similar to: Is Bell HDSL in Ontario good solution for VOIP?

Displaying 20 results from an estimated 3000 matches similar to: "Is Bell HDSL in Ontario good solution for VOIP?"

2005 Feb 09
1
Please share the experience on VoIP phones heavy using.
Hi there, Does someone can share the experience with Cisco and Polycom Phones? How rock solid are they? And who will win in sound quality contest? I heard that Cisco phones is a Polycom replicas with changed design. Is that true? What else phones is better to implement to the medium sized business? The rock solid stability and superb sound quality is a must. -- All the Best! Sergey.
2005 Feb 16
3
IAX2: Connection rejected
Hi there, I am having a problem. It looks like this: Feb 16 15:01:10 WARNING[11122]: chan_iax2.c:5546 socket_read: Call rejected by XXX.XXX.XXX.XXX: No authority found Feb 16 15:01:10 NOTICE[11122]: chan_iax2.c:1375 iax2_destroy: Avoiding IAX destroy deadlock -- Hungup 'IAX2/user/1' Even I have entry in iax.conf for this user as a friend, and * server of this user is already
2005 Feb 09
4
G.729 codec for X-lite soft phone
Hello all, Is X-lite soft phone support G.729 ? I actually use it but there is no G.729 support. Anyone know where to have it? Regards. Daniel. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050209/8cdbeeec/attachment.htm
2005 Feb 19
2
I have a odd question...
Hi all. I am going to do a simple "voting application" for a radiostation. The idea is to have listeners call in to vote on songs. What I want to do is to take a phonenumer for each song and present the result on a simple webpage. Eg. To vote on song number one, call 555-1111 To vote on song number two, call 555-2222 etc etc. When the listener calls in, a playback tells him:
2004 Nov 28
17
Wiki down?
Hi All, The wiki seems to be struggling this evening. Anyone else seeing this? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005 o800-905-6412 c713-201-1262
2005 Feb 04
3
Server Criteria
I've been doing a lot of background reading/searching of this list, voip-info.org, and Google, looking to define a good candidate for a server platform. I'm very interested in thoughts from others! So here goes... Axiom 1: if you are not doing doing much transcoding (converting between codecs), the bottleneck for supporting high volumes of simultaneous calls is system bus speed,
2005 Jan 18
4
TE110P as E1
Hello, I'm having problem with a wildcard TE110P. As soon as I load the module (wcte11xp for kernel 2.6.10), it spawns a yellow error with or without an E1 plugged-in. Any one managed to set it up in France? Here are my files: zaptel.conf: span=1,0,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 zapata.conf: [channels] language=fr context=default switchtype=euroisdn pridialplan=unknown
2004 Dec 12
1
Will Adtran TSU 600 work with *?
People on the list tend to think you can't make many cards work on a regular desktop. If you're willing to wait a couple of week I might have an answer for you. _____ From: Robert Augustyn [mailto:augustynr@yahoo.com] Sent: Saturday, December 11, 2004 7:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Will Adtran TSU 600 work with *?
2005 Feb 02
2
Asterisk@home - problem getting console output ...
Hi, I am connecting to the asterisk using asterisk -r command but I never get anything on the console? How can I enable it? Robert Btw: it is version 0.4
2005 Jan 23
4
Any experience with Sangoma cards?
Hi, I am considering A101/102/104 cards for my asterisk installations. Has any of you used these or any Sangoma cards in such environment? Any thoughts? How do they stack up against Digium cards? Any input would be greatly appreciated. robert
2007 Jul 24
1
Testers needed for VoIP router solution
Hi all, We have put together a firmware for a range of inexpensive routers. It has been configured to provide optimum VoIP performance. We have internally tested it for number of months and it looks very good. You should be able to run it easily with 20+ phones on local network ( we still did not hit the upper limit ) assuming that you have bandwidth. Your VoIP will get prioritized over other
2006 Apr 25
4
About Softphone IAX free for Pocket PC
Hello, Has anyone Knowledge about softphone IAX for pocket PC totally free? Tkanks for all. -- Sandra Salmer?n Ntutumu <makevuy@ehas.org> Tlf. Analog: +34 914888405 / M?vil: 653574298 Tlf. IP desde FWD: 656212. Ext: 10 / Tel. IP desde EHAS: 010010 Fundaci?n EHAS: Enlace Hispanoamericano de Salud - www.ehas.org Telemedicina rural para zonas aisladas de pa?ses en desarrollo
2005 Mar 28
13
Asterisk@Home 0.7 released
We had added a lot to this release to our one button install of Asterisk. Now you can have even more features automatically installed and configured. Asterisk 1.0.7 AMP 1-10-007 Flash Operator Panel 0.20 Redesigned WebMeetme weather agi scripts Midnight Commander We have added some of our most requested features. - Web Meetme is now installed by default and the meetme2
2007 Apr 20
3
Developing Marketing materials ...
Hi, I am working on developing a professional Marketing Materials for my systems. I plan on using a very good(expensive) company to do that so splitting the costs with several people would be nice. Let me know if you are interested on taking part in it. robert -------------- next part -------------- An HTML attachment was scrubbed... URL:
2009 Mar 06
5
How to verify availability of the DID connection?
Hi all, Occasionally, DIDs from different providers stop working for some reason. I would like to be able to monitor situations like that and react before any of my clients start going ballistic on me. Any ideas? Scripts you know of or wrote and willing to share? Any info?would be greatly appreciated. ? Robert ? -------------- next part -------------- An HTML attachment was scrubbed... URL:
2008 Dec 28
2
Problems with sip registrations through HP Procurve 7102dl
Hi, I have a strange problem, when I try to connect to les.net from our local asterisk server through Procurve router I seems to be connecting on any port above 1024 and when I reload sip the port is changing too ... So I never get 5060? Any ideas on what is going on and how to resolve it? ? Sincerely, Robert Augustyn 519-997-3106 ext:802 www.linqone.com ? ? -------------- next part
2007 Jan 30
2
Should I use sip gateway of PCI card?
Hi, I am planning couple small business installations and wader what should I use for 2 to 6 lines a gateway or pci card? Any comments greatly appreciated on pros and cons and brands. Thanks, robert
2006 Sep 10
2
Issue with radiobutton and remote_function in IE.
Hi there! I did implemented the radio button with remote_function to update some div via RJS. It works perfect with FireFox, but behaves weird with IE. When I click the radio button it loads data (I see the indicator showing) but not updates the div with info until I click left mouse button anywhere. It''s weird =( Here is the snippet from .rhtml code <td>
2008 Dec 13
3
Standard error of mean for aov
Hi all, I'm quite new to R and have a very basic question regarding how one gets the standard error of the mean for factor levels under aov. I was able to get the factor level means using: summary(print(model.tables(rawfixtimedata.aov,"means"),digits=3)), where rawfixtimedata.aov is my aov model. It doesn't appear that there is an equivalent function to get the standard
2009 Jan 10
2
How to monitor asterisk with SNMP?
Hi, We have zabbix running and would love to be able to monitor our asterisk box with it. I believe that some sort of SNMP is build in 1.4+ correct? Where do I find more info or a how to on what is supported and how to use it? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: