Re: [asterisk-biz] POTS/FXO Gateway Alternatives
Brian, What was quirky about the GWX-4108? I was thinking of giving one a try. Did you come up with a better solution? Thanks, Bob On Tue, Aug 19, 2008 at 2:19 PM, Brian Franklin wrote: > Can anyone recommend an alternative to the Digium TDM FXO cards? I've > experience too many call quality and echo issues with production > systems. I've tried the Grandstream GWX-4108 but it is a bit quirky. I > prefer an external FXO gateway solution due to deployment flexibility. > > Brian > www.ntginc.net > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-biz mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-biz > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] The final Asterisk SIP Masterclass - Register now for beautiful Barcelona, Spain!
Olle, I've always wanted to get to one of your SIP Masterclasses. Will this course be tought in english? Bob On Thu, Apr 12, 2012 at 2:46 AM, Olle E. Johansson wrote: > Friends, > > I've been running the Asterisk SIP Masterclass for many years now. It's time > to run the last show - partly with new material. Compared with the very first > Asterisk SIP Masterclass I would say that I've rewritten 90% of the material. > That's what happens during the class. Students ask questions, you write a > slide. The word changes, you write a slide. You realize you've been wrong, > you delete or edit a slide. It's a moving target. > > The last one of these classes that I teach will be in Barcelona, Spain - June > 11th to June 15th. > > * Why the last one? > -- > Things change and you need to follow. During the last couple of years I've > been running many, many in-house trainings and workshops covering both > Asterisk, SIP in general and Kamailio. There seems to be more demand for > customized trainings that boost a team and help them move forward. I will > continue with these trainings, as well as try to come up with other trainings > that will run just a few times - more lab oriented possibly. > > * What is this class? > --- > From the sales material at http://www.avanzada7.com: > > "This class is focusing on building a scalable SIP realtime network. With a > combination of theory and practical labs, you will learn how to setup and > configure Asterisk and Kamailio - the Open Source SIP server - in a scalable > enterprise or service provider network. We will go through various kinds of > setups and give you insight in the design of real SIP networks with Asterisk > running in enterprise and service provider networks. The teacher Olle > Johansson, has many years of experience as an Asterisk developer as well as a > community member of Kamailio.org. By spending a week with Olle, you will get > a lot of insight into current and future features, bugs and implementation > details in a way that's hard to get otherwise. > Olle is a consultant working with architecture and implementation of large > scale communication platforms based on the SIP protocol. He has experience > from service providers, universities, call center platforms as well as > enterprise solutions. With experience of Unix and TCP/IP networking for over > 20 years, he has a lot of insight and knowledge, which he is using as a > teacher." > > The class is a five day high level class. You will meet not only myself, but > also other students that work with these tools and protocols, learn from them > and work together to solve issues in the labs. You need to have a basic > knowledge of Linux (how to start/stop applications, edit text files and build > applications) and Asterisk. This class is starting at a high level with > Asterisk. If you rather use FreeSwitch but want to learn Kamailio that is no > problem. You will just have to endure a few slides on Asterisk - but many of > the issues apply to FreeSwitch as well as other PBXs too. > > The cost is 3.200 Euro ex VAT. Companies outside of EU do not pay VAT as well > as companies in EU with a VAT registration number. > > If you have any questions or want to register, feel free to contact me > directly. > > Looking forward to seeing you in Barcelona! > > Best regards, > /Olle > > > > --- > o...@edvina.net - http://edvina.net > Open Unified Communication - building platforms with SIP and XMPP > From PBX to large scale implementations for carriers. Contact us today! > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-biz mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-biz -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
[asterisk-biz] Rhino Channel Bank
We have a Rhino Channel bank (CB24-24FXS) that we have taken out of production simply because we replaced it with a 24 port SIP ATA. If anyone is interested, let me know. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Polycom IP330 for sale take the lot at 40% discount
That's exactly what I'm thinking... I might be interested in 5 or 10 of them. On Wed, 2008-11-19 at 11:37 -0500, Steve Totaro wrote: > Provided they are new or very slightly used with power supply, I would > be willing to spit them with someone. > ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Polycom IP330 for sale take the lot at 40% discount
Do any of these come with power supplies? On Tue, 2008-11-18 at 22:04 -0600, jordan wrote: > I have 55 IP330's to get rid of. I have been asking $3500 US for these. > I need to get them written off before the end of the year. Make me a > deal, I might just take it! We are doing all transactions via paypal, we > ship out of Wisconsin. > > Jordan Novak > Communique LLC > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-biz mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-biz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] DIDs For Sale
Can you point us to a list of the rate centres that you offer? Bob On Wed, 2008-09-10 at 10:24 -0700, Jai Rangi wrote: > All, > I discussed about this offer with you all few weeks back and got a > very good sugestions from some experts. I do appreciate their input > and comments. > Finally we are able to come up with the right products and plans on > sip DIDs. I can offer you dids in 2600+ rate centers all over US. > Please see the price list and let me know if you are interested. We > can provide you DID numbers all over the US with no minimum > commitments. > > Number Rate > 1-10 $6.99 > 11-20 $6.50 > 21-50 $5.99 > 50-100 $5.50 > 100+ Contact US > > Some important feature are: > 1. More that 2500 rate centers all over US. > 2. Its all SIP, no need to configure TDM card, which makes things much > simpler. > 3. No Minimum requirement from Each rate center most of service > providers have. I know XO for sure have this limit. > 4. No Limit on Minutes. > 5. Each DID comes with Upto 20 Simultaneous channels. > 6. Perfect solutions for VoIP service providers, Call centers and > Calling Card applications. > 7. We can do Fail over/Load Balancing for your DIDs for Extra $1 for > each DID and Each extra IP. > 8. Work with any SIP Service (Asterisk, SER, OpenSer, FreeSwitch) > > If someone is looking for really Low pricing, I can also offer DID for > as low as $1.99 with the upto 4 channels. The maximum minutes usage on > this DID will be 3000 minute. There will be extra charge for .3 cents > per minute if you go over 3000 minutes and $1 for each extra channel. > > Please feel free to drop me a line if you think your needs are > different or if you are looking for something else. > Thank you, > > -Jai > jprangi @ > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-biz mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-biz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Canada DID origination and LNP.
I've had good success with LES.net Check them out here: http://les.net Bob On Wed, 2008-08-27 at 12:58 -0700, Nitzan Kon wrote: > Hello all! > > Anybody on here offers wholesale Canada LNP and DIDs (or knows > of a company that does)? > > Thanks! > > -- > Nitzan Kon, CEO > Future Nine Corporation > www.future-nine.com > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-biz mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-biz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Used IAXy devices...
How many do you have, and what price are you looking for? Bob On Tue, 2008-08-12 at 18:15 -0400, Michael S. White wrote: > Sorry for the SPAM > > Anyone out there interested in a few of these? - Just curious. > > Michael S. White > .e4 | 877-4E4-VoIP | 231-668-4285 DID | [EMAIL PROTECTED] > Positioned at the Intersection of Business and Strategy > www.8774e4VoIP.com > www.e4voip.com > > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-biz mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-biz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Videos of Asterisk-Tag.org online
On Sun, 2008-06-01 at 18:17 +0200, Stefan Wintermeyer wrote: > for those who haven't been in Berlin for the Asterisk-Tag.org we have > uploaded many talks (mostly the english once). You can find the videos > and the slides on > >http://www.asterisk-tag.org Thank you very much! I'm enjoying them right now. Bob Pierce Canada ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Fwd: CRI Crystal Recording interface new version
On Thu, 2008-04-03 at 18:01 +0300, Alexander Argov wrote: > The new version is faster, Admin can see call route, supporting GSM, > WEV, WEV compressed files. The demo on your site still says 1.06. Will this be updated soon so we can see the new functionality? Bob ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Friday Feb 29th Leap Year Special wih Aastra
On Thu, 2008-02-28 at 12:13 +0100, randulo wrote: > A PDF document accompanies this > presentation for those who wish to follow-along Powerpoint style. Where will the PDF be available? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Paranoia, Dell & 3CO
On Tue, 2008-02-05 at 10:18 -0500, Drew Gibson wrote: > Sorry to disagree, but RAID 1 would be far more responsible than RAID > 5. No need to apologize, Drew. I think you make some valid points here. I only wanted to state that I believe redundancy of disk and redundancy of PS are important. Bob ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Paranoia, Dell & 3CO
On Tue, 2008-02-05 at 08:55 -0500, Steve Totaro wrote: > I would go after RAID 5 and redundant power supplies. > I totally agree on the RAID 5 and redundant power supplies. It would be irresponsible not to get these. Bob ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Paranoia, Dell & 3COM
We are using Dell PowerConnect 3448P for voice&data switches with qos and separate vlans for data and voice. Bob On Mon, 2008-02-04 at 09:55 -0500, John Williams wrote: > Bob, > What are you using for a LAN switch? > Thanks! > > > On Feb 4, 2008 9:24 AM, Bob Pierce <[EMAIL PROTECTED]> wrote: > We are using a Dell 2950 with 2 Dual core 2GHz CPU, 2GB Ram > and a > Sangoma A104d-x quad T1/E1 card (which now seems to be called > the A104DE > on Sangoma's site). Using this system we are currently > handling 82 users > quite easily. We have had at least 23 concurrent calls on a > few > occasions with no problems. I'm pretty sure it could handle 48 > concurrent calls as the load on this box averages around 0.11 > and peaks > to about 0.4 > > -- > Bob Pierce > Network Analyst > Westman Communications Group > > > On Mon, 2008-02-04 at 09:04 -0500, John Williams wrote: > > Thanks for that wisdom, Bob. For this current job, we have > the > > luxury of running a separate network for voice, and can put > aside QOS > > (for now). > > > > Being completely paranoid, I want to buy equipment (switch > & > > PC) proven to operate an * PBX for 50-90 users and 48 > concurrent > > calls. > > > > -- Dell is our preferred PC vendor. Can anyone recommend, > based on > > actual experience, a Dell PC model for *, and 48 concurrent > calls? > > (The Dell recommended models on the Wiki are out of date) > > > > -- 3COM is our preferred basic switch vendor. Can anyone > recommend, > > based on actual experience, a 3COM switch model for *, and > 48 > > concurrent calls? > > > > Thanks in advance for aiding my considerable paranoia! > > > > > > On Feb 3, 2008 5:46 PM, bob murphy <[EMAIL PROTECTED]> > wrote: > > Once you start adding L3 to even L6 and 7 services > the party > > gets smaller and more expensive. At the point you > are > > building a network that requires QOS, Priority, Per > port > > VLAN's etc you may as well build with service > provider class > > switching gear like Foundry and Cisco etc. It will > help us > > all by eliminating complaints about "VOIP and > Asterisk" being > > not ready for prime deployment due to maturity > issues when in > > reality, it works fine on a well designed and > constructed > > network. Most of the time the Telephony system > get's blamed > > for what is actually a poorly designed network. > "Low cost" > > and "Business grade" may not coexist yet. So just > bite the > > bullet and use Foundry or Cisco. We have deployed a > couple > > systems with over 400 IP voice endpoints and things > are lookin > > good because of the proper L3 and QOS functionality > that was > > properly designed in to the final solution. > > > > Or, with cheap MAC switching you can just run > seperate > > networks. I mean, why complicate things by > converging voice > > and data. With L2 switching equipment so cheap. > But you'll > > have to run two cat6 drops everywhere. It's a trade > off. We > > have done it both ways. > > > > Bob > > Arreva Communications > > > > > > On 2/3/08, John Williams <[EMAIL PROTECTED]> > wrote: > > Deal List, Who are the low cost QOS LAN > switch > > vendors with products supported by * for > business > > grade voice service? Thanks > > > > > ___ > > --Bandwidth and Colocat
Re: [asterisk-biz] Paranoia, Dell & 3COM
We are using a Dell 2950 with 2 Dual core 2GHz CPU, 2GB Ram and a Sangoma A104d-x quad T1/E1 card (which now seems to be called the A104DE on Sangoma's site). Using this system we are currently handling 82 users quite easily. We have had at least 23 concurrent calls on a few occasions with no problems. I'm pretty sure it could handle 48 concurrent calls as the load on this box averages around 0.11 and peaks to about 0.4 -- Bob Pierce Network Analyst Westman Communications Group On Mon, 2008-02-04 at 09:04 -0500, John Williams wrote: > Thanks for that wisdom, Bob. For this current job, we have the > luxury of running a separate network for voice, and can put aside QOS > (for now). > > Being completely paranoid, I want to buy equipment (switch & > PC) proven to operate an * PBX for 50-90 users and 48 concurrent > calls. > > -- Dell is our preferred PC vendor. Can anyone recommend, based on > actual experience, a Dell PC model for *, and 48 concurrent calls? > (The Dell recommended models on the Wiki are out of date) > > -- 3COM is our preferred basic switch vendor. Can anyone recommend, > based on actual experience, a 3COM switch model for *, and 48 > concurrent calls? > > Thanks in advance for aiding my considerable paranoia! > > > On Feb 3, 2008 5:46 PM, bob murphy <[EMAIL PROTECTED]> wrote: > Once you start adding L3 to even L6 and 7 services the party > gets smaller and more expensive. At the point you are > building a network that requires QOS, Priority, Per port > VLAN's etc you may as well build with service provider class > switching gear like Foundry and Cisco etc. It will help us > all by eliminating complaints about "VOIP and Asterisk" being > not ready for prime deployment due to maturity issues when in > reality, it works fine on a well designed and constructed > network. Most of the time the Telephony system get's blamed > for what is actually a poorly designed network. "Low cost" > and "Business grade" may not coexist yet. So just bite the > bullet and use Foundry or Cisco. We have deployed a couple > systems with over 400 IP voice endpoints and things are lookin > good because of the proper L3 and QOS functionality that was > properly designed in to the final solution. > > Or, with cheap MAC switching you can just run seperate > networks. I mean, why complicate things by converging voice > and data. With L2 switching equipment so cheap. But you'll > have to run two cat6 drops everywhere. It's a trade off. We > have done it both ways. > > Bob > Arreva Communications > > > On 2/3/08, John Williams <[EMAIL PROTECTED]> wrote: > Deal List, Who are the low cost QOS LAN switch > vendors with products supported by * for business > grade voice service? Thanks > > ___ > --Bandwidth and Colocation Provided by > http://www.api-digital.com-- > > asterisk-biz mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-biz > > > > -- > Bob Murphy > Principal > > > Arreva Communications > www.arrevausa.com > > 949-334-2022-SIP Connect > 949-842-8450-Wireless > 949-349-0209-Fax > ___ > --Bandwidth and Colocation Provided by > http://www.api-digital.com-- > > asterisk-biz mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-biz > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-biz mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-biz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] IAX Channels limit on Asterisk 1.4.17
On Wed, 2008-01-16 at 11:26 +0100, Trixter aka Bret McDanel wrote: > These are some of the biggest reasons I personally avoid iax. I just > dont consider this type of bottlenecking to be 'carrier class'. It is > an inherent problem of using a single port for UDP (connectionless > really) communications. Trixter, Thanks for explaining these IAX issues so clearly. I had been tending to lean towards using IAX, but what you are saying makes perfect sense. Bob ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Austalia Route
Vijay, Your service sounds interesting. Do you support IAX2 trunking? Bob On Wed, 2007-10-10 at 08:12 -0400, Vijay Shan wrote: > Hi Jorge, > > VoIPInvite can provide you A-Z termination purchased off of multiple > Tier 1 carriers and to some destinations using our own PoPs. > > Please send an email to sales ((at)) voipinvite.com and we will send > you instructions to test. > > Thank you, > > Vijay > > > > VoIPInvite > 3888 Duke of York Blvd, Suite# 1424 > Mississauga, ON, L5B 4P5 > Canada > P 1 416 828 5262 > F 1 360 483 2170 > [EMAIL PROTECTED] > www.voipinvite.com > > -Original Message- > From: Jorge de Diego <[EMAIL PROTECTED]> > Sent: Wednesday, October 10, 2007 7:33am > To: Commercial and Business-Oriented Asterisk Discussion [EMAIL PROTECTED]> > Subject: [asterisk-biz] Austalia Route > > Hi, > > We are looking for one provider with SIP/IAX route to Australia with > CLI. > > Anyone have it? > > Thnks in advance > > > > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-biz mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-biz > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-biz mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-biz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] Canada 902 DIDs (Nova Scotia)
On Tue, 2007-08-21 at 13:44 +0100, Chris Bagnall wrote: > Does anyone out there have area 902 DIDs available that can be routed > via SIP or IAX? It looks like les.net has 902 DIDs for both Sydney and Halifax. They offer both SIP and IAX service. I'm using an IAX trunk from them for outbound calls, and it seems to be pretty good. Bob ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
Re: [asterisk-biz] FXS gateway
On Tue, 2007-07-31 at 18:51 -0400, Henry L.Coleman wrote: > Hi all, I'm playing around with the idea of using a couple of 24 port > FXS gateways and using analog phones for extensions. > Can anyone recommend a bullet proof device that works well with * and is > less than $70 per port ? > We're using the Rhino CB24-FXS. It's been working great. At about $1500 your looking at about $62.50 per port. Bob ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz
