Re: [asterisk-users] OT: Google Plus
On Sun, Jul 10, 2011 at 7:48 PM, Tzafrir Cohen wrote: > I corrected a few factual errors on your part. Then I answered some > direct questions by you. But if you only look for feedback from the > believers, why do you bother asking here? My bad, in that case. Apologies! :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Google Plus
On Sun, Jul 10, 2011 at 12:39 PM, Tzafrir Cohen wrote: >> I don't see you on G+, are you there? > > Me? You may see me there if it proves to be a federated service. Tzafrir, I know you so I know you won't take this as a personal insult. Why comment on something you aren't a part of? I can easily understand people not wanting to be on any of these networks, but I don't understand how they (not you in particular) can know what they're talking about if they haven't even seen it first hand. I guess it ends with the statement, not federated, not worth doing. That is a limitation I don't agree with, but we're not all the same. :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Google Plus
On Sun, Jul 10, 2011 at 12:25 PM, Tzafrir Cohen wrote: > On Sun, Jul 10, 2011 at 12:17:52PM +0200, randulo wrote: >> On Sun, Jul 10, 2011 at 10:16 AM, Tzafrir Cohen I don't see you on G+, are you there? :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Google Plus
On Sun, Jul 10, 2011 at 10:16 AM, Tzafrir Cohen wrote: > Google Plus seems to be a walled garden. Wait for the API. :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Google Plus
On Sun, Jul 10, 2011 at 11:07 AM, Steve Davies wrote: > Thanks for that Gordon. What appears to be missing at the moment is > the ability to interface or collaborate with a group of 'strangers'. You watch the stream to discover people but obviously that's a long process. It will be even longer as adoption grows. > It would be good if there were a way to broadcast a 'we're here, come > join us' to bring a group of VoIP people together, a bit like an IRC > channel name can do, or a Facebook fan page. Hangouts are broadcast to the public (everyone's stream) unless you state otherwise. Nothing stops anyone from blasting out names. If you can find my post about VoIP people, you can add your name in the comments or asl me and I will blast it out. > I thought that sparks might cover that, but I'm not entirely sure how > sparks work yet. Sparks is currently just a topic search and it's pretty lame in everyone's opinion. > I agree that SIP integration would be great. I think it'll be a while > yet but if anyone will allow it, it'll be Google. :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Google Plus
On Sun, Jul 10, 2011 at 11:07 AM, Steve Davies wrote: >> Can you suggest a good way of finding/following appropriate >> VoIP/Asterisk people once on Google+? How do you then group them? Just >> in a Circle, or some other mechanism? It's word mouth now, but I think there will be discovery mechanism soon. >> I've just created a "VoIPy" circle - So I can then invite people I know into >> the circle by email address, and/or looking at someone else's circles and >> seing if they have something relevant in their summary tag and adding them >> into your own circle... (Or using their people search - e.g. for 'randulo' :) Once you found me, you should have been able to find the post where I've put names of most of the VoIP USers COnference people. More then added their own. https://plus.google.com/104027218792812194992/posts/Xvnbp1YWf9K >> You can have people in more than one circle. Right now, it's a bit like a >> media-rich version of twitter with excellent filtering (the circles). I >> don't have camera/microphone/speakers on my PC, (got real desk SIP phones!) >> so haven't tried the audio/video chat yet, but the typing "instant >> messaging" type chat works just fine. You have to try the Hangou because that's an amazing feature and it's the one I want to see with SIP interface so we can bridge to a SIP conference. >> I think Google are still slowly gating people into + though. I did have some >> invites, but seem to have used them all up now (google didn't tell me how >> many, the "invite" button just went away after a while!) >> I'd love to see SIP integration into it, so I can use my existing SIP toys >> with it. That would be my wish, too. In the end, it is a process of finding the right people. You can see all public posts in the stream. However if there were 20 people say, from this list in my "Asterisk" Circle talking aout SIP integration, we'd keep it private, NOT to hide, but to not bore our other friends in "Basket weaving" Circle. I encourage anyone who's the + and interested to look me up. I can easily blast out more names as suggestions. :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Google Plus
Go ahead and lambast me for this post, it isn't specific to Asterisk, but: G+ has only been open at all for a week and I already am chatting with over 200 people who are into VoIP, Asterisk and all the rest of the stuff we here care about. If you don't care or are anti-social, fine. But you owe it to yourself to check it, because a lot of cool VoIP people are there and after all, Google themselves are doing some great stuff with VoIP, XMPP and video, and steadily moving towards open source. Come drink the Kool-Aid! :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Polycom - Which provisioning protocol to choose ?
On Wed, Jun 29, 2011 at 11:58 AM, Olivier wrote: > Among TFTP, FTP, HTTP and others, which protocol would you select to > provision Polycom phones. > At the moment, I'm using TFTP but I'm wondering if I should pick something > else. I use HTTP at home for better boot time and FTP is the server is distant. I didn't like using TFTP because I didn't want to install yet another server. If you already run one, it may be the best choice. :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice receiving call problem
On Thu, Jun 23, 2011 at 3:12 PM, Tim Panton wrote: > You should probably not mention the voipusersconfere...@gmail.com address > this for week's VUC > as at the moment the gateway ignores any calls to it. > > If/when it comes back to life, we can realistically expect wideband through > to zipdx. This said, I see that http://Bluejeans.com/vuc works with Gtalk so we'll see if anyone shows up there today or Tues-Wed. :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] eComm - Are you going to San Francisco next week?
eComm is next week, Monday-Wednesday. Anyone here presenting? I know a few names, Bryan Johns of Digium, for example. If you are there, please stop by and talk to VUC as we will be there live at the breakfast each day talking to participants. You can listen to these chats or call in with questions and comments by calling sip:200...@login.zipdx.com at the conference times. More info will be listed on http://vuc.me later. Mon, Tues, Wed the chats will begin at 07:45 PDT (10:45 EDT, 4:45 PM in Europe) . Friday's VUC is an open mic (meaning no guests) so if you've ever wanted to ask a question about Asterisk, VoIP or telephony, join us. As always new participants are very welcome. Hope to hear from you! :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice receiving call problem
On Thu, Jun 23, 2011 at 1:58 PM, Tim Panton wrote: > > The good news is that it supports a load of nice codecs now, including g722 > :-) And you know what that means? :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inbound SMS
On Wed, Jun 22, 2011 at 10:23 AM, Administrator TOOTAI wrote: > Le 22/06/2011 01:10, ERIC HERRON a écrit : >> >> I know Asterisk 1.8 can send out texts via SMS() >> >> Can I send Asterisk a text via a DID and it do something? To do something with SMS to a DID, I'd recommend you take a look at http://smsified.com - they do all that and more. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko
On Tue, Jun 21, 2011 at 5:47 AM, Alex Balashov wrote: > I nominate this for most imaginative use of Asterisk-users of 2011. It's already qualified to win in the grammar and spelling categories. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SMS with Asterisk
But Steve... didn't you just top post? On Mon, Jun 20, 2011 at 10:52 AM, Steve Totaro wrote: > Two requests, not from me but the community. > > 1. Don't top post > 2. When you find your solution, reply to this thread so others will be > (silver) spoon fed the answers and blindly accept them without trying things > and going through a learning curve and experimentation when they find your > post in Google. > > Thanks, > Steve T > > On Mon, Jun 20, 2011 at 1:44 AM, virendra bhati wrote: >> >> Hi Steve, >> >> >> Thanks for share your knowledge. I will revert back to you after testing >> with asterisk. >> >> On Sun, Jun 19, 2011 at 6:46 PM, Steve Totaro >> wrote: >>> >>> On Sun, Jun 19, 2011 at 8:49 AM, Steve Totaro >>> wrote: >>> > On Sun, Jun 19, 2011 at 5:13 AM, virendra bhati >>> > wrote: >>> >> Hi List, >>> >> >>> >> I have installed Kannel server into my Linux server. I have asterisk >>> >> installed into the same server. Now I want to connect both opensource >>> >> project. As per the VoIP-info website I read that in asterisk there is >>> >> an >>> >> option to send SMS. You how to do it. If you have any idea then please >>> >> help >>> >> me so thatI will make asterisk as per my need. >>> >> >>> >> - >>> >> Thanks and regards >>> >> >>> >> Virendra Bhati >>> >> +91-9172341457 >>> >> virbh...@gmail.com >>> >> Software Engineer >>> >> >>> > >>> > Asterisk has some built in features for SMS. You don't need them, you >>> > have already setup Kennal which is light years ahead of Asterisk's >>> > native in SMS apps and features. >>> > >>> > The way I do it is to use the System application. It allows you to >>> > run programs and such. >>> > >>> > With system, I call a program called Lynx which is just a simple text >>> > web browser.) to open hit a URL that Kannel deciphers and sends the >>> > SMS however you Kannel setup. The URL contains all of the information >>> > needed to send the SMS, so part of the URL is the destination phone >>> > number, part is the body, obviously you need to set your variables in >>> > Asterisk and then use the variables in in the Lynx URL. >>> > >>> > I just found this article that should answer most if not all of your >>> > questions. It work just fine for me at many locations. >>> > >>> > >>> > http://www.kannel.org/download/1.4.1/userguide-1.4.1/userguide.html#AEN834 >>> > >>> > Thanks, >>> > Steve Totaro >>> > >>> >>> This link show how to send SMS using HTTP(s) and the format of the URL. >>> >>> >>> http://www.kannel.org/download/1.4.1/userguide-1.4.1/userguide.html#AEN4201 >>> >>> The previous link is good news to me. Now I can do anything by >>> hitting a URL. it is so simple. >>> >>> Thanks >>> Steve T >>> >>> -- >>> _ >>> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.asterisk.org/hello >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> >> -- >> >> >> >> - >> Thanks and regards >> >> Virendra Bhati >> +91-9172341457 >> Software Engineer >> >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending SMS on Friday at 12 Noon EDT
On Thu, Jun 16, 2011 at 11:52 AM, virendra bhati wrote: > If I am right then will you discuss about the sending sms with asterisk into > that conference ? We can if someone wants to, that's how the VUC works. :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sending SMS on Friday at 12 Noon EDT
Hi,
This Friday Chris Matthieu of SMSified.com will explain how to send
SMS from your apps. As usual, there will be talk about Asterisk,
questions, answers, and comments about telephony, networks, VoIP and
even some OT. All are welcome to join the weekly average of 35-60
callers live. If you can't, join see http://vuc.me for the recorded
versions. If you are working on something that might be of interest to
the VoIP USers Conference, please get in touch and we can book you as
a guest.
Friday at conference time (http://vuc.me/next for the time in your
zone) please join us via mp3 stream, Gtalk, PSTN, SIP, Skype or web
widget.
SIP:200...@login.zipdx.com (g722, g711)
Skype:vuc.me
Gtalk:voipusersconfere...@gmail.com ("use call computer")
The widgets and mp3 URL will appear during the call on http://vuc.me
Hear you there,
:r
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[asterisk-users] Friday: Sipgate, Yate and Astricon
Hi All, Today on the VUC, we'll be welcoming Sipgate CEO Thilo Salmon back to tell about their choice of partners in their latest services. I will be announcing the first #VUC VoIP & Tell discount code for Astricon in Denver, October 25-27. Join us on sip:200...@login.zipdx.com using g722 or g711 codec or see http://vuc.me for the other ways to connect. Hear you there! /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voxbone numbers
On Fri, Jun 3, 2011 at 11:28 AM, devr devr wrote: > I am thinking about using numbers from voxbone. Before I make up my mind if > this is the right service for me I want to know what kinds of details will > be found when checking up on a voxbone number. > > I am interested in UK numbers. Can anyone give an example on an actual > voxbone number in service. Try this: +44 1259340614 Temporary UK number for the VoIP Users Conference. Let me know what you see? /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream + IPv6
This Friday on VUC : Part I : Grandstream Networks, they used to be entry-level phones, now moving to video and surveillance cams Part II : Junction Networks engineers discuss their move to IPv6 - this should be compelling listening for all of us (Hi Olle!) All welcome to join: 200...@login.zipdx.com SIP g722 or g711 or go look at the many ways to take part on http://vuc.me IRC is #vuc on Freenode.net http://vuc.me/irc Your time zone: http://vuc.me/next Hear you there. :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk - RIP
On Wed, May 25, 2011 at 10:53 AM, A J Stiles wrote: > Bgeh. Serves 'em right for using that POC! Who honestly *hadn't* seen this > coming since the day Skype was first released? Tim Panton, who's beenworking with SfA since it came out, posted this article today: http://vuc.li/meBRJd /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk - RIP
On Tue, May 24, 2011 at 10:50 PM, Matt Darnell wrote: > "We expect that users of Skype for Asterisk will be able to continue > using their Asterisk systems on the Skype network until at least July > 26, 2013. Skype may extend this at their discretion." It's widely believed. However, it's very possible that this was not a Microsoft decision but planned by Skype before the acquisition. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VUC: Sangoma NetBorder 4.0 and an Androis SIP client from Media5 Corp
Today, sessions 320-321 of the VoIP Users Conference will take place at the usual time, 12 Noon Eastern [ http://vuc.me/next for local times ] We'll be talking to Sangoma's Frederic Dickey about NetBorder 4.0. You can download or watch his accompanying slide presentation here: http://vuc.li/Sangoma-2011 Next, Pascal Doré, mVoIP chief over at Media5 Corporation will talk about the upcoming Android version of their SIP client Media5fone. Info and all the various ways to connect are here: http://vuc.me SIP via g722: 200...@login.zipdx.com (will accept g711, too) IRC #vuc on Freenode.net or use the web : http://vuc.me/irc Future Topics : http://vuc.me/future See you there :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype-like dialing from web page
On Tue, May 17, 2011 at 7:30 PM, Mike wrote: > Is there any softphone or TAPI plug-in that allows one to dial from a web > page? As you may know, Skype has a mechanism that converts phone numbers on > a web page to a click-to-dial application. I’d like to use this but on a > normal softphone (Bria, Xlite, other). There are two general ways to accomplish this, Flash or Java. Flash and ajax http://phono.com Java: http://phonefromhere.com/ :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday VUC: Discussion of Mobile SIP, Microsoft Lync
This should be interesting, a double header Friday at 12 Noon EDT, session 2 at 1PM EDT. 1) Pascal Doré, Media5corp. Pascal will talk about what they've been up to in the year since his last visit. Thanks to the Asterisk mailing list and VoIP community, their Media5fone was able to fix its g722 implementation. I like their product a lot and used it extensively on my old iPod Touch to make and receive phone calls on our server. SIP does rock when it works. 2) Dave Michels, VUC pillar member talks about Lync, good timing in light of Microsoft's purchase of Skype. Should be a lot of interesting commentary around the whole context. Join the call on sip:200...@login.zipdx.com or see http://vuc.me for all the call in options including: - GTalk voipusersconfere...@gmail.com - Skype bridge skype:vuc.me - live mp3 stream - IRC: #vuc on freenode.net - even PSTN and iNum VUC in your time zone: http://vuc.me/next Hope to hear you there! :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not
On Wed, May 11, 2011 at 12:48 PM, Steve Totaro wrote: > Thanks Randulo, > > I am surprised you noticed that. > I truly give thanks to all productive members of the Asterisk community. Second that! > Would you say that I am a productive member of the list and go pretty far > out of my way to help people? Most of the time give useful info, like the > Outbound Caller ID thread? I believe so, yes. Especially since you were careful to include the "most of the time" disclaimer. > I may off topic sometimes I'm sorry, I think I can claim the award for OT posts over the past 8 years or so. That includes one that allowed me to find a perl genius (Hi Dave VG) who did some great but totally unrelated work for me. I knew this community would have people like that, so I threw caution to the winds and did the OT post. Got a lot of great answers, too. Thanks! :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When someone helps you, at least let them know if the problem is resolved or not
On Wed, May 11, 2011 at 6:47 AM, Steve Totaro wrote: > I don't need a public or private "Thank You" When I was posting all the > time, I figured the ratio of "Thank you" emails to silence to be about 20 to > 1, maybe as high as 50 to 1. I agree with the others who are saying that at least a results post (It worked!) for the benefit of people trying to accomplish or fix something or even an added RESOLVED in the subject wouldn't kill anyone, busy or not. Ending it with a "thx" is optional, but like chicken soup for a cold, "It wouldn't hoit". btw Steve, for the last five years or so, every post you ever write ends with "Thanks", so you got it covered. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Which Android handset with Wifi-only ?
On Mon, May 9, 2011 at 2:20 PM, wrote: > Lots of Android handsets support wifi, like my G2, aka HTC DesireZ. Wouldn't ANY modern one have wifi? That would be odd if it didn't, would it not? :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slightly OT: Android phone as sip-gw?
On Mon, May 9, 2011 at 9:47 AM, Jay R. Worthington wrote: > gateway for asterisk? I could not find any SIP-Gateway in the Market, and i Portech has made GSM and CDMA gateways for years - nothing that works with your "old" Android phones, though. http://www.portech.com.tw/p3-product1.asp?Cid=6 :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday on VUC: Jabber/XMPP
Hi all, Friday at 12 Noon EDT, we'll be talking to Emil Ivov of Jitsi.org (formerly SIP Communicator) and Thiago Rocha Camargo (of Nimbuzz) about Jabber, something the Asterisk community is becoming more interested in by the day. Join us to learn more about Jabber and SIP or to share your knowledge and experience. As always, the VUC discussion includes people from very diverse backgrounds, so it should be a unique approach to the subject. All the info to connect is on this page: http://vuc.me - SIP:200...@login.zipdx.com (g722, g711) - Skype:vuc.me and ld.vuc.me - IRC #vuc - PSTN +15672522286 - iNum +883510012394882 - gtalk voipusersconfere...@gmail.com During the conference hours, there's a widget to join on the above page as well as an mp3 stream link. Join us! :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Friday April 15 at 12 Noon EDT
On Fri, Apr 15, 2011 at 2:16 PM, Satish Patel wrote: > Is this online conf? Or are there archived files we can review? There are over 300 recordings here: http://vuc.me :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Friday April 15 at 12 Noon EDT
On Fri, Apr 15, 2011 at 1:56 PM, virendra bhati wrote: > I want to join this conference but please tell me the topic of conference > and the process of joining step by step. You're welcome to join us! All this information is at the top of the main site: http://vuc.me > Please tell me the time in IST too http://vuc.me/next -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday April 15 at 12 Noon EDT
Hi all, You're welcome as always to join the talk on the VoIP Users Conference, VUC for short. VUC began as the "Asterisk Users Conference" but for obvious reasons, we changed the name in the first year, although Digium was our sponsor for three years. We still have plenty of you who are asterisk users and developers on this our fifth year. Come on by, listen, talk ro text on IRC. All the links are on the main URL: http://vuc.me Today's guest is Matt Bramson of InPhonex to talk about the launch of Televate. The second hour we talk about anything you can think of. If you have a g722-capable SIP client or phone, call 200...@login.zipdx.com and get on #vuc on irc.freenode.net We're also testing a bridge from Gtalk so you can do both voice and IRC with that today, by adding voipusersconfere...@gmail.com to your contact list and calling it. Thanks to asterisk community and VUC member Tim Panton for that effort and for skype:vuc.me See you there in a little over 4 hours, :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Today on VUC, Dan York on Google Voice + SIP
Hi, Today at 12 Non EDT, Dan York will be with us to talk about the recent on and off moments of Google Voice SIP URI calling. Like Skype + Asterisk (or any SIP), Google Voice and SIP compose the other shoe waiting to drop. We're following this with interest. So GV turned on SIP URI and then a few days later, turned it off. Why? Did the geeks (like us) jump on this too quickly or too heavily? Dan's Disruptive Telephony site is a reference in the field and we'll likely be talking about other news of interest as well. We'd love to have you join on on our call with this week's guest OnSIP.com by connecting via these technologies: SIP:200...@login.zipdx.com - use g722 if you have it Skype:vuc.me PSTN: +1 567 252 2286 iNum: +883 5100 123 94882 Text backchat on #vuc channel of Freenode.net - use http://vuc.me/irc if you don't have a client You can also talk to us on Twitter @voipusers or the hashtag #vuc More info: http://vuc.me Hope to hear you soon. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Experience with Phones, Asterisk and other pbx, cloud services, etc
Hi, There's been a wave of questions on Asterisk lists about phones that work well with Asterisk, services, etc. This week's VUC is all about sharing your experience with various equipment and service providers. The call begins on Friday at around 12 noon EST (9AM PST, 5PM GMT). Info, recordings: http://voipusersconference.org IRC: #vuc Freenode.net (or http://vuc.me/irc) - Local times: http://vuc.me/next SIP:200...@login.zipdx.com using g722 or g711 Skype:vuc.me Looking forward to your questions and answers... /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip/google
On Tue, Mar 8, 2011 at 1:51 AM, Dean Collins wrote: > http://www.disruptivetelephony.com/2011/03/google-voice-now-offers-sip-addresses-for-calling-directly-over-ip.html > > Nice ;) Hi Dean, What I'm waiting for is when you can send GV calls to a SIP URI without all the gymnastics needed today. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Provider Recommendation in US
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent A. > Torrenga > Sent: Thursday, March 03, 2011 11:22 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] SIP Provider Recommendation in US > > I am becoming frustrated with our current VOIP provider. Does anyone have > any suggestions for a provider that supports asterisk well and provides > solid service? Voip-info.org has a husge list of providers, but it is > impossible to tell the fly-by-night operations from the reputable providers. I've been using Asterisk with the following providers over the last 8 years: OnSIP (and before that Junction Networks), Teliax, Vitelity, Sipgate. All have been around for a while, none are fly-by-night and each one has a set of features that may or may not be what you want. [Shameless plug] As it happens, we are talking with OnSIP today at 12 Noon EST (disclosure: OnSIP is a sponsor of the VoIP Users Conference). You can also come by anytime to discuss this topic which is always a valid concern. Join us: http://vuc.me /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday #vuc at 12 Noon EST
Hello, This Friday March 4th is the 307th VoIP Users Conference. In a few weeks, we'll be starting our 5th year of this weekly live event that began life as the Asterisk Users Conference. We'd love to have you join on on our call with this week's guest OnSIP.com by connecting via SIP:200...@login.zipdx.com Skype:vuc.me PSTN: +1 567 252 2286 iNum: +883 5100 123 94882 Tex backchat on #vuc channel of Freenode.net - use http://vuc.me/irc if you don't have a client You can also talk to us on Twitter @voipusers or the hashtag #vuc Hope to hear you soon. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AstriEurope coference
On Tue, Feb 22, 2011 at 11:49 PM, Albert wrote: > Yeah, this is messages which i saw before. Weird is that its hidden > somewhere under registration form and there was no notification about > cancellation for registered users. Yes, it's in a popup when you try to register. I imagine they didn't want the people they will pitch for other events to see that not enough sponsor support came on board to have it. I was pretty surprised that it worked well enough last year. True, it's nice to have such events in Europe, but apparently the bottom line is that not enough business was generated last year and the majors backed out. I saw that Aastra is doing it's own tour of cities. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AstriEurope coference
On Mon, Feb 21, 2011 at 11:56 PM, Albert wrote: > does anyone know is AstriEurope coference is still on ? http://www.astrieurop.com/fr/cloture.php Cancelled. "Hello, It is with regret that we announce you the cancellation of the AstriEurop exhibition on May, 3rd and 4th 2011 in Paris. We thank all the companies/partners having supported this project." /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Hardphone that works well with asterisk
On Fri, Feb 18, 2011 at 2:44 PM, Sherwood McGowan wrote: > I'm VERY partial to Aastra's devices. Seriously, they don't take as long to > boot as Polycoms, they're relatively inexpensive but are not CHEAP (like a Great info. > I do have a complaint about Aastra though...Because of my client's happiness > with the brand, and because I personally think they're worth suggesting, I > spoke with Aastra about becoming an authorized reseller...filled out the > paperwork, scanned it and emailed it to the rep I was working with..and > never heard another word...For a phone device company to never get back to a Isn't this irritating? This is the era of recommendations on the net and sales via email and e-commerce. The traditional phones mfrs are so 1990 - wanting to control everything. I guess they can "afford" to piss consultants like you off, which is a shame IMO. And if they're not reading this list, they're even more lame than I thought. Meanwhile, they're probably spamming the heck out of people who aren't interested. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday 18 Feb at 12 Noon EST: SylkServer and Blink
Hi, I'm excited to announce that the guys from AG Projects are stopping by for a beer tomorrow on VoIP Users Conference, aka VUC. You should already be familiar with their excellent multi-platform SIP client, Blink (http://icanblink.com) While Adrian and Saúl enjoy a few exotic brews with us, they'll also be telling us about SylkServer: - creation and delivery of rich multimedia applications accessed by SIP User Agents. - g722 and Speex and g711, gsm - supports SIP signaling over TLS, TCP and UDP transports, - RTP and MSRP media planes, - built-in capabilities for creating ad-hoc SIP multimedia conferences with HD Audio, IM and File Transfer - easily extended with Python programming The best part of the Friday VUC is your participation. It's basically the biggest and oldest online gathering of VoIP professionals, enthusiasts, service providers, programmers and product manufacturers. SIP:200...@login.zipdx.com - use gè22 if possible, else g711 Skype:vuc.me (thx PhoneFromHere.com) Skype:+990009369991481664 (thx Tropo.com) PSTN: +1 567 252 2286 (thx ZipDX.com) iNum: +883 5100 123 94882 Text backchannel IRC: #vuc on Freenode.net - http::/vuc.me/irc Hear you there? /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
> Also OT: Google combines message context with your personal search > history to do ad targeting, so look in the mirror. > > I just made that up, though. >Not your mirror - your cookies! No, it's true! Now I'm seeing "Untimate Black Hat SEO" (yes misspelled because Ultimate was too expensive) I was just looking at an SEO report site about "top posting" and they say lists.digium.com is number 1 and needs no help. And I do kind of look like Justin Beiber will in about a half-century from now. That's why I have broken all the mirrors in the house. /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
On Wed, Jan 19, 2011 at 6:47 PM, Don Kelly wrote: >> 11:39 Parker said >> That would fall under Quirk's Exception: Intentionally invoking Godwin's >> Law to attempt to kill a thread is rarely successful. :) > > Didn't work this time :) Slightly OT: why is the Gmail ad server, which is usually all about PBX, Asterisk, etc, now showing me Justin Beiber concert tickets on this thread? Are they seeing it as that childish? /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
Although there's no requisite mention of ${Horrible_Dictator}, can't
we pretend there was, call a Godwin and kill this subject?
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[asterisk-users] Friday Jan 14th @ 12 Noon EST: Humbug
Greetings ${FellowVoIPuser},
When I saw the word "Humbug" in the Asterisk mailing list, I
remembered my friend Nir Simionovich had mentioned it to me at some
point, possibly at the big wine tasting party in Rostock during
AMOOCON, which may explain why I had forgot about it. Seeing the
thread on the ML, I checked in with Nir and Boaz and invited them to
join the VUC for a status report on the project. This should be a
great call! You can hear all about it and ask questions by joining us
live, or check later for the recorded session.
At 12 Noon EST (http://vuc.me/next for local times) dial
sip:200...@login.zipdx.com with g722 if possibly or g711. Those are
the only two codecs of ZipDX.
There is a call widget provided by PhoneFromHere.com on the home page
of the main VUC site, displayed during conference hours, as well as a
URL for the mp3 stream. Finally, you can also join by calling
skype:vuc.me
Main site for info: http://VoipUsersConference.org
Hear you there!
/r
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Re: [asterisk-users] SIP Hard Phone with SMS
2009/10/9 "Juan E. Rodríguez" : > Does any one know about a SIP hard phone capable of sending SMS messages > (Or a SIP MESSAGE) that could be read from Asterisk dial plan?? The Gigaset S675IP series of DECT/SIP phone has SMS capability, but not sure it can work with Asteris. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday Noon VUC with guest Alex Robar
Quick reminder before Astricon (from which we will be reporting from live): Tomorrow's guest will be VoIP author Alex Robar. Alex has worked with open source telephony solutions for the past four years, and has collaborated on the development and growth of an international Asterisk-based VoIP peering network. His book is FreePBX 2.5 Powerful Telephony Solutions and we'll be chatting with him Friday Oct 9th at 12 Noon EDT. To check that in your local time zone, you can go to http://tr.im/nextvuc To see how to join the weekly call, see http://VUC.me We're pleased to see more callers on g722 "HD Voice" than on the old g711 Talkshoe bridge. If you can do g722, give it a try by calling 200...@login.zipdx.com IRC #voip-users-conference anytime 24/7 but especially during the call. See you at Astricon I hope where we'll be giving away some T-Shirts and other interesting products. If not at Astricon, see you tomorrow on the air! /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday Oct 2: Digium's new Speech Recognition for Asterisk
This week Steve Sokol stops by to describe and field questions about Digium's new affordable speech recognition solution. Later on in the call, we'll also be looking at iVoIP, clients and uses for mobile VoIP. Join us on IRC anytime #voip-users-conference During the conference, call via SIP g711 or wideband g722 - or Try the web page widget to call in wideband. The details on all the above are at http://VUC.me - pronounced "Vee You See Me" :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom push application for microbrowser
Hi, Sorry about posted a protected link, I forgot we'd closed the site to spammers since we don't use it anymore. The useful content was re-posted in our list. --- URI dialing on Polycom phones http://tr.im/polyapp One of the applications originally posted in the VoIP Users Conference site by Dave Van Ginneken --- I encourage anyone interested in this kind of thing to join us in this quiet and friendly mailing list (which is a Google Group so I found the link to Dave's application). It's another way we, the VoIP users community, keep in touch and share. Also of possible interest: June 2009: Polycom Applications with Mike Seto, Polycoms VP of market and business development for Polycom’s voice communications division. http://VUC.me/2009/02/polycom-applications-with-mike-seto/ Hope some of this helps. Dave's scripts work and they point up the serious barriers to writing apps for Polycom: the docs are horrible to wade through and Polycom does not make an effort to make application programming easy of efficient. AT worst, someone should write a book with example. The microbrowser can be a useful addition to the phone. I use it to display an RSS feed. That script should also be in the VUC list somewhere. Regards, Randy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom push application for microbrowser
Hi, Take a look at this: http://food4wine.ning.com/forum/topics/submit-an-application-for Way down the page Dave VG submitted some scripts that hold the answers. We also did a Polycom App conference at the VUC, but I can't find the link right now. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and VoIP Users Friday Meeting
Greetings, We'll be getting together as usual at 12 Noon Eastern US Time for a chat with David Duffet, a well-known member of the Asterisk community and hopefully one or more of his co-authors of the new book Asterisk 1.4 Professionals Guide. In fact, I've been offered two ebook version to give away during tomorrow's meeting, so you might want to participate. Connect with us by grabbing the SIP or PSTN numbers at http://VUC.me and getting on IRC #voip-users-conference I'm happy to be able to say that several regulars of the conference will be at Astricon and we hope to meet any of you who plan to go. We'll probably do a live version from the hotel Friday, after Astricon is over: http://Astricon.net for more on that. Hope to hear you tomorrow. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoIP Users Conference Friday: Andy Abramson of VoIP Watch
Hi, This week we are pleased to welcome Andy Abramson (http://andyabramson.blogs.com/voipwatch/) as our guest. Andy is one of the most avid observers of the world of VoIP, from Asterisk and its variations to all kinds of ramifications of VoIP. I'm sure we'll pass a lot of the VoIP News in review tomorrow at the usual time and place: Friday 18 Sept at 12 Noon EDT, your local time: http://permatime.com/America/New_York/2009-09-18/12:00/VoIP_Users_conference Be on IRC #voip-users-conference on Freenode.net if you can sip:7463#2262...@proxy.ideasip.com for g711 sip:200...@login.zipdx.com for g722 http://VUC.me for general information Join us any Friday and come to meet many of the VUC regulars at Astricon! /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple Time of Day Branching problem
On Tue, Sep 15, 2009 at 3:35 PM, Tilghman Lesher wrote: > That was my fault. My apologies, but this has been added in 1.6.2. I suspected as much, since I couldn't find a single example searching on the web. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simple Time of Day Branching problem
On Tue, Sep 15, 2009 at 1:28 PM, Jim Hankins wrote: > Yes but I also have different hours for tue,thu and wed,fri. In the > o'reilly book > I have, it shows examples of using & to group them, but I am getting > the error. Not sure about the & (what version?) but you can easily just put in one line per day and be done with it. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Question about Wifi sniffing on network
François, Just to be clear, I am not on Free at all but with our mutual friends at Acropolis Télécom. I see this Freebox traffic come out of nowhere and it looks like someone trying to sniff my computer's wifi. A lot of traffic is seen and I can't help thinking it isn't a good thing. I should go see if the neighbor's MAC is the one I am seeing or if it is someone else's. randy On Sat, Sep 12, 2009 at 7:58 PM, F6HQZ wrote: > Hi my friend, > > Free and other WiFi ISP have open their boxes to public Hot Spotting and WiFi > phones. > So each box offer more than one SSID to public (often 3 in fact : first for > your own use, a second for Public Hot Spot, a third for > WiFi Phones), and you will see traffic for public, payed to the Hot Spot > owner - the ISP - by the mobile subcribers, founding very > practical to have access everywhere. > > But, it's your bandwidth which is used, and you will have more traffic that > your own like this. > > Take a look into the admin web pages of your box, and check if you can cancel > this "public" feature. > Mine is closed... > > Friendly yours, > Francois > F6HQZ > Ce message sortant est certifie sans virus connu. > Analyse effectuee par AVG - www.avg.fr > Version: 8.5.409 / Base de donnees virale: 270.13.89/2359 - Date: 09/12/09 > 06:37:00 > > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Question about Wifi sniffing on network
Hi Gordon, On Sat, Sep 12, 2009 at 6:04 PM, Gordon Henderson wrote: > It shouldn't "reach out", but it will pass on ARP and other broadcasts > from Ethernet to Wi-Fi and back again. > > I don't know what "Free" is - I'm guessing some sort of community mesh > network? Snooping any public wi-fi is going to reveal lots and lots of > stuff you really would rather not see - or that the users would rather you > didn't see!!! "Freebox" is the name of the branded router from Free, an ISP. They do DSL and are one of the biggest ISP in France. I can't figure out why I'm seeing these things as they are not TCP/IP or at least Wireshark doesn't understand them at all. It does show MACs though and when I looked them up they were belonging to the company that makes the Freebox. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Question about Wifi sniffing on network
Most of you have needed at one time or another to sniff network traffic for trouble shooting purposes. Today I noticed that one of my SIP phone's web interface worked much faster with Opera, so I wanted to see what exactly was going on. I set up Wireshark and toook a look, but I got distracted by the fact that I saw a bunch of strange things coming from "FreeboxS_nn". I know my neighbor uses Free and has this device and there are probably others in the neighborhood as well. I'm seeing this on a wifi connection to a Mac Mini. The SIP phone was a Gigaset S675IP. If you have any slowness in the web gui, try Opera 10, it somehow talks very quickly to the phone. My question is this: On a local network, there are devices like pronters that announce themselves, files sharing, computers, servers, etc. On a WiFi router, is it normal that it "reaches out" to try to talk to devices? I'm seeing groups of packets a couple times a minute, it seems like a lot. Is this normal or is it some kind of hack attempt or honeypot within range of my computer? Thanks for any relevant info. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VUC: RE: Friday 11th: Aswath Rao: "Trapezoidal VoIP is Evil" on VoIP Users Conference at Noon EDT
On Fri, Sep 11, 2009 at 4:18 AM, SIP wrote: > See... I would say the 'trapezoid' is one of the great strengths of SIP. Which is why we need you all to come and discuss this, bringing up other aspects, thoughts (even I have a few occasionally) and ideas. @Dean - you will always learn new things on VUC, even if it's only that Ebay is the devil who comes after you mercilessly when your account is taken over and they know it and did nothing to stop the bleeding. Kristian - if you do happen to make it, be sure you are on IRC and/or use the wideband bridge at 200...@login.zipdx.com (if you're doing g722) Speaking of g722, isn't there a SIP client for linux? I know that so far we haven't had anything that works for OS X, but I am hoping that Counterpath will add g722 to eyebeam or Bria on that platform. In the meantime, I get on in Windows XP with Counterpath. Recently I realized that setting Eyebeam to auto-answer and ZipDX to phone me at conference time, things are even easier. I think it will be interesting today with Aswath, but other voices are needed: be there you techies! /r PS, A brand new Gigaset S675IP and A58H accessory handset will be given away at the end of the first segment. This phone works great, I've had mine for over a year. Do I have to repeat what makes it good? I will when we ask the contest question. Be there! batteries ARE included. And it runs a long time on a charge ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday 11th: Aswath Rao: "Trapezoidal VoIP is Evil" on VoIP Users Conference at Noon EDT
Hi, We're pleased have a 25-year telephony veteran with us tomorrow, Aswath Rao. Aswath maintains that "Trapezoidal VoIP is Evil". Join us and ask questions, make comments, argue about geeky details... and maybe win a Gigaset S675IP SIP/DECT g722-capable phone with an additional handset. Those of us who have these phones like them a lot. All dial in info is here: http://VUC.me - we have a g722 bridge you can call for wideband audio IRC: #voip-users-conference on Freenode.net - web irc http://java.freenode.net In the next few weeks, we have some great guests: Sept 18th Andy Abramson the A-list VoIP blogger and industry watcher (VoIP Watch) Sept 25th The authors of Asterisk 1.4 Professionals Guide - I've had a look it's a book worth having. You might get an e-copy free Sept 25th. See you Friday! /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Today @12 Noon EDT: Skype for Asterisk, Floor Show at Astricon
Hello, In the run up to Astricon [ http://astricon.net ] we'll be talking today to Tim Panton about his experiences with SfA. You're welcome to join in! Speaking of Tim, you can join the conference in W I D E B A N D at http://api.phonefromhere.com/gateway/zdx.xsql?conference=200901 - come early, channels are limited, seriously! 12 Noon EDT today is http://tr.im/nextvuc in your time zone. More info: http://VUC.me or http://VoipUsersConference.org IRC back channel: #voip-users-conference on Freenode.net Coming up in the next two weeks, authors of a brand new book on Asterisk, guest Aswath Rao (next week) and a free wideband capable SIP/DECT phone to give away. More news of all that on the site http://VUC.me /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IPKall and FWD
> Alternatively, get a SIP account with a proper ITSP and have then register > a number for you, then you just connect to them via SIP rather than have > them rely on connecting to you. That would work well with the IdeaSip or OnSip solution among others. There are many scripts to report a new IP to DynDNS, too. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IPKall and FWD
On Thu, Aug 20, 2009 at 11:15 AM, SIP wrote: > IdeaSIP, GizmoProject, IPTel, maybe OnSIP (don't quote me on that one, > I'm not sure, but someone around has surely used it), etc, etc. There > are a lot of alternatives about. Sorry, I forgot to mention IdeaSIP.com which works great, I've had an account for years, and of course your own Asterisk box as Gordon says. You need to be able to accept call via a SIP URI but that's easy enough to configure. We will be talking about this in relation the "Free DID" subject tomorrow on http://VUC.me /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IPKall and FWD
On Thu, Aug 20, 2009 at 10:57 AM, David @ULC wrote: > How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite One way is to get a free test account at OnSip.com and create a SIP URI. Then you configure IPKall number to call that SIP URI. Works fine. You can't register to IPKall directly. Sipgate.com also will give you one free DID in an area of your choice and you can configure it to allow registering with a SIP client such as eyebeam. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday Aug 21st @ 12 noon EDT: VoIP Users Conference, More Wideband Madness
Tomorrow, Friday August 21st at noon EDT, we'll be on the rampage again with the VoIP Users Conference, much of which concerns Asterisk and the Asterisk community. We've covered a lot of ground in the past 2 1/2 years, and had a lot of great guest presenters. In the meantime, there's a new gadget to test at PhoneFromHere.com, you can hear the wideband demo on the ZipDX.com bridge: http://api.phonefromhere.com/gateway/zdx.xsql?conference=3366 This connects in wideband and you can toggle between narrow and wide band to hear the difference. Check in to the conference when it is live Friday by going here: http://api.phonefromhere.com/gateway/zdx.xsql?conference=200901 We'll probably be getting updates on subjects like Skype for Asterisk, Astricon and whatever other news is current. We may be talking about some of the big threads on the Asterisk mailing lists, such as free DID and "unlimited". Please feel free to join us and add your 2 cents or just listen. Details : http://VUC.me IRC #voip-users-conference ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP app for iPhone that works well with Asterisk?
On Tue, Aug 11, 2009 at 1:57 PM, Philip A. Prindeville wrote: > Anyone have a chance to test any of the various iPhone SIP apps? Here's a discussion of a few we've tried: http://VUC.me I like iSip but the other two are good, too. iSip has multiple accounts which is important to me. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on theiPhoneplatform?
> I was about to post on this thread that I have contacted the makers of > iSip and they got back to me, we're working on a fix. We because I did For info, the fix seems to solve the problem, VNET is waiting for Apple approval on the new version of the app. I really like the multiple accounts of iSip. More when I've tested more extensively. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on theiPhoneplatform?
On Mon, Aug 10, 2009 at 4:53 AM, Alex Balashov wrote: > Word of advice: When you try SIP clients, focus on how the far-end is > hearing you, not whether you can hear them. In my experience, that's > where 90% of the deal-breakers lie with the iPhone. Absolutely right! When testing call quality I always used to call my mother and after several minutes ask her how well she heard me, etc. We joked about how she was my VoIP tester. Now that she's been gone a few years, my brother, the "user from Hell" can't fill those shoes. Regardless of call setup or source: - he will ALWAYS find something to complain about and often it's an issue with his phone - his phones all sound like crap - he lets the batteries on his cordless run way down causing beeping and clicking - he is unable to describe in any useful way what the problem is, he just says it "stinks" I miss being able to call my mom on four different providers and ask her which was better :) Back to your comment Alex, I agree whole heartedly which is why being able to record (like the Skype test call) is better than an echo test for checking clients out locally. Then you can test by leaving messages on a PSTN connected phone of know quality. Then and only then, call people who you know have an ear for sound quality of a call and are in a location quiet enough to judge. I was about to post on this thread that I have contacted the makers of iSip and they got back to me, we're working on a fix. We because I did not realize something specific to the iPhone platform. Because of the App store, an new test verison of an app can not be installed because it is not approved by Apple. Developers can make ad hoc versions for a single phone, though and the need the UDID to do this. I never heard of UDID before, but it's like a MAC. I hope the iSip people will be able to fix the issue because I like the client a lot. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on theiPhoneplatform?
> On Sat, Aug 8, 2009 at 12:42 AM, Enrique Mora wrote: >> Finaly tried the WeePhone yesterday. >> >> The WeePhone registered with Asterisk on the first try and call quality is >> perfect with WiFi. >> It's on the same LAN as the server. For some odd reason, yesterday I tried weephone again and it worked perfectly! It has the lest good look and feel and few features, but the call's sound was good, so it moves up to second place, after iPico. iSip still introduces a clicking sound in the outgoing audio on all accounts. I have checked the mic and cable and they're fine. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on theiPhoneplatform?
Enrique, On Sat, Aug 8, 2009 at 12:42 AM, Enrique Mora wrote: > Finaly tried the WeePhone yesterday. > > The WeePhone registered with Asterisk on the first try and call quality is > perfect with WiFi. > It's on the same LAN as the server. Thanks for the extra info on the VUC.me site and for bring your results back here. I haven't yet tried any extended length calls on any ITSP. I need to try to get in touch with the developers of weephone and isip to see if they are interested in responding to the sound quality issue. > I tried several calls into the conference through ZipDX but I got cut off. > Sometimes the call would go silent after 30 seconds and at other times I > could follow the conference for several minutes. But I think this was a > problem with ZipDX and not the terminal or SIP software. I did successfully connect to ZipDX but during one call it was dropped a few times. We can try to work with David from ZipDX on this. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on theiPhoneplatform?
If you want to hang more results on this subject, please see the thread here: http://www.voipusersconference.org/2009/08/sip-for-apple-iphone/ I'm very interested in anyone who is doing development in this space so keep in touch. Basically, even though I've always preferred DECT/SIP phones to wifi/SIP ones, a good SIP client adds the wifi capability to an otherwise very good mp3 and video player, provided you invest in a new headset with a mic. So far, so good. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on theiPhoneplatform?
So far, the best iPhone platform app I've found is a $10 one called iPico. It is a one account SIP client, better designed than the others and it actually works and can dial SIP URI. I learned about it directly from Ruben Olsen mentioning it on the VUC call an hour ago. I will be posting the edited recording of the session later today on http://VUC.me but it is available on Talkshoe now: http://www.talkshoe.com/talkshoe/web/tscmd/tc/22622 We had several technical issues that I will edit out later today, but there it is, warts and all. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on the iPhoneplatform?
Ok, so now let me ask the question more directly: I am looking for the best SIP application for the iPod Touch (Wifi only). I don't care about 3g, Gsm or anything phone-related. The app has to be able to register with an arbitrary SIP service and/or dial arbitrary SIP URI. If it could dial one like this: 7463#2262...@proxy.ideasip.com That would be the ultimate app for me. If not, I can usually set up an alias on OnSIP (but they don't do # in the URI) I tried Fring, but it didn't seem optimal. I'll try again since many of you have said it works well. The two apps I mentioned both work and sound fine incoming, but the outgoing audio is either noisy (iSip) or inaudibly distorted. Note these apps are NOT free. ALso, I haven't tried contacting support yet, but I will do so after the conference call today. Thanks, r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone had any luck with SIP clients on the iPhone platform?
On Thu, Aug 6, 2009 at 10:15 PM, Alex Balashov wrote: > Which generation of the handset are you using? They differ in their > processing power and that may account for at least some of it. Alex, this is just an iPod Touch, not even a handset. It doesn't have a mic at all, I had to add one. But using fairly standard "debug" logic, The mic isn't noisy because it records beautifully. The SIP services all exhibit the same problem Skype works well! So I inculpate the two SIP clients or their configuration. iSip and WeePhone. Although Skype works, it doesn't satisfy the obvious requirement of connecting to my services via SIP. That would allow me to get calls within wifi range on a SIP pbx of my choice. Although I could make calls as well, that is better done with a real phone ;) r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Anyone had any luck with SIP clients on the iPhone platform?
Hi, I've tried two SIP clients so far and both have unusable outgoing audio quality. Skype app sounds fine, and recording the same mic sounds fine, so I can only assume there is an issue with the clients themselves. Both clients allow you to register and make calls via SIP with any abitrary provider and credentials, so they'll work with Asterisk. I've tried them with two good providers and one has unrecognizable audio and the other has noises as if the cable was badly soldered. I've never experienced such troubles with "regular" SIP clients. Anyone have any recommendations? Thanks /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday Aug 7th @12 Noon EDT Mobile VoIP
The subject of tomorrow's VoIP Users Conference will be mobile VoIP. If you have any interest, please join us. I myself am tesing a bunch of iPod applications to use with all the usual suspects: OnSIP, Sipgate, Gizmo, Skype, your asterisk box, etc. Details for joining the call are are at http://VUC.me or http://VoipUsersConference.org IRC: #voip-users-conference on Freenode.net See you there. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk: Public Beta available
On Sun, Aug 2, 2009 at 8:24 AM, Pascal Bruno wrote: > So what do you think I can do to register my license? I am running > Asterisk 1.6.10 on CentOS 5. >>> Could not generate Host-ID. >>> Make sure that you have eth0 enabled. The MAC is used in the scheme to register and it looks like it can't be read for some reason. There must be a direct channel to Digium for the support of this kind, though. Have you tried contacting them? [waits for John Todd to chime in here...] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday July 31st at 12 Noon EDT: Dave Nelsen, Skype for Asterisk beta opens, Gizmo Voice + Google Voice = free SIP calls
Hi, Plenty to talk about today. Dave Nelsen, a founder of Talkshoe, has a lot of experience in the telecommunications space and he joins us today to chat about its current state, conferencing and whatever else comes to mind. So we have a meta conference aout conferencing, it won't be the first time :) You probably saw John Todd's message on one of the lists: Skype for Asterisk is in open beta: go for it. Pricing isn't yet established, but other questions may be answered today. One of our regulars, Maxim, mentioned GizmoVoice.com which explains everything you wanted to know about getting a free Google Voice number and being able to use it with SIP, thanks to Gizmo & co. I've done this with OpenSky and it has worked well so far. Join us at 12 Noon EDT, 9AM PDT, 5PM UK or: http://tr.im/vuctime for your time zone. Info at http://VUC.me Backchannel text chat IRC #voip-users-conference (irc.freenode.net) See you there /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Analog FXO or IAX DIDS for new facility?
On Thu, Jul 23, 2009 at 5:08 PM, Danny Nicholas wrote: > My .02 - IAX may not be an option and is probably not a good one if it is. > It requires a good bit of overhead to work reliably and well. You won't go > wrong using SIP DID's, but if you use Analog FXO, I'd go for an 8 or 16 port I second what Danny said, go for SIP DID, there are many good providers and you could even have local DID in different countires if that made it easier for your correspondents. There are IAX providers too , though if you have a compelling reason to use IAX. Go with a solid, long running company on the DIDs. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Do analog gateways detect a phone is plugged in or out ?
>> > Is it usual for analog gateways to detect when an analog phone is >> > plugged in or out ? > It certainly would seem "possible" and would be a great feature request. > There probably is no circuitry existing to do it, but I would assume that > ohms, volts, or something could be measured while sending a small amount of > voltage down the FXS lines. I read this with interest. The geek in me finds it amazing that they don't detect something plugged in. YOu think in the old days especially, it'd be easy based on what Steve says and that any proprietary system would do this to aid in setup and debugging, alarms etc. Nowadays, it might be a lot harder, although for SIP phones there are ways to detect any of the common ones. Druid does this during setup, for example. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday 2009-07-24 12:00 EDT: Voxeo Labs on VoIP Users Conference
Hi all, You may have heard yesterday that Adhearsion and Voxeo have created a new baby, Voxeo Labs. From our (non-biz) point of view, I'd recommend following the blogs: http://blogs.voxeo.com/ to see how what they do might be of interest to you and your asterisk/voip activities, commercial or private. Since I myself know little about what this all means, I've invited a lot of bright people to our weekly conference. I've met Jay Phillips and Jason Goecke and they're interesting people to talk to on any subject, even outside the bounds of the usual geekdom. I only know Dan York from a few online exchanges and a visit he paid the VUC as a guest long ago. So I'm recommending you join us at 12 Noon EDT Friday July 24th to not only hear what all the buzz is about but also ask questions, make comments and drink the free virtual beer (you must be of legal virtual drinking age in your area). It's that virtual beer that got me in trouble on the second, non recorded, R-rated portion of our session last Friday. If you have a decent phone, it probably does g722 so join our call on the ZipDX wideband bridge: 200...@login.zipdx.com or call in to the Talkshoe g711 SIP URI: 7463#2262...@proxy.ideasip.com IRC back channel #voip-users-conference There's also a live stream and more information at http://VUC.me Thanks to Digium, OnSIP, e4strategies and ZipDX for all the help and support. Several regulars on the VUC come from those companies and provide a lot of insight in their areas. Should be a fun call this week. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday reminder
Please join us today at 9AM PDT, 12 Noon EDT for the VoIP Users Conference to talk about the latest news and events in the wonderful world of VoIP. IRC #voip-users-conference SIP 7463#2262...@proxy.ideasip.com for g711 SIP 200...@login.zipdx.com (for g722 wideband-capable devices) See http://VUC.me for more details /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Friday July 10th: Gigaset DECT/SIP phones have come to the USA
(thank you gmail) so if you have DID without voice mail service, your local Gigaset will handle the SIP channel as if it were a PSTN line. This feature is selectable on a per account basis. The phones also do g722 so they work with our ZipDX wideband bridge. If you are considering new DECT phones for use with SIP, the line is something you should look at. Come and ask questions Friday at 12 Noon EDT on the VUC: IRC: #voip-users-conference SIP 7463#2262...@proxy.ideasip.com g722 SIP: 200...@login.zipdx.com or see http://VUC.me for more info See you at Astricon 2009 in October ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday July 10th: Gigaset DECT/SIP phones have come to the USA
Hi, This week Tony Stankus, North American product manager of the Gigaset line is our guest on VoIP Users Conference. I have had a two handset S675IP in our small business for about a year now and my wife and I both like the phone. But as a geek, I like it a lot more than she does :) 6 SIP lines avails for all those ITSP accounts I have, plus a regular PSTN. Vmail works on both, so if you have DID without voice mail service, your local Gigaset will handle the SIP channel as if it were a ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk. Any return of experience ?
>>> Though they have written me back twice to say "coming soon" I am still >>> waiting for the software... >>> >> So you'd rather have it even when it hasn't been finished? > > Umm, no, but then when a company says "looking for beta testers - please > sign up now!" and then four months later has nothing to let me beta test, > I am a bit put off. The beta was limited. Digium wants to open it but says Skype themselves are delaying the operation. I have compelling reasons to believe this, even though I can't put them out in public. I was surprised too at the apparent slowness, but I think it will happen in good time. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk. Any return of experience ?
On Sat, Jun 27, 2009 at 11:06 AM, Olivier wrote: > Hi, > > Has anyone tried it ? > Is there any available pricelist ? It is possible no one wants to answer this due to the NDA they had to sign? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday at 12 Noon EDT: VICIDIAL
Hi, I met Matt Florell at AMOOCON and tried to record an interview. I was pleased with the results, but later found that the battery deleted the audio file when it went dead. Today, we'll have Matt live to talk about VICIDIAL and answer any questions you may have about it. For more on this: http://VUC.me IRC #voip-users-conference on Freenode.net Call 7463#2262...@proxy.ideasip.com to join or join in wideband G722 (see the site for details on that). r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and EC2 today at 12 Noon EDT
Nir Simionovich is about to become a father. He will be joining our conference at 12 Noon EDT today from the Maternity Ward to talk about Amazon EC2 cloud computing with Asterisk. Nir gave a very good presentation on this at AMOOCON a few weeks ago (see http://www.amoocon.de for more on that). The advantage here though is that he'll be live with us for your questions. All the details on how to join us are here: http://VUC.me IRC: #voip-users-conference You all have free or cheap dialing: Call (724) 444-7444 and enter 22622# PIN# - Get your PIN at Talkshoe.com or use 1# as a guest sip:7463#22622#...@proxy.ideasip.com (g711 u) or talks...@vuc.onsip.com and enter 22622#PIN# wideband, sip:200...@login.zipdx.com (g722) See you there with your EC2 questions and comments! /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ODP: Re: Polycom Stop Downloading Config
>> What happens if the http server is down? My point is that I don't >> want it >> to try and pull any config from a server. I just want it to use >> its local >> config. I don't recall this looping probelm. The value of tries is supposed to prevent this from happening. r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday 12th June @ 12 Noon EDT: VoIP Users Conference Skype to ZipDX
Hi all, In about 4 hours from this writing, the G.722 conference bridge will be brought up, the Talkshoe G.711 also, so you can call in via SIP, PSTN or Skype (experimental) http://vuc.me for all the gritty details IRC #voip-users-conference anytime today We'll also be talking about hosted PBX systems and what they might need to bring telephone into not the 20th, but the 21st century. It seems like only yesterday, but it was 130 years ago in 1879, that the Bell company acquired Edison's patents for the carbon microphone from Western Union. This made the telephone practical for long distances and it was no longer necessary to shout to be heard at the receiving telephone. Funny, but on most cell phones it is again necessary to shout to be heard; See you in a few at sip:7463#2262...@proxy.ideasip.com or sip:200...@login.zipdx.com or skype:pfh-zdx?call&topic=200901 /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamic DNS (was asterisk-users Digest, Vol 59, Issue 28)
On Fri, Jun 12, 2009 at 8:51 AM, Kengie Ho wrote: > I am having some problems with Asterisk on static IP and Sipura-1001 on > dynamic IP. Is there any solutions to in the Asterisk configuration or > Sipura-1001 to re-register when the router change IP dynamic IP? Thanks. If the ATA is connected to a router you can use DynDNS if the roiuter supports it; This will update a domain name of your choice with the dynamic IP address. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday June 5th @12 Noon EDT: Sipgate invades the USA, more HD Voice, Video chat
Already Friday, the week went by in a Flash. If you haven't yet registered for a free Sipgate DID, I suggest you go do so. If you are in, or interested in the business, Sipgate has a few tricks up their sleeves and you should be aware of them. Someone from Sipgate will be joining the conference, which begins at 12 Noon EDT (9AM PDT, 10 Mountain, 11 Central, 5PM UK, 6 Central Europe) ZipDX has a G.722 conference bridge available to VoIP Users Conference members. See the site for numbers and instructions. Sessions and info site: http://vuc.me We are also testing a Flash "video chat" that will be available at the same time as the conference. It shows up to 12 people at the moment. Be sure to have your audio muted on the video page if you use it. Video: http://vuc.me/video That page will open at the same time as the wideband bridge, at 11:45 AM EDT. You can test the video chat any time by going to http://tinychat.com/geek. IRC #voip-users-conference on Freenode.net anytime day or night PSTN (724) 444-7444 DTMF 22622# 1# SIP: 7463#2262...@proxy.ideasip.com or connect to ts.x2z.eu SIP G.722 200...@login.zipdx.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IP phone recommendation
How did I miss thins gem? >> >> >> Polycom /will/ reboot on the drop of a hat /and/ take a damned >> >> long time >> >> >> to do it (~45-60 seconds) In addition, the web interface should >> >> be >> >> >> taken away and shot - the only real way to configure them is >> >> through (T)FTP. I didn't say that, I said the person who developed the web interface should be forced to use it. Which is way worse than being taken out and shot. Especially when the "exit" button changes position in the middle of a menu operation. Maybe they need Flash in the phone? This said, damn fine sounding phones, especially in g722 on a wideband conference bridge like the one we talk on every Friday :) /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday at 12 Noon EDT: Jim Van Meggelen on the VoIP Users Conference
Hi, Like me, some of you probably remember Jim as one of the pioneers along with Leif and Jarod. These guys "wrote the book", literally. Jim is our guest tomorrow and he'll be talking about system building, among other things. We always have a good time AND get stuff done on the Conference so come by and join us: Silently, on IRC: #voip-users-conference (freenode.net) - stop by any time, there'll usually be a party starting an hour or so before noon EDT Streamingly: http://www.voipusersconference.org/listen-now/ SIP in g722 wideband via ZipDX or g711 via Talkshoe (see the site http://tr.im/voip for instructions and URI) PSTN: (724) 444-7444 and enter 22622# PIN# - Get your PIN at Talkshoe.com or enter 1# instead The conference gives us all a chance to hobnob with celebrities like John Todd, find Asterisk people and groups in your neck of the woods and just generally usher out the week with a little light-sided view of some great technology. Thanks again to all of you who've been there and made it happen. For those of you who just getting in to Asterisk and VoIP, here's more about Jim: "Jim is probably a bit of a masochist, which would explain why he got into the telecom business in the first place, and why he now loves Asterisk. Jim is pretty friendly, kinda like a puppy that gets your shoes dirty. His enthusiasm is infectious, but also a little bit frightening if you stand too close. Jim is a partner in Core Telecom Innovations Inc, and iConverged Inc. He lives in Toronto with his wife and three kids, and loves writing, photography, speaking, improv, choral singing, and old shoes." About the shoes... /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum cable length for analog phone from FXS port
On Tue, May 26, 2009 at 5:31 PM, Danny Nicholas wrote: > I run my analog telco over cat5, but that's in-house and definitely not 3km. > That sounds really far for current loop stuff. I was doing that too. I asked this same question a few years ago and the answer was 100-200 meters. This is just a quick rule of thumb, but it seems about right. 3km, I doubt that would work, but it depends, as someone said, totally depending on ohm's law :) r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom Productivity Suite
On Thu, May 21, 2009 at 10:04 AM, Matt Darnell wrote: > 1. Set the phone to automatically record all calls to the USB stick, > now you have to press three keys. Not possible AFAIK. > 2. Put Record on the main screen when a call is active. This would > eliminate having to press the 'more' softkey. I wish Polycom would hire someone with ergonomics skills. The whole menu system is the most painful ever designed outside entry-level phones. Polycom is an acknowledged leader in sound quality and robust hardware but their idea of a menu sucks rocks and always has. Most of their menus require multiple click just to *exit* without doing anything. The 'x' (delete) button would do nicely with no additional cost. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Friday May 15 @12 Noon EDT with Askozia pbx
Hi, I met Michael Iedema of the Askozia pbx project at ANOOMA (ANOOMA is the Cannes Festival of the Asterisk world) and begged him to take time off from porting Askozia from FreeBSD to linux to tell us about his project and the problems of porting as well. Hope you'll all be able to stop by and lmisten or download or stream the recordings. We are having a lot of people call in on widenabd (g722) these days, and what better way to test your g722-capable phone than to call in via the ZipDX HD conference bridge? Check our IRC channel for the SIP URI to call. We also hope to get some on the spot news from Jeff Pulver's HD conference coming up next week. Hear ytou in a few hours. r/z ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMOOCON debriefing
Zoa, It was a pleasure to meet you! Please do come by some day, many people would like to talk about your work and your client! Does it do SIP URI? Call sip:7463#2262...@proxy.ideasip.com Best, Randy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A side of Digium you may have never seen
I caught Mark Spencer, Kevin Fleming, John Todd, Russell Bryant, "the other Mark" in a truly Digium moment in Rostock, Germany on their way to listen to the sea shanties. http://tr.im/rawhide - be afraid, be very afraid (Adhearsions' Jason Goecke is also in the picture somewhere) /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMOOCON debriefing
Anyone who was at AMOOCON and who would deign to join us (ahem, Zoa, alors?) to hash out what happened and make fun of the presenters, please join us Friday at 6PM Paris time (5 PM UK) or 12 Noon EDT. I myself was really pleased to be there and meet so many interesting and amusing people. Some recorded discussion is also posted here: http://sessions.voipUsersConference.org Join us in a few minutes and every Friday for more about VoIP asterisk and the price of fish. /r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Preferred language for Asterisk AGIs development ?
On Tue, May 5, 2009 at 9:31 PM, Steve Edwards wrote: > I doubt any language is going to replace any other language "for all > future developments." The day one religion replaces all other religions, it may happen because languages = religions = distros = platforms = your subjective belief structures. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
