[asterisk-users] RE: Web based call control
There's a better way. Take a look at how to do a Find me at http://www.voip-info.org/wiki/view/Asterisk+tips+findme You can have the call only completed when they press a key on the receiving phone. No voicemail will trigger that. Chip Schweiss -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jordan Novak Sent: Monday, May 14, 2007 10:45 PM To: asterisk-users@lists.digium.com Subject: Web based call control Does anyone know if it is possible to use a manager command to answer an incoming call and not consider it answered unitl it is received. Here is an example, I am deivering a call in the dialplan to a home telephone number. I don't want his voicemail to answer and I have no idea how long it will take to go to their home phone voicemail, but I don't want to deliver the call there, I want it to go to the next priority in asterisk. So I was thinking that it would be nice to build a web interface that they could have a button to answer with. This would send a manager command to the server telling it to answer the channel, any thoughts on how to do this. ___ --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] WiFi phones and WDS (Wireless Distribution System)
I'm about to start working with WiFi phones on my Asterisk installations. Can anyone tell me if they are using WiFi phones on wireless network that is extended with WDS and how well the phone handles jumping from access point to access point while on a call? Do any WiFi phones support WPA encryption or are they all still under the impression they are only being used on public hot spots? Thanks! Chip Schweiss ___ --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] Speeding up the dial of DTMF's in SIP channel
When using dtmfmode=info, Asterisk is hard coded for 250ms
tone duration in chan_sip.c
I've changed this to 100ms without any ill effects.
Chip Schweiss
-Original Message-
From: apalma [mailto:[EMAIL PROTECTED]
Sent: Wednesday, March 15, 2006 3:26 PM
To: asterisk-users
Subject: [Asterisk-Users] Speeding up the dial of DTMF's in
SIP channel
I'm dialing DTMF's in a SIP channel using the options:
[sip.conf]
dmtfmode=info
[extensions.conf]
exten = _XXX,1,Dial(SIP/gateway,,D(${EXTEN}))
(this is a custom SIP gateway, which receives the DTMF's sent
from
softphones through Asterisk, and based on them, build the
destination
PSTN number).
My problem is that Dial send the DTMF's to the SIP/gateway
user at a
rate of about 1 DTMF each 300ms. I'd like to know if it's
possible to
speed up that rate, or even, if it's possible to send the entire
extension as a single DTMF string.
Does anybody has a clue about how to do this? I was looking
the options
for the Dial command, and nothing like that appears on it.
Thanks a lot for your help.
--
Atly.
??lvaro Palma
___
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RE: [Asterisk-Users] TE411P Really Bad Echo
I can definitely vouch for Sangomas cards with hardware echo cancel. Ive been installing Asterisk boxes for about 6 months now using Digium TDM cards and Sipura SPA-3000s in small installations. This past month I installed in a small office with 3 pots lines. The echo was very bad and of course the phone company (SBC) claims the lines pass all tests. I exhausted all the echo cancel combinations in zaptel and still had echo and bad noise during double talk. I installed a Sangoma A200 with hardware echo cancel and you would never know there was a problem. Its the best sounding connection Ive heard through Asterisk. Chip Schweiss -Original Message- From: asterisk [mailto:[EMAIL PROTECTED] Sent: Friday, February 10, 2006 10:35 PM To: asterisk-users Subject: Re: [Asterisk-Users] TE411P Really Bad Echo It was Digium's opinion that perhaps the card had a VPM. We got a replacement TE411P, I implemented it tonight and still the exact same echo problem. At this point I feel like I can rule out failed hardware. I contacted Digium support and now they are telling me it's something with my carrier, and I should call them. I called Bellsouth, and they ran a full stress test on the circuit taking me offline for about 30 minutes. The end result is that the circuit test passed with no errors. Bellsouth says it's not in their network, Digium says its not their card, and I have a te411p with VPM disabled in the wct4xx kernel module because something doesn't work the way it should. My customer is wanting to know about sangoma cards with the echo cancellation, and at this point I'm nervous to recommend any hardware. I'm going to look into the sangoma that you suggested. Are there any other kinds of products that I could look into either Passive or Active. Thanks Stagg Shelton www.oneringnetworks.com Matt wrote: try sangoma carrier grade 104d hardware EC card. we're using it ourself. Best Regards Matt - Original Message - From: Anthony Rodgers [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 07, 2006 12:57 PM Subject: Re: [Asterisk-Users] TE411P Really Bad Echo For what it's worth, we have been going through very similar issues with a TE411P - with Digium support, we have basically gone as far as we can with the HW EC, and are now using MG2 with much better results. We have a Ditech EC box on order. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Feb 7, 2006, at 7:36 AM, Matthew Fredrickson wrote: On Feb 5, 2006, at 9:36 PM, Stagg Shelton wrote: I just implemented a system using a TE411P hardware echo cancellation card. Per Digium, I setup zaptel.conf, and zapata.conf the same way as I always have. To my surprise calls out to the PSTN had a terrible echo. 1 - 2 second delay, and quite clear. The echo was so bad that I had to remove the hardware echo cancellation module from the card. We are only using the 1st span of this card right now, and we have a tdm400p with 4 fxs modules installed as well. If anyone has experience with this card, can you tell me if I am missing something. 1 to 2 seconds?! That's ridiculously huge. I don't think you'll find a echo canceler anywhere that can fix your echo problem. If it gets better with the VPM disabled, then definitely contact Digium tech-support about it. Matthew Fredrickson ___ --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 ___ --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] How do you force Asterisk to use only specific codecs?
Is there a way in Asterisk configuration to force the use of specific codecs only... for example: Never use GSM Try G.723 if available with end point Try G.729 if available with end point Try G.711 if available with end point Something like that? Thanks! Chip __ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month! http://sbc.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How does Asterisk handle connecting two IP end points?
When a connection is carried between two IP end points, does the Asterisk server incur CPU usage to pass the voice bearing circuit between the two end points? Is it possible to have Asterisk setup the call but hand off the voice traffic to be handled directly between the two end points? Thanks! Chip __ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month! http://sbc.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Another Newbie Question
I'm getting ready to give asterisk another shot here. Didn't have a lotta luck last time, about 7-8 months back. I have been scanning the list all this time though, lurking. A question that comes up from time to time, that I have yet to see answered is; Is anyone actually using * as a primary phone system in a small/medium sized business with more than a dozen stations and a real receptionist who handles calls? If so, could you email me so we could chat some? Thanks kindly for any input Take care chipper ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Terrible audio quality using Asterisk and X-Lite?
Greetings! I have made great progress thanks to this group. My Asterisk seems to be working for the most part. I am using the following equipment/software: * HP Vectra VL - Pentium Pro CPU - 256MB RAM * Redhat Linux 8 - Loaded straight from distro CDs as Developer Workstation - latest updates from RHN * Asterisk (latest as of two weeks ago when I used CVS checkout) * X-Lite SIP Client on a Windows 2000 PC * 10 Base-T network between the PC and the Asterisk server * The Win 2000 PC is acting as a router to the Internet. It has a 10/100 NIC connected back to Asterisk and a wireless NIC to the Internet. I have configured using Andy's quick start and the Asterisk manual as guides. I have setup a single extension with voicemail. When I call, the generic voicemail greeting answers and allows me to leave a message. I receive an e-mail with the message attached. However, the quality of the recording is terrible. I also notice that the quality of the audio during the voicemail greeting is intermittently great and sometimes choppy/missed completely. Are there any settings that I can tinker with to enhance the VoIP quality in this configuration? Thanks! Chip __ Do you Yahoo!? SBC Yahoo! DSL - Now only $29.95 per month! http://sbc.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using Asterisks with old Rhetorix 4108s?
Does anyone know of drivers/software that will allow me to use the old Rhetorix 4108 T/R boards with Asterisk? Thanks! Chip __ Do you Yahoo!? Yahoo! Calendar - Free online calendar with sync to Outlook(TM). http://calendar.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
