Friday Calls About voIP - IRC #voip-users-conference on Freenode.net
voip users conference
RSS / MP3 Downloads
Snom M3 and Siemens S675/685IP
We discussed the two cordless SIP phones and also SIP providers and attended transfer. Thanks to e4strategies.com, and onsip.com for their support.INUM, Polycom firmware, asterisk 1.4 and 1.6, ulaw/alaw changes
Is voxbone INUM initiative going to work? Polycom makes firmware downloadable from their site. How long will asterisk 1.4 be around before 1.6 takes over? New ulaw/alaw code.Recording from G722 Conference Bridge
Thanks to Michael at http://www.mgraves.org and David http://zipdx.com here's the wideband recording.Bounty and Donations
Do we need a new system to post and accept bounties for asterisk development? Follow the discussion here.VoIP in the Financial Crisis, Polycom Slide Show
How will VoIP fare during a recession or depression? It's about value added services. Polycom microbrowsers can show slides from Flickr.com and more topicsPari II: Security system using GSM
Discussion of using a GSM SIM card in a remote region having no landline.Phone Provisioning, Polycom config files, asterisk 1.6
Asterisk 1.6 has a module for provisioning phones. Polycoms and how to avoid re-modifying files every firmware release.HD Voice, Skype and the price of fish
Live capture, edited out motorcycle, dinner call and other frivolities. Interesting visitors.Safi Systems visual call flow and IVR
With SafiWorkshop you can design, test, debug and deploy advanced call routing applications from a single, unified development environment. Also more on asterisk 1.6 and various asterisk-related betasPhotos
Videos
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Post-prandial stroll
Added by randulo/Zeeek
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ACN Video Phone
Added by Samuel Morales
fridays at 12:00 pm eastern time (usa), 16:00 utc
New DTMF Information
Instructions on how to call are HERE.
Join the conference using the instructions on this page. Download Past Conference Archive Recordings
Michael Graves Blog
Naming The Cafe At Digium
Ok, I just have to record this here as it’s too funny. Digium has a contest to come up with a name for the cafe at their new offices in Huntsville AL. This generated lots of idea, comments, Twitter traffic, etc. They settled upon “Beans & Bytes” which results in this reply: OUCH! …and then another… …these must [...]Soft Phones: An Opportunity For Someone
<RANT> Let’s face it, the soft phone segment of VoIP space is stagnant. There’s been little change in literally years. I’ve spent the past six months looking for a Windows soft phone that was G.722 capable. In the course of my search I’ve tried a number of soft phones. The list is getting lengthy: Counterpath’s X-Lite [...]Blog & Blog Some More
Over the past couple of weeks I’ve split my blogging in two. Respecting the fact that many readers have a singular focus SOHO VoIP and networking topics remain here as usual. However, more varied topics and curiosities are on a new personal blog at http://www.mgraves.org/personal/. Both blogs are cross-linked with buttons on the main menu [...]Street Price On Polycom IP450 (updated)
Giving the online retailers a day to get their shops updated it seems like the new Polycom SoundPoint IP450 is going to start out selling for around $220. Given that the IP550 is listed in places for only a little more ($230) I suspect that the actual selling price of the IP450 will eventually drop [...]Celebrating One Year Of Blogging
Funny how things just sort of happen. In looking back through some emails I just discovered that it was one year ago today that I started writing this blog. At least that’s when I setup my account at Wordpress.com. The first real post came perhaps a few days later. This blog was back then an effort [...]listen live or to archives
Blog Posts
New blog posts
Posted by Fred Posner on November 19, 2008 at 4:24am
Asterisk + Skype: unbeatable connectivity!
Posted by randulo/Zeeek on September 25, 2008 at 7:28pm
AstriCon Swag
Posted by Fred Posner on September 23, 2008 at 11:12pm
Retiring my Polycom IP500 for a new IP650
I am anxious to see if Michael Grave's prediction of me becoming a "Polycom fanboy" will come true.
Geekdom as deep as mine sometimes makes becoming a fan of a technology almost too easy. I just ordered the "productivity suite" and can't wait to see if the one feature I badly n… Continue
Posted by randulo/Zeeek on August 24, 2008 at 4:49pm
Looking for VoIP service providers..
If you do offer Voice Services to your clients, I'd like to spend a few minutes with you and see if we can partner our services and form a mutually beneficial service.
thanks
Mike
Posted by mike - IPID on August 21, 2008 at 4:49pm — 1 Comment
VOIP Users Conference Mailing List
Re: [VOIP-Users-Conference] Re: Soft Phones Suck!
Randy,Thanks. Indeed the Saturday morning call is usually uncomfortable for
me. But I have always been following up what you guys are discussing about
Woody
AR1688 and PA1688 based IP phone mailing list --
[link]
VOIP and AR1688 BLOG -- [link]
Re: [VOIP-Users-Conference] Re: Soft Phones Suck!
Well, that's good news. I'm not sure where you are, but if you arelocated in Asia, I guess our Friday calls are Saturday morning at 3AM.
If you are in a time zone where it's possible and comfortable, I hope
you'll join the call sometime.
[link]
Of course you can also hear recordings.
Re: [VOIP-Users-Conference] Re: Soft Phones Suck!
Randy,VoIP is quite a small world :)
We are still working on hardware IP phones, moving into wide band codec
world as everyone else.
Woody
AR1688 and PA1688 based IP phone mailing list --
[link]
VOIP and AR1688 BLOG -- [link]
Re: [VOIP-Users-Conference] Re: Soft Phones Suck!
Hi Woody,Nice to see you here! What's new in your world? I used to follow the
yahoo group since I have several of those PA1688 IAX hardphones. One
of them was working very well, in fact.
Randy
Re: [VOIP-Users-Conference] Soft Phones Suck!
Michael,Nice to know G.722.1 is also license free now from your article. Can you
also share with us what you found about Speex wide band codec usage among
those soft phones?
Woody
AR1688 and PA1688 based IP phone mailing list --
[link]
VOIP and AR1688 BLOG -- [link]
Soft Phones Suck!
After half a year of searching for a G.722 capable soft phone I'vereach some conclusions, and created a lot of frustration. But
underlying all of this is a chance for someone to take over this entire
product category. The emperor is dressed very shabbi
indeed.
[link]
Re: [VOIP-Users-Conference] Re: Talkshoe DTMF issues, possible solution
They almost did last year!Hey, Neil of ideaSIP told me there's yet another URI that can be
dialed via a phone, I just tried this:
7463#22622...@proxy.ideasip.co m
it works on the Polycom and eyeBeam on XP, but not on X-lite on the
Mac.Probably my config, though.
Check it out on a live show. Or wait until about 11:45 EST and try it
Re: [VOIP-Users-Conference] Re: Talkshoe DTMF issues, possible solution
I would have thought that since they actually generate revenue throughPSTN based termination the use of direct SIP based connections would
simply not be allowed. After all, why diminish their revenues
deliberately?
The fact that they allow such connections, if occasionally problematic,
is a bonus. Push the issue too hard and you may see the connectivity
Re: [VOIP-Users-Conference] Re: Talkshoe DTMF issues, possible solution
Awright now, reality check:All this theory is great, but in the end there are no free lunches,
and no free cell phones and no $1 cellphones. The iPhone is NOT
cheaper now than it was before.
Talkshoe is in a certain business. They provided the SIP port as a
favor to hosts and the small number of people who can install a SIP
Re: [VOIP-Users-Conference] Re: Talkshoe DTMF issues, possible solution
no you are not understanding the fact that it is talkshoe who should fix theirservice. the fact I cant sip url dial in from a polycom directly and get into
the conference is a issue. I have fixed the dial code but I cant sit and fix
it every time they break it.
the dtmf issue should not fall back to the enduser to fix all the time. its
Re: [VOIP-Users-Conference] Re: Talkshoe DTMF issues, possible solution
Rich are you seriously that lazy or that inflexible? You could easilyaccomplish all of this with 3 lines in your dialplan:
[talkshoe]
exten => 8559,1,SIPAddHeader(Subject:
<passcode>22622</passcode><pin >5555551212</pin>)
exten => 8559,n,Dial(SIP/...@66.212.134 .192,60)
exten => 8559,n,Hangup()
Dial 8559 from any phone that includes that context and you should be
Fwd: WEBINAR: HD VoIP - the next sound barrier? TUESDAY!
I thought this might be of interest....I sat in on the presentation aweek ago.
Michael
==================BEGIN FORWARDED MESSAGE==================
Free Webinar Promo - HD VoIP - the next sound barrier?
HD VoIP - the next sound barrier?
Tuesday, November 18, 2008 2:00pm ET / 11:00am
For over 100 years, telephones have grown to become a primary means of
Re: [VOIP-Users-Conference] Re: Talkshoe DTMF issues, possible solution
I would guess this is a significant part of their revenue, but I don'tknow. Yes, they do get money from the termination and SIP is only
there to keep hosts happy and to allow international callers.
In other Talkshoe news, Dave told me they had alrready started looking
into g722. It requires special hardware development for their
Re: [VOIP-Users-Conference] Re: Talkshoe DTMF issues, possible solution
OTOH, there are the things that you control and the things that otherscontrol. You know you can improve your situation if you can make the
change. You can only hope that you might influence the others. Their
priorities are usually not ours.
I've always wondered if Talkshoe was generating some revenues through
Re: [VOIP-Users-Conference] Re: Talkshoe DTMF issues, possible solution
Oh, I agree with Rich's comment that you shouldn't have to change adialplan and that that must be on their end. But on his problems
dialing in, there's something else in play there because tens of
thousands of people call in all the time.
What I am interested in here is that (hopefully) their use of the
About VoIP Users Conference
randulo/Zeeek
created this social network on Ning.
for asterisk beginners
A Beginners Guide To Successful VOIP Over DSL
Michael Graves always has interesting VoIP articles in his blog. This is one of them. Check it out.Digium Mailing Lists
This is the master list of the Digium mailing lists. The most busy is asterisk-users. Another useful one is asterisk-biz, the business list. There are a lot of other specialized ones and the most widely known is probably asterisk-dev, the developers list.Asterisk The Future of Telephony
Amazon.com: Asterisk: The Future of Telephony: Books: Jim Van Meggelen,Jared Smith,Leif Madsen by Jim Van Meggelen,Jared Smith,Leif MadsenONLamp.com: VoIP and POTS Integration with Asterisk
Learn how to configure an Asterisk system so that it can receive calls from other SIP clients and interoperate with both analog and VoIP telephony services. John Todd takes asterisk setup and configuration to a new level by adding POTS hardware to connect your pbx to the phine line(s)ONLamp.com: Asterisk: A Bare-Bones VoIP Example
Asterisk is both an open source toolkit for telephony applications and a full-featured PBX application. Learn how to configure a simple telephone system with Asterisk in this tutorial. Excellent first-look article!© 2008 Created by randulo/Zeeek on Ning. Create your own social network
