ${CFIM_STATUS},4,Playback(digits/2)
exten = ${CFIM_STATUS},5,Playback(extension)
exten = ${CFIM_STATUS},6,SayDigits(${TEMP})
exten = ${CFIM_STATUS},7,Goto(105)
exten = ${CFIM_STATUS},102,Playback(call-forwarding)
exten = ${CFIM_STATUS},103,Playback(is-currently)
exten = ${CFIM_STATUS},104,Playback(disabled)
exten = ${CFIM_STATUS},105,Wait(2)
exten = ${CFIM_STATUS},106,Hangup
;
;off
;
exten = ${CFIM_OFF},1,DBdel(CFIM/${CALLERIDNUM})
exten = ${CFIM_OFF},2,Playback(call-forwarding)
exten = ${CFIM_OFF},3,Playback(has-been)
exten = ${CFIM_OFF},4,Playback(disabled)
exten = ${CFIM_OFF},5,Wait(2)
exten = ${CFIM_OFF},6,Hangup
;
and then check the status of CFIM for each call.
I just want to make sure I'm reading this right. You're using three
variables here, CFIM_ON, CFIM_STATUS, and CFIM_OFF, right? So somewhere
above this section of code, you've defined them like:
CFIM_STATUS=*10
CFIM_ON=*11
CFIM_OFF=*12
I didn't realize you could put variables in as the extensions.
This could be the beginning of some kind of dial-plan modules
collection, where you post your macro or dial plan logic and show sample
usage.
--
Andrew Thompson
http://aktzero.com/
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Re: [Asterisk-Users] Asterisk VIA SSH Tunnels
Christopher Jacob wrote: Anyone ever set up Asterisk to use SSH Tunneling? Anyone know the pros cons? Thanks, Chris This search on google turned up close to 300 messages: site:lists.digium.com ssh If you give it some more keywords, it'll probably narrow the results down to exactly what you're looking for. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk VIA SSH Tunnels
Christopher Jacob wrote: Anyone ever set up Asterisk to use SSH Tunneling? Anyone know the pros cons? Asterisk has a command line interface that can be called from probably any shell. I ssh into my Linux box that runs asterisk then tweak my settings/run asterisk -r with no special configuration other than actually turning on and configuring the sshd, which should be done anyway. Are you sure you mean ssh? Could you possibly mean VPN(in all it's varieties)? If you want to know about securing the voip traffic, remove ssh from my previous statement and try these keywords: site:Linux.digium.com ipsec site:Linux.digium.com vpn Sugar to taste... (ie, add any other keywords that you think are helpful) -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] nufone config
Can someone post or forward me the relevant sections of their nufone
configs?
I seem to be brainfarting on making it work. All my outbound attempts
end up with results like this:
bebop*CLI iax2 debug
IAX2 Debugging Enabled
bebop*CLI set verbose 9
Verbosity was 0 and is now 9
-- Executing SetCallerID(SIP/710-1980, 9104108307) in new stack
-- Executing Dial(SIP/710-1980,
IAX2/[EMAIL PROTECTED]/19104108307) in new stack
-- Called [EMAIL PROTECTED]/19104108307
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 6ms SCall: 1 DCall: 0 [198.22.67.70:4569]
VERSION : 2
CALLED NUMBER : 19104108307
CALLING NUMBER : 9104108307
LANGUAGE: en
USERNAME: andrewkt
FORMAT : 4
CAPABILITY : 14
ADSICPE : 2
DATE TIME : 155805489
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
REJECT
Timestamp: 00016ms SCall: 00170 DCall: 1 [198.22.67.70:4569]
CAUSE : No authority found
bebop*CLI
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 00016ms SCall: 1 DCall: 00170 [198.22.67.70:4569]
-- IAX2/nufone/1 is circuit-busy
-- Hungup 'IAX2/nufone/1'
== Everyone is busy/congested at this time
-- Executing Congestion(SIP/710-1980, ) in new stack
== Spawn extension (trusted, 19104108307, 3) exited non-zero on
'SIP/710-1980'
bebop*CLI
I've tried several variations of friend/user/peer in iax.conf but
haven't been able to make anything happen. My most recent config looks
like this one, which is basically a duplicate of my voicepulse connect
entry, which does work.
register = andrewkt:[EMAIL PROTECTED] ; knock knock...
[nufone]
type=friend ; yes, i know friend is evil, it was a last resort attempt
host=switch-2.nufone.net
username=andrewkt
context=default
auth=md5
secret=mypass
At some point last night, I think I had * registering properly with
nufone, as it showed up when I did iax2 show registry Now, it does
not. I'm not worried about that yet, as my (brand new) toll free did
doesn't seem to be working anyway (doesn't ring to failover, I get a
message from my LEC saying the number is disconnected).
My dialout line is a copy of my working voicepulse out section:
[NuFoneOut]
exten = _1NXXNXX,1,SetCallerID(9104108307)
exten = _1NXXNXX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _1NXXNXX,3,Congestion
exten = _1NXXNXX,4,Hangup
My context in iax.conf is NANPA or some such. Cannot look at it now.
Although I do remember seeing NANPA in some example configs somewhere,
it didn't change my results.
--
Andrew Thompson
http://aktzero.com/
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[Asterisk-Users] is the feature list online somewhere?
The main speaker (that I believe is Mark), is looking at a list of post 1.0 wish features. Is this list online somewhere? -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1.0 Mirrors
Greg Boehnlein wrote: Hello, Please be conscious of Digium's bandwidth and use a Mirror when downloading 1.0. I have mirrored the tarballs at: I've got a little extra bandwidth laying around: http://xninja.net/asterisk/asterisk-1.0.0.tar.gz http://xninja.net/asterisk/asterisk-sounds-1.0.0.tar.gz http://xninja.net/asterisk/libpri-1.0.0.tar.gz Note: I have no idea if this will remain available, of if my host will turn it off! :) -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM phones, bluetooth and general happiness
Joe Antkowiak wrote: if you bought a 2 phone no-minute plan with unlimited mobile to mobile, and used * to connect one of your phones to your unlimited ld at home, you could essentially get very cheap unlimited mobile calling... this was the point I was trying to make... Yes, it would be a pain to dial twice, but with all these smartphones and pda phones out there, it shouldn't be too hard to write something that could easily talk to * via the mobile to mobile call... Just be careful, my VZW contract states that mobile-to-mobile doesn't apply when one of the lines is fixed or makes a majority of it's call from a single cell site. Take the phone out and use it somewhere else a couple of times a month. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: Hardware solutions to tie two offices together
Good fill in local time of day I'm looking for a piece of hardware that we can place in two offices that have decent bandwidth, but are in two different US states. There are phone systems on both sides, that have extra CO analog line ports that I'd like to connect through. One side has an IVR, the other side does not use one during office hours. The best configuration would allow callers from either office to be able to dial an extension for the other office and have it ring through to their desk. If the device could just hotline to the other side when picked up, that would be acceptable as well. I would prefer that there be no other intermediary involved in call setup. I have a SPA-2000 at home, but I couldn't find anything in the setup for doing this. If anyone believes that the SPA can do this, please provide me with configuration info. Thanks. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Hardware solutions to tie two offices together
Henry Devito wrote: What type of phone systems do you have in either office? I have done different applications for my customers in the past that wanted the same type of service. Basically if you are just using co ports this would be a tie line service. There may be a better solution though. I'd rather not link to the phone systems in such a way that they couldn't later upgrade them without upgrading the voip connection. One office is an old Nitsuko 124i(??), I'm not sure what the other one is. For the service I'd like to deliver, we shouldn't need more than CallerID and DID/DNIS/called number delivery. Even if both of those were not feasable, or not currently supported by their phone systems, it would be ok. We just would like to take advantage of the two existing SHDSL pipes. Currently there is 100-500/month spend in ld/tollfree between the two offices. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Status of conference calls at Astricon ?
Mike Benoit wrote: Is there an IAXtel 1-700 number by chance that can get me in to the conference? Then to truly experience Asterisk's power, join the same conference... ...Using SIP: [EMAIL PROTECTED] I can't make the sip address work either, at least not with the fwd communicator beta. If there is an IAX in, I could download diax again. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Status of conference calls at Astricon ?
Michael Bielicki wrote: IAX2/[EMAIL PROTECTED] you can connect gsm/g726/alaw or ilbc server is in NL Ok, an italian link to nufone astricon conf room is up running. Connect it to: IAX2/[EMAIL PROTECTED]/meetme OR IAX2/[EMAIL PROTECTED]/meetmeq The first one is to listen speak. The second one is to listen only, use that Does anyone know a schedule of when there will be people in this conference room? PS: If you're in the conf now, HIT YOUR MUTE BUTTON, I can hear someone typing :) PPS: Whoever had A Tale of Two Cities running, you got kicked off(or maybe you left on purpose?). Hmm... Now there's some psychadelic/chanting on-hold music... :) -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Suppressing CallerID in .call files
Christian Victor wrote: Wich is possible in Extensions by setting CallingPres=32. Now I am looking for a way of disabling presentation in .call files. Can you not just have your .call file dial back through an extension passing in a parameter of the destination number, so that you can activate that setting as a part of the extension context? Something like: exten =987NXXNXX, 1, (set calling pres, however you do that) exten =987NXXNXX, 2, Dial... Or, are you doing something that I'm just missing? -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cvs stable
Michael Bielicki wrote: Stable seized to exist quite some time ago. To expand on Michael's answer, stable wasn't being kept up to date like it should have been, so the statement get the latest stable version became get the latest cvs version as the standard answer for resolving people's issues/bugs in asterisk. Without enough effort being directed at stable, it was not really worth calling stable, so it was dropped. Someone correct me if I'm wrong, but as of this writing, there are no specific revisions that you can point to with known this works/this doesn't work lists for asterisk. You can pull cvs as of right now, or from any point in time, but that's about it. As I understand it, there is a push in progress for a true 1.0 release. Hopefully, it will be properly maintained for bugfix'es by digium, or the best answer might be copying the release, whenever it happens, to someone else's cvs and having the developer community that's interested in doing it fix the bugs there. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Anti Ex-Girlfriend feature for entire area codes?
Nate Carlson wrote: But if I try: exten = 8005551212/408XXX,1,Congestion exten = 8005551212/408XXX,2,Hangup() It doesn't catch it. Is there any way to do something similar and allow wildcards? Thanks! See: http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf Look a few blocks into the examples section, you need an underscore in there. When I saw this message, I realized that I goofed in my example for the .call file earlier today. -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
