Re: [Asterisk-Users] John Brown from Chagres!
On Wed, 2003-12-03 at 03:26, Aaron Martin wrote: Sorry to everyone on the list, but for some reason this is the only reliable way to get hold of John. John Brown of Chagres Technologies, please contact me! I have been trying for weeks now to get hold of you via email and phone after wire transfering money into your account for the Grandstream phones we ordered, but so far I have not had a single response, nor have the phones arrived! I've been trying to contact him as well. The last contact I've had from Chagres was on the 10th of November, but their voice menu says it was changed on the 20th, so he was still around as of then. I'm about to charge back the item I've ordered from him. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Has anyone else had problems with Chagres?
I have an order for an SPA-2000 through them, and they won't respond to any email I send them. I've also tried calling them, but I can never get a human. I've left voice messages, but they haven't responded. Does anyone know any other way I can get in contact with them? Thanks! Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P stopped working
On Sat, 2003-10-25 at 18:49, Ken Godee wrote: You did do a make clean first before recompiling? Yes. Not only that, I tried deleting the zaptel, libpri, and asterisk directories and re-checking them out. Then I decided it might be a heat issue, so I turned it off for 6 hours before trying again. Still no luck. Then I figured it might be a corrupt library somewhere, or something like that, so I formatted and re-installed RH9. I still got the exact same error messages. All I wanted was the aggressive echo cancellation... Now I have nothing. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P stopped working
On Sun, 2003-10-26 at 08:41, Steve Meyers wrote: On Sat, 2003-10-25 at 18:49, Ken Godee wrote: You did do a make clean first before recompiling? Yes. Not only that, I tried deleting the zaptel, libpri, and asterisk directories and re-checking them out. Then I decided it might be a heat issue, so I turned it off for 6 hours before trying again. Still no luck. Then I figured it might be a corrupt library somewhere, or something like that, so I formatted and re-installed RH9. I still got the exact same error messages. I spoke too soon. After the re-install, I forgot to add fxsks=1 to my /etc/zaptel.conf. Now it works again! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P stopped working
I recompiled Asterisk with the aggressive echo cancellation on. That's all I changed, honest. After recompiling, it refused to run. I tried updating the source, etc, and eventually went back to no echo cancellation. Every time, I got this error while starting Asterisk. Please help! I have no idea what went wrong. Oh, and yes, wcfxo and zaptel are loaded, I checked with lsmod. I rebooted a few times too, to make sure everything had been cleared out. === [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found WARNING[1074404064]: File chan_zap.c, Line 6986 (load_module): Ignoring rxwink WARNING[1074404064]: File chan_zap.c, Line 626 (zt_open): Unable to specify channel 1: No such device or address ERROR[1074404064]: File chan_zap.c, Line 4949 (mkintf): Unable to open channel 1: No such device or address here = 0, tmp-channel = 0, channel = 1 ERROR[1074404064]: File chan_zap.c, Line 6730 (load_module): Unable to register channel '1' WARNING[1074404064]: File loader.c, Line 301 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[1074404064]: File loader.c, Line 396 (load_modules): Loading module chan_zap.so failed! Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Wed, 2003-10-22 at 07:44, Andrew Kohlsmith wrote: Can you _please_ trim the quoted text? There's absolutely no reason to quote the entire post you're replying to, signature lines and all... +2 points for bottom-posting though. :-) No, -10 points for bottom-posting but not trimming. If you're not going to trim, I'd prefer you save me the hassle of scrolling and top-post. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Tue, 2003-10-21 at 17:13, John Brown (CV) wrote: Can you provide more specific information. Saying Its Broke Jim doesn't provide enough content :) True that. :) My biggest complaint was how they used to sometimes take over the server's MAC address, confusing the crap out of my switch. We only detected that because we were on an HP ProCurve that we could log into and view stats on, and the MAC address kept switching between two ports. But that is fixed in the .81 release, thankfully. However, it doesn't give me much faith in their TCP/IP stack... The switch they don't work with now is a CompUSA brand 8-port switch. I don't know the model number. I admit that it's a cheap switch, but it works with everything else in my house. With the BT phones plugged in, weird things happen. When I try to access the BT web page, the phone will give me the login page fine, but when I post the password, it freezes. As in, the phone requires a hard reset, it doesn't respond at all after 20 seconds or so. I tried to look at it in Ethereal, but everything seemed normal. I have no more data than that. I replaced the switch with a Linksys, and the phones no longer lock up now. What version of code are you running on the GS ?? 1.0.3.81 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Tue, 2003-10-21 at 11:36, James Sizemore wrote: 9Fix the tftp configs so that I can host my own provisioning server. Or make a command prompt based tool kit, so that I can use Gaps with out writing a http screen scraper. So I'm not the only one who wrote an http screen scraper to handle configuring a network of phones? :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Survey: Grandstream improvements.........
On Mon, 2003-10-20 at 20:38, John Brown (CV) wrote: So please rate your ideas on a scale of 1-10 10 - Fix the TCP/IP stack. The phones don't work with certain switches (i.e. the one at my house), and occasionally do other weird things (although they fixed the MAC address takeover bug, apparently). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo for 15 seconds 002401c38308$2e05e0a0$0102010a@JUPITER m2brszwm6k.fsf@tnuctip.rychter.com 1065158738.26944.4.camel@penguin.isyourdaddy.net
On Sat, 2003-10-04 at 15:09, Jan Rychter wrote: Any chance you could describe the hardware? Was it a Via-based board? I have a setup where I use two *'s, both on Via boards. One is a Mini-ITX and the other is a full-form motherboard. Would interrupt-sharing between the X100P and another card cause this problem? (there is simply no way to avoid it on some hardware!) I can't remember exactly what mobo it was. It was made by a company called Syntax. It was mini-ATX, or whatever the step down from ATX with only 2 PCI slots is called. I believe that it was interrupt sharing that caused the problem. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] echo for 15 seconds 002401c38308$2e05e0a0$0102010a@JUPITER
On Thu, 2003-10-02 at 12:04, Jan Rychter wrote: I'm also hearing this, with an analog phone (connected to an S100U). Rather annoying. Incoming calls have an entirely different problem for me, a disastrous 5-8 second crackling/clicking sound, which seems to go quiet a while after you start speaking. The other side doesn't hear it, but it makes you miss the beginning of a call, e.g. you usually don't know who's calling :-/ This happens in a phone - S100U - * - * - X100P - PSTN setup, when somebody is calling from the PSTN. The first server that I set up asterisk on had the same problem. I was using BudgeTones and a couple X100P's. Internal calls had no echo, etc, but calls over the X100P's had tons of echo for 10-15 sec. We also got a beeping sound. However, since the problem didn't seem widespread among X100P users, we decided it might be our server hardware, which while decent spec wise, was on the cheap end quality wise. We got some nicer hardware, and the problem went away. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] eBay Sip Phone Scam.
On Thu, 2003-10-02 at 07:51, Josh Roberson wrote: Ok, see, now you're confusing what I said. Nowhere did I say I had the 102D. I said he never mentioned that it was the 102, irregardless of the D. I *DO* have the 101, which is what he was talking about. No, it doesn't mention it's the 101. This argument has now proved silly, especially since you're confusing what I'm saying, with what he supposedly is. Actually, when this was first posted to the list, I looked at the eBay listing. It specifically said that the phone had a 16x2 display, which is only found on the 102D. It seems that the listing has been changed since then, which would explain the confusion between you two. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with GPL license of Asterisk
On Mon, 2003-09-29 at 13:23, Jeff Dodge wrote: So -- If you don't distribute the compiled app to me -- I have no right to ask you for the source. Even if I pay you for your custom application and you must provide me with the source (Upon request!) I have no redistribution rights to that source code. I may utilize it internally -- but you still own the copyright. Actually, that last part is wrong. Part of the GPL is that you must extend them the terms of the GPL, which allow them to redistribute the code. In any case, if you don't like the evil terms of the GPL, then write the bloody thing from scratch. You don't have some God-given right to do whatever you want with GPL'ed software. The authors have given you something for free, with a few restrictions on how you can redistribute it. If they hadn't been so kind as to release it under the GPL, you wouldn't have access to it at all. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Security vulnerability report
On Wed, 2003-09-10 at 21:06, Tilghman Lesher wrote: Odd, I've found CVS-current to be extremely stable, so I run it on all of our production machines. No machine is ever more than a couple weeks out of sync with CVS (except for a few machines in the field which I can't get to right now). The first time I downloaded the CVS code for Asterisk, there were missing semi-colons in one of the files, so it wouldn't compile. I, for one, would be far happier with a organized release schedule. I think it's far better to have fairly frequent (but stable) releases. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I need your help
On Wed, 2003-09-10 at 20:19, Anderson Clayton wrote: Where can i find a instalation guide for asterisk? is there anyone? This is about the best you'll get: http://www.digium.com/handbook-draft.pdf http://www.wwworks-inc.com/asterisk/ also has some links. Steve P.S. Anyone want to take bets on how long it will take for Steven Critchfield to berate this guy for improper email usage? :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BudgeTone-100 Early Dial
On Tue, 2003-09-09 at 11:41, Hielke Christian Braun wrote:
I have one problem with the BudgeTone phones and early dial. When i
dial a long external number with 9+, * starts to dial to early with
just a few digits. The outgoing call is placed through the SIP provider
Nikotel. Is there some timeout i can increase so that * waits
for all the digits before placing the SIP call? The firmware on the
phones is 1.0.3.81 and they use SIP Info to sent DTMF. Sending via
inband or RFC2833 did not work at all. The * version is a week old
from CVS. When not using early dial it works fine.
I told the Grandstream guys about the problem about a month ago, they
said they'd look into it. The BudgeTones handle 4-5 digits okay (I
can't remember which), but at some point they crap out from too many
484's.
The way I handled it was to make the extension 9 go to a context that
plays a fake dialtone in the background, and handles the actual phone
number from there. In my main context, I have:
exten = 9,1,Goto(dialtone,s,1)
Then I have a dialtone context:
[dialtone]
exten = s,1,Answer
exten = s,2,Background(dialtone)
exten = 11,1,Macro(localcall-number,911)
exten = 911,1,Macro(localcall)
exten = _NXX,1,Macro(localcall)
exten = _1NXXNXX,1,Macro(localcall)
And a couple Macros:
[macro-localcall]
exten = s,1,Macro(localcall-number,${MACRO_EXTEN})
[macro-localcall-number]
;${ARG1} - number to call
exten = s,1,Dial(${POTSGROUP}/${ARG1})
exten = s,2,Playback(vm-goodbye)
exten = s,3,Hangup
exten = s,102,Congestion
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Re: [Asterisk-Users] BudgeTone-100 Early Dial
On Tue, 2003-09-09 at 16:03, Tilghman Lesher wrote: Why not just use DISA: exten = 9,1,DISA(no-password|outgoing) Because I didn't know about it. :) I'll try it out. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Restricting concurrent SIP calls
On Fri, 2003-08-29 at 23:27, Lubomir Christov wrote: we made available this patch few weeks ago: http://lists.digium.com/pipermail/asterisk-dev/2003-July/001202.html Any chance of this making it into the main source? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ADTRAN TSU 600 VP24 FXO 24 Port Channel Bank
On Wed, 2003-08-20 at 07:58, Mark Spencer wrote: The FXO ports will only allow you to connect phone lines, not actual phones, but since FXO ports are more expensive in general than FXS ones, it's likely you could find someone to trade. We probably should have a list dedicated to trading/selling/buying asterisk related hardware, but failing that i would suggest people just contact you off-list. Yeah, but will it work? What if he wants 24 port FXO, not FXS? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX IAX trunking... DP cache?
On Wed, 2003-08-20 at 11:09, Ian Blenke wrote: Brian West wrote: I would use the latest CVS for one. And try again. Unfortunately, I've tried numerous times to get a current CVS trunk snapshot to talk to *anything*, to no avail. Even getting my Grandstream phones to register with it was an apparent excersize in futility. Dropping back to 0.4.0 *immediately* worked with the same configs. I'll give it a go again with today's snapshot and see if I can get *anything* to work again. Is there any hope for a 0.5.0 release on the horizon? I would also like to see a more structured release program. It's kind of scary to tell people that they should just use the latest CVS code. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cordless SIP phones
On Sun, 2003-08-17 at 17:55, Nathan wrote: Does anyone have any recommendations for a cordless phone that uses SIP (or IAX)? It doesn't have to use 802.11b, but that would be appreciated. I think you're only solution is going to be the Cisco ATA-186, an analog-to-SIP device. Or, you could use the SIP software from TheKompany for the Sharp Zaurus PDA. :) Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode
On Wed, 2003-08-13 at 09:46, Dave Cotton wrote: I've had a few problems with my system holding the line after a call has been made, just now I rebooted and noticed the following in /var/log/messages When you say holding the line, do you mean that asterisk still believes a channel is in use even after you hang up? If so, I've seen the same thing happen several times with the X100P. If I do show channels it will show one of my SIP phones connected to one of the outside lines, but if I check that SIP phone, it is not in use, and there is no way to re-activate the channel from the SIP phone. Running soft hangup zap channel will hangup the channel (you don't need to reboot). I'm not entirely sure what causes it. So far, I've only seen it happen from 2 of our 9 SIP phones, but they're the ones most often on the phone. It always involves an outside line, so I believe the X100P is the problem, but I can't be sure. What other information can I gather to pinpoint the problem? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO mode 2147483647.1060797007@[192.168.1.210] 1060794258.27544.62.camel@RobinHood.LinuxAutrement.com
On Wed, 2003-08-13 at 11:13, Emmanuel Bergmans wrote: In order to test CTR21, I was forced to comment the line in the source file as I did not find a define or a zaptel.conf directive. It's really bad but... In my case this change has not solved the problem (see previous posting) Well, I'm in the US, and I still have the problem, so I'm assuming the problem isn't some European-only problem. Mine is sporadic, however - if you're getting the same thing consistently, then maybe your problem is worse. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ring while on phone
On Mon, 2003-08-11 at 10:19, Jim Friedeck wrote: Our CSR people need to be informed when a call is ringing in when they are on the phone. Is there a mechanism for informing an off-hook target channel of an incoming call? We have a guy who should get first shot at all incoming calls on our local lines and our customer service line. If he is on the phone, he should get beeped and then be able to place the current call on hold to answer the other calls, possibly 'parking' them for other people, transferring them, or answering their questions quickly. If these calls are not answered in a small amount of time they should go to the next CSR in line. Is this scenario possible? I'm not currently a 'phone guy' so I apologize for any incorrect terminology. I assume you're using BudgeTones. I think the problem is on the BudgeTone end, although it might be possible to fix it from the Asterisk side. If I get a chance, I might look into writing a patch for chan_sip.c that limits the number of channels per phone, or something like that. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] list proposal
On Fri, 2003-08-08 at 12:25, Steven Critchfield wrote: With the increased traffic as of late, I'm wondering if it is time to split the list again. Specifically I am wondering if it should be split along the various VoIP protocols and zap hardware, then leave a general list that does configuration other than VoIP related? The hope is that those asking SIP or H323 questions could get help from the various supporters while the main list can deal with transport neutral content like extension logic and voicemail configs. I second the motion. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] list proposal
On Sun, 2003-08-10 at 21:31, Steven Critchfield wrote: On Sun, 2003-08-10 at 21:25, Andy Hester wrote: Perhaps there is another way to cut down on increased traffic... Specifically, I would go back to the suggestion of a collaborative website for documentation. Collecting info and organizing into Howto's would reduce the number of times people ask the same questions. Also, the documentation could grow as quickly as the project. Unfortunately, I don't have a place to host it currently. Ideally, the list would just be for issues that aren't already addressed. Any one else interested in this? While it still needs to be done, the majority of those type questions will still happen as the newest users still don't use google until told to do so. I don't buy that. I think that people are much more likely to check out documentation linked to directly on the site than they are to utilize Dr. Google's resources. Even if you google, the results can be confusing. Also, some people aren't quite sure what question they need to ask, and some entry-level documentation would help that. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP phone recommendation
On Tue, 2003-08-12 at 11:45, WipeOut . wrote: The Cisco is from what I have heard a good phone but is VERY expenisve.. My suggestions would be to go with either a SNOM 200 or a Grandstream Bugetone.. Where can one get a SNOM 200 for less than a Cisco 7960? The Cisco's are about $300 on eBay (with power supply). I can't find a SNOM 200 on eBay, and retail seems to be $300. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Leftover Budgettone issues
On Thu, 2003-08-07 at 01:56, Brian Capouch wrote: 2. This phone does not act like all my others do when I am talking and a call comes in. Instead of the jarring ADSI !!!BOING!!! followed by a series of call waiting beeps, instead I get a ringing tone in the earpiece which is audible to the other party as well. If you find out, please let me know! I've tried all sorts of settings to make it stop that. I'd like to just make it not support call waiting at all on the SIP connection, that would be easiest, but I can't find a way to do it. The BudgeTone configuration doesn't seem to be able to turn this off, either. Hopefully they'll fix this soon... Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [OT] unsubscribe
On Thu, 2003-08-07 at 10:01, Justin Carlson wrote: unsubscribe Has anyone ever been on a mailing list where you could unsubscribe simply by sending a message with unsubscribe in it to the mailing list? I swear, every list I've been on, people try to do that, but it doesn't work on any of them. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New SIP Phone
On Wed, 2003-08-06 at 16:20, Andy Powell wrote: It's just a proxy service like fwd it will work with asterisk... The phones they are selling with the deal are Grandstreams. Perhaps that explains why nobody can get to the site to order Grandstreams right now. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] callwaiting in sip can't be disabled
On Mon, 2003-08-04 at 14:31, Brian West wrote: What type of phones? Grandstream BudgeTones. Is it a function of the phones? Is there any way to limit them in sip.conf to one channel each? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel banks, etc.
Where can I find a good tutorial on how channel banks work? I need to get a 6 port (or so) channel bank for FXO. I need to find some information on which ones are supported well under Linux and with Asterisk, how to configure them, what specifically to look for in a channel bank, etc. I'm pretty new to all this, so I'm not familiar with a lot of the terms and such. Thanks! Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] phone rings while already on a call
Our office is set up with Budgetones internally. Occasionally, someone will be on the phone, and their phone will ring. How can I make it so that it will go straight to voicemail? Thanks! Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phone rings while already on a call
On Fri, 2003-08-01 at 13:50, Dan wrote: I think that you must disable Call Waiting functionality. I can't find where to disable it... I set callwaiting=no in zapata.conf and sip.conf, but neither seemed to help. I grepped for callwaiting in /etc/asterisk and couldn't find anything helpful. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] callwaiting in sip can't be disabled
At least any way I've tried. I put callwaiting = no in sip.conf in the [general] section and in the section for my specific phone, and it still sends through calls even though I'm already on the line. How can I disable it? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Vonage
I know this has probably been rehashed a million times, but please bear with me for a little bit... Vonage claims that I can't use their service without having it go through the ATA 186. I see no reason to do that, when I can have Asterisk simply connect directly. Has anyone been successful in spoofing Vonage into believing your Asterisk server was one of their ATA 186's? If I could do that, we would probably switch our phone lines over to Vonage. Thanks! Steve Meyers ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Vonage
On Thu, 2003-07-31 at 10:07, Ricardo Villa wrote: There is no way for you to know the vonage password associated with your account. Even if you sniff out the tftp download, its encrypted. Is there any comparable service that isn't as anal? Or even better, is there any service that uses IAX instead of SIP? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Vonage
On Thu, 2003-07-31 at 10:25, nathan wrote: Iconnecthere (www.iconnecthere.com) works without any problems here, even behind NAT. I looked into them, but there are a couple of problems with them. First, they don't seem to have numbers in my area. They have my area code, but only for a city that's not in my local area. Second, there's no way to contact them without joining, so I can't ask them any technical questions. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Vonage
On Thu, 2003-07-31 at 10:20, Humberto Atristain wrote: 8x8 is the only one I know (or packet8) a little less important What specific information do I need to get from them in order to get Asterisk to connect directly? I assume I'll need the following: * SIP id * SIP password * Codec * Server IP Anything else? Has anyone else connected directly to Packet8 from Asterisk successfully? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Vonage
I just found this link: http://www.dslreports.com/forum/remark,7292324~root=voip~mode=flat It suggests that your username is your phone number, and your password is the 10 digit activation number. Steve On Thu, 2003-07-31 at 15:23, Joe Cooke wrote: I haven't tried it yet, but I believe the following is correct: SIP id: the original 10-digit activation number that you use to initially register your phone - this is *not* your phone number. SIP password: unknown Codec: g723.1 Server IP: packet8.net I would assume that a packet capture would confirm most/all of this. I'll see if I can get a capture from my DTA tonight. - Joe On Thu, 2003-07-31 at 10:20, Humberto Atristain wrote: 8x8 is the only one I know (or packet8) a little less important What specific information do I need to get from them in order to get Asterisk to connect directly? I assume I'll need the following: * SIP id * SIP password * Codec * Server IP Anything else? Has anyone else connected directly to Packet8 from Asterisk successfully? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
