Re: [asterisk-users] FWD and IPCall
192.246.x.x =! 192.168.x.x if that is what you're thinking. Anyway, registration Refused sounds like you're getting up to FWD and attempting to authenticate, but failing at that point. double-check your iax.conf settings against the FWD Extra Features settings... of course make sure IAX is selected too with FWD. -Troy Shane D wrote: I get the following output: Jan 7 10:56:22 NOTICE[4666]: chan_iax2.c:7536 socket_read: Registration of '886036' rejected: 'Registration Refused' from: '192.246.69.186' It shouldn't be trying to do something on my network (192) should it? On 1/7/08, Shane D [EMAIL PROTECTED] wrote: I will try that when I return home in about two hours. I'll let you know. On 1/7/08, Benchev [EMAIL PROTECTED] wrote: I have a problem. I have tried everything that is in the book The Future of Telephony as well as on the FWD (freeworlddialup) website, and there is still a problem. My asterisk box is not able to associate with the FWD server. I get: Registration Rejected by [insert IP], and I can't use my IPCall number to reach my Asterisk box. Any suggestions? If you try dnsmgr.conf enable=yes; what happens? Boyko ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -Shane Blog: http://blind-geek.com/blog/ CoOwner: http://sjtechzone.com AIM: inhaddict Skype: chatter8712 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisc for Taking Calls for Radio
Shane D wrote: I had thought of that. And sorry for the misspelling of asterisk. For some reason, I thought C. Anyway, I would prefer not to use a softphone, but I could if need be. I just want a phone number, located who cares where, that will ring my asterisk box. My friend has done this successfully for free. Does anyone know of a service for this? IPKALL.com There may be others too. -Troy ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zoiper iax registation: facility rejected
sean darcy wrote: I'm trying to setup zoiper ( formerly idefisk ) to use my asterisk server at work from home. I've setup zoiper for iax, set the ip address to work's fixed ip address, user: home, password: password but the zoiper log shows: 11:02:35 Rejected registration for 'home@my-office-ip-address' with cause 'facility rejected' 11:03:35 Rejected registration for 'home@my-office-ip-address' with cause 'facility rejected' and on the asterisk server at work I get: NOTICE[5072]: chan_iax2.c:5252 register_verify: No registration for peer 'home' (from my-home-ip-address) Could it be something simple, like missing registeriax=yes for the extension? -Troy ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Data calls through TDM2400E
Stephen Kratzer wrote: Hello. Has anyone been able to successfully make data (dialup modem) calls through a TDM2400E? We're able to make fax and credit card calls fine, but cannot successfully make modem calls using a 56K modem connected to a patch panel connected to an FXS port which then gets bridged to an FXO port connected directly to a phone line. We have 'echocancelwhenbridged=no' set in /etc/asterisk/zapata.conf. We're dialing into a Lucent TNT which drops the call with a cause code of 11 (DCD-Detected-Then-Inactive. The modem detected DCD, but became inactive). The modem call works fine when connected directly to a phone line. Is there anything else that I can do to get this working? Thanks. I dunno about TDM2400E, but perhaps you might get a connection if you could slow the modem down by using the appropriate extra settings for the modem (IE +ms=v34 or -v90=0 etc) to force a non 56K/v90 connection? -Troy ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ***SPAM*** Announcing: Click-to-Call with VIDEO
snip Please don't change the title of my post. It is disrespectful. One thing is to give your opinion about its content, and another to be self appointed editor of this forum. We posted in this forum because it is a contribution to the asterisk community, and because it is free for a month, and maybe even longer if the community so demands it. If you agree or disagree with it fine, but let others decide. They know spam when they see it. Thank you. C. Savinovich VideoReps.net snip As one of the others I say it looks like spam to me too. I won't be trying your product anytime soon, partly because of the way the matter was handled. -Troy ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tone in linksys pap2t
Walter Willis wrote: i have the problem in the hardware linksys pap2t, I am install asterisk with asterisk-gui and work fine but the hangup the phone (linksys pap2 t), no tone and sound like tu,tu, tu , tu , tu , tu , tuu what is the problem with phone ??? add param special??? Note: i am mark number phone and wait ... sesonds and call. thank you. Are you hearing stutter dial-tone? If that is the case, maybe turn off MWI (Message Waiting Indicator) or similar on the pap2. -Troy ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk GUI
I would have been convinced if you had not top-posted! heh Rob Schall wrote: Tom, I disagree with your argument for a number of reasons. Each of these reasons should be more than enough to convince you I'm correct and you should do it my way and only my way. And for the record, VI and CLI. Rob Tom Rymes wrote: On Jun 19, 2007, at 12:37 PM, Senad Jordanovic wrote: Tom Rymes wrote: [snip] How many times does it have to be said? Don't feed the trolls! Tom Tom...Who in your opinion is a troll? Senad Well, technically, I was calling the original post a troll, not the original poster. More specifically, the usage of troll I am referring to resembles the fishing technique more than the mythological creature. Basically, a troll in this context is a post that someone makes simply for the purpose of starting a heated discussion on a very touchy subject. In other words, the original poster is trolling for people who will get all bent out of shape about their post and fire back a heated response. For example, a user could post a message to the list asking I'm new to Linux and Asterisk. Should edit my dialplan by hand, use FreePBX, or buy a commercial solution? Imagine the response as you tried to convince them to buy PBXWare, FreePBX users try to convince them that they should start out using FreePBX, and others go on about how hand coding a dialplan is the one-true-way® to learn Asterisk. Generally, the original poster is just looking to get everyone stirred up over nothing. In other words, Paul's original post of GUI bad! CLI good! was just the sort of post that is going to get folks fired up re-re-restarting the age-old discussion of which is better: CLI or GUI. Basically, it could be like posting any of the following: - Which is better: emacs or vi? - Which linux distribution is the best? - Which is better: Macs or Windows? All of these questions share the following: 1.) They have no right answer (macs are better for some, Windows for others, and linux for others still, not to mention OS/2, BSD, etc) 2.) People on the various sides of the debate have extremely strong feelings on the matter 3.) Nobody is likely to be convinced that the other side is right and that they are wrong. 4.) They have all been discussed thousands of times before, and nothing new is likely to be said on the matter. 5.) The only purpose served by the discussion, due to the reasons above, is to clutter up the mailing list. 6.) Any discussion thread regarding these sorts of topics is best avoided. For a more thorough description of an internet troll, see the following wikipedia article:http://en.wikipedia.org/wiki/Troll_% 28internet%29 In other words, if you see a post that is just going to result in a re-rehashing of the last rehash of a specific subject, just hit the delete key instead of clogging up the mailing list with yet another thread on whether a GUI or a CLI is better. (for example). In Paul's defense, it looked to me like his original post was simply a joke that was misunderstood. (I thought it was funny, anyway) I suppose I should take my own advice on this one, but sometimes I guess we all just can't resist. grin Tom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] linksys pap2 version2 ata DTMF issue
Doug wrote: At 12:51 5/31/2007, Bruce Ferrell wrote: Troy Ayers wrote: My asterisk box doesn't recognize DTMF from my analog phone, plugged into my ATA(linksys pap2 version2). I can make/receive calls fine... it's just that, for example, I cannot login to my asterisk voicemail. Softphones (such as x-lite) are fine. I've turned up a few articles via google where some people have this trouble, but have not seen suggestions on how to fix. I presume this is an ATA problem, and I expect not much is tweakable on the Asterisk side regarding this, but still I am looking for suggestions for either Asterisk or the ATA. -Troy Can you access IVR systems outside of asterisk? I've found that when I can then voicemail doesn't work. When voicemail works, outside IVR doesn't. No IVR anywhere. Later I found rfc2833compensate. Setting this to yes helps so that I can talk to voicemail, but still no IVR outside asterisk. I changed my outgoing calls to my FXO port and IVR works fine there, so I presume my voip provider is to blame. I have asterisk 1.4.2 -Troy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] disable musiconhold
Patrick Fortin wrote: Hi I would like to disable correctly musiconhold for my users when they are using the callwaiting feature. I have set in modules.conf noload = res_musiconhold.so Now I don't have music on hold when I use call waiting but I have this warning: -- Music class default requested but no musiconhold loaded. Is that the correct way to disable it Is there a way to get rid of this warning Thanks Patrick Not sure if it's the right way or not, but a simple way would be to put in a silent audio file as your MOH. -Troy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] linksys pap2 version2 ata DTMF issue
My asterisk box doesn't recognize DTMF from my analog phone, plugged into my ATA(linksys pap2 version2). I can make/receive calls fine... it's just that, for example, I cannot login to my asterisk voicemail. Softphones (such as x-lite) are fine. I've turned up a few articles via google where some people have this trouble, but have not seen suggestions on how to fix. I presume this is an ATA problem, and I expect not much is tweakable on the Asterisk side regarding this, but still I am looking for suggestions for either Asterisk or the ATA. -Troy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GUI: Not Found. Move along
Tim Verscheure wrote: Hi there, I just installed the GUI for Asterisk 1.4.4 and correctly set my settings but when I use my browser to access it, it gives me an error saying Not Found. Nothing to see here, move along with asterisk in the header and footer... anyone had this problemn before? greetz Try https:// not http:// -Troy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk answering machine
I'm learning asterisk, and decided to make myself an answering machine out of it. Seems pretty straightforward to use an agi (perl) to do what I want. What I want is: Answer the phone. check for time of the day If TOD is during the time I sleep I announce i'm sleeping prompt caller to dial1 (or whatever) to connect to my extension then go to voicemail if busy/una, otherwise go straight to voicemail.if no digit was pressed. If TOD is during normal waking hours or caller ID matches whitelisted numbers, just connect to my extension then go to voicemail if busy/una. I'm nearly done, but I had a thought: before I re-invent the wheel, does anyone know if this has already been done? My searches only saw basic answering machines examples. -Troy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
