[Asterisk-Users] lilte help please
In his Outgoing context, should it not be 9|1NXXNXX, to strip the 9 from
being sent to the provider?
Kevin
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent: October 31, 2005 7:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] lilte help please
problem i can't get asterisk to dial to sip provider no matter what
provider i choose
the prefix and telephone format is the main problem and i cant figure it
even thoug i looked at example and
diD not work for me
i took exmple on nufone and net2phone configs !
IF I UNDERSTAND THINGS WELL, i should dial 9 then phone number !, i always
get you dialed worgn number
any ideas
[OUTGOING]
exten = _91NXXNXX,1,Answer()
exten = _91NXXNXX,n,Dial(SIP/[EMAIL PROTECTED]/${EXTEN:1})
exten = _91NXXNXX,3,Congestion
You do not want to answer a call that is in the calling process.
Remove that.
To provide any better answers, we'll need ot see the context that
your sip phones are in along with any other contexts that are
included.
In your example above for nuphone, do you have a context like [nuphone]?
If so, what statements are included in it?
Can you copy/paste what the CLI is showing when you place a call?
It would be helpful to see that.
Until you understand exactly what you're doing, get rid of the n as
a priority and simply use numeric sequential numbers. In the above
example, change to 91NXXNXX,2,Dial and watch your CLI when placing
a call.
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[Asterisk-Users] goiax.com
As a spin off of that, 10 or so numbers you can call anytime, and then 10 more numbers after that in 24 hours for the random occurrences of ordering pizza. But youre right, there are really normally only 10 people I ever try and call, the problem is, they have 3 phone numbers each. But the 10 standard and 10 random, I would be content with that. Kevin From: snacktime [mailto:[EMAIL PROTECTED] Sent: October 18, 2005 8:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: RE:[Asterisk-Users] free dids on goiax.com On 10/18/05, Sergey Okhapkin [EMAIL PROTECTED] wrote: I completely agree. No reason to provide unlimited free service, put some reasonable restrictions like no more 10 different numbers could be called a day or no more than 20 calls a day. Be able to configure up to 5-10 numbers that you can call and no more than 10 or so calls per day total. That's more than the average person calls in a day anyways. And like you said it's free. And if someone complains show them the door. Chris ___ --Bandwidth and Colocation sponsored by Easynews.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] free dids on goiax.com
That really is a shame, goiax.com has been the best free termination service I have seen. The call quality was excellent, better then some paid services I have used. One idea, I'm not sure if you already did it, only allow one concurrent call per account? And now DIDs, thanks from all of us for the great service. Kevin -Original Message- From: Matthew Simpson [mailto:[EMAIL PROTECTED] Sent: October 18, 2005 2:05 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] free dids on goiax.com GoIAX, the Asterisk community's free IAX provider, is offering free US dids now. I loaded about 175 dids in and put up a very beta sign in page. Unfortunately I had to restrict the free us/canada outbound calling back down to toll-free only. There was a lot of war dialing and prank calling going on. I'm working on some stuff to hopefully curb that kind of stuff down so I can unrestrict outdial again, but this is the problem with free.. there is always someone that will abuse it. If anybody has any ideas on how to keep the abuse down let me know. The best ideas I have now is to only allow a certain amount of calling per month, add velocity checking, and somehow put some accountability into the sign up process to keep the prank callers and multiple account abusers away. yours, Matthew Simpson GoIAX -- www.goiax.com TxLink -- www.txlink.net ___ --Bandwidth and Colocation sponsored by Easynews.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] DID on analog line
Same reason why some ISP's charge so much for IP addresses. Because they can. Every carrier ISP I have ever looked at, Unlimited IP Addresses, but when a company chooses that particular carrier, they charge between $5 and $20, seen as high as $25, per IP address to their clients. Why? Once again, because they can. Kevin -Original Message- From: Jeremy Gault [mailto:[EMAIL PROTECTED] Sent: October 14, 2005 3:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] DID on analog line George Pajari wrote: Depends on your ILEC/CLEC. Here is Vancouver they are the same price as non-DID trunks with DID numbers $2/ea in quantities 1000 (from at least one CLEC). I have heard of CLECs in the US where DIDs are a tenth of this cost. Yep. We're using US LEC here (with a PRI) and they charge us ~$4/month for a block of 20 DIDs. I'm not sure if they would do any of the analog DID stuff, though. Actually, I don't think you can purchase straight analog lines from them (unless you co-lo at their switch.) Instead, if you need analog, they'll bring in a T1 and setup a channel bank for you. I've always wondered why some places charge so much for DIDs, though. Jeremy -- Jeremy Gault, KD4NED[EMAIL PROTECTED] Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 Want a free GMail invite? E-Mail me and let me know! ___ --Bandwidth and Colocation sponsored by Easynews.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] Wanting to Make a PocketPC have a secureConnection to asterisk server
What kind of connection? Voice or Configuration? SIP? IAX? ssh? I have an ssh client for PPC in one of my archives somewhere. Sorry, dont know what its called, where its from, its been a while since Ive needed it. But it does exist. If thats what you need, Ill take a look for it on the weekend. Kevin From: Kellner, Peter [mailto:[EMAIL PROTECTED] Sent: October 12, 2005 11:18 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Wanting to Make a PocketPC have a secureConnection to asterisk server Does anyone know of a good solution to create a secure (encrypted) connection from a pocketpc (IPAQ 6515 in my case) to an asterisk server? Thanks Peter Kellner http://PeterKellner.net ___ --Bandwidth and Colocation sponsored by Easynews.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] country code list
I'm curious, for a $2511/min call, which +1 number was this? +1900? Kevin -Original Message- You think country code 44 is a mess, think about country code 1, it spans many countries ... some in +1 have had $2511/minute rates. Yes twenty five hundred eleven united states dollars per minute! Country code 1 is really a region code (north america) and becuase of the different countries there are different rates, interconnection fees, laws governing the numbers, etc. So it could be worse :P -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] goiax caller ID
I'm not sure what he/she was sending as the caller ID information, what I was trying to do, was send a normal 10 digit number as caller ID. Is there any solution to this? Or anything planned? Thanks for your time, Kevin ___ --Bandwidth and Colocation sponsored by Easynews.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] re: DTMF woes, continued
I'm afraid this may not be helpful, but I will try, When I was working at a previous company, they where just starting to switch everything over to VoIP, since I was the new guy, I got one of the VoIP lines because of not having any more real phone lines. I had the same problem, the Asterisk guy fixed it by changing the way DTMF tones were sent in the phone settings, from InBand to one of the others, and made a change in Asterisk. After this, I had no more problems. I hope this can help you with a starting point. Kevin -Original Message- From: Esteban Guana-Jarrin [mailto:[EMAIL PROTECTED] Sent: September 26, 2005 9:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] re: DTMF woes, continued Hi Yair, Please let me if you managed to fix the DTMF tone issue, which you were experiencing couple of months ago. If not can you share any advancement. I'm currently experiencing the same issue, I can make outbound calls but DTMF will not work when dialing IVRs. My configuration is [EMAIL PROTECTED] 1.5, registering to Voip provider (Symbio), codec is g.729 and dtmf mode is set to rfc2833. Your help will be much appreciated Thanks Regards, Esteban _ View 1000s of pictures, profiles and more now at Lavalife http://lavalife.com.au ___ --Bandwidth and Colocation sponsored by Easynews.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
