Hello:
One of my customers needs data like 411xml.com and I recommended this
service, however they have not responded to our inquiries for sales
information. Does anyone know another service similar to 411XML?
Regards,
Andrew
--
On Thu, Aug 27, 2009 at 13:09, Wes Reece wreec...@gmail.com wrote:
Anybody have a provisioning guide for this phone? I have several hunder
that need to be configured by this weekend, but cant find the guide with
tftp provisioning instructions. My distributor is of no use either.
--
Thank
See if you can get the provider to do an express or force port of
the number. Cost will probably be approx $100 but it should get your
number back.
On Fri, May 22, 2009 at 18:53, Hermann Weckeherm...@wecke.com wrote:
On 5/22/2009 19:15, Nitzan Kon wrote:
If your number is with J2 you can pretty
On Mon, Feb 2, 2009 at 09:48, Rehan Allah Wala re...@supertec.com wrote:
Hi All,
Super Technologies (www.supertec.com) will be in the IT Expo, arriving at
Miami
Would love to meet you all, Do drop by if you are here.
Collect your DIDX ( www.didx.net) Coffee mugs and get pics with us, we
On Sat, Jan 3, 2009 at 13:09, Trixter aka Bret McDanel
trix...@0xdecafbad.com wrote:
On Sat, 2009-01-03 at 12:14 -0500, Andrew Joakimsen wrote:
Can you look at ticket # 702556000194?
This is very simple:
apparently it isnt.
Asterisk is down, I am simulating that with the command stop now
Can you look at ticket # 702556000194?
This is very simple:
Asterisk is down, I am simulating that with the command stop now,
Calls should then go to the failover SIP address, but they do not.
I have been back and forth for weeks with your support and they do not
figure it out. I am not even
I am looking for a VOIP provider that can offer origination and
provide the RDNIS with each call. I am not looking for any large
volume commitment.
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asterisk-biz mailing list
To
If I am not mistaken the Intel Core2Duo are 65watt at the minimum. AMD
has some 45W offerings in 2.3ghz and 2.5ghz both dual core (4450e
4850e). I have one sitting next to me right now actually running the
2.5ghz 4850e. I can not for the life of me get the CPU temp to go past
50 C even under full
On Wed, Oct 1, 2008 at 12:11 PM, Rehan Allah Wala [EMAIL PROTECTED] wrote:
Anrew,
You can browse all the inventory once you login to the web site, We do not
show numbers
without logging in though, but you can view the cities available from
www.didx.net/did/
Rehan
I I login and go to
On Tue, Sep 30, 2008 at 2:09 PM, Peter Beckman [EMAIL PROTECTED] wrote:
DIDx: API, Full DID (eww)
DIDX, for a while already, no longer lets you browse the full
inventory, at least through their website.
___
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I'm not trying to insult anyone but if you are sending out mailings
you should be providing an unsubscribe link or at least simple
instructions (such as reply with unsubscribe in the subject to
unsubscribe from our mailings)
Requiring people to login to an account is annoying. Click, confirm,
The stuff on the Digium site is like an EULA... most of the provisions
are not enforceable.
Just as I can get a BMW engine, put it in a Yugo and sell it in the
free market as a Yugo with a BMW engine you can do the same with
Asterisk. Look into fair use if you create a product that works with
for no reason, and
charge the wrong credit card...lovely! It is not this hard to run a
business!
On Feb 11, 2008 8:04 PM, Andrew Joakimsen [EMAIL PROTECTED] wrote:
After I became frustrated with VoIP Supply (6 months to RMA) I
started using Telephony Depot. No complaints.
On Feb
After I became frustrated with VoIP Supply (6 months to RMA) I
started using Telephony Depot. No complaints.
On Feb 11, 2008 10:32 AM, Ron McCarthy [EMAIL PROTECTED] wrote:
Hello,
I am looking for any Grandstream resellers who can provide good pricing and
fast in stock shipping. I was
Can someone recommend any provider that can provide US SMS DID?
Preferably nationwide?
On Jan 17, 2008 8:50 PM, Talking Voice [EMAIL PROTECTED] wrote:
Hi!
Can someone recommend a cheaper SMS provider..
On Jan 10, 2008 1:26 PM, M. Emran [EMAIL PROTECTED] wrote:
you can check
for a detailed quotation.
Warmest regards,
Andrew Joakimsen
2008/1/17 [EMAIL PROTECTED]:
hello everyone,
I have two asterisk pbx , server A with a WAN IP,and server B with a LAN IP.
Server B connect to Service Provider through a E1 trunk and users at server B
can make PSTN calls.I want users
So basically Google just doesn't want to be the judge to determine
what is fair use and what isn't?
Is the EU/US distinction based on the searcher location, the
advertiser location or what region site the user visits (or a
combination of all three)?
On Jan 18, 2008 1:59 PM, Eric Chamberlain
Ok.. every phone has DND..
A rather interesting bug I found out in a rather interesting way on
the GXP-2000 I wonder if you could see if its fixed on the newer
phones: .If you dial a call and press mute while the call is ringing
the phone indicates MUTE on the display, however the call is not
Rugged cordless phones for large scale deployment? Try SpectraLink
On Jan 17, 2008 12:27 PM, John Goerzen [EMAIL PROTECTED] wrote:
Hi folks,
We're looking for wifi SIP phones to use with our Asterisk PBX in a
manufacturing environment. We are presently using the Hitachi ones.
While they are
Do you think the quality is improving on the Grandstream phones, say
compared to a GXP-2000 (I think it's a fine phone and always said if
they made a $100+ it would probably be top-notch). I think the main
issues on the 2000 is the keypads can be a little firmer/less cheap
feeling the display
Asterisk is a product of Digium... yes.. But it is fair use to say
you are selling something that works with asterisk.
Its like when you go to an aftermarket autoparts place and buy a part
for a BMW... the part is not made by BMW but it is compatible with
BMW so thats why they can use the
message --
From: David Aldworth [EMAIL PROTECTED]
Date: Oct 2, 2007 1:01 PM
Subject: RE: [asterisk-biz] Need LNP 352-416-, 352-672, 352-505 ASAP
To: Andrew Joakimsen [EMAIL PROTECTED]
Cc: Geoff Love [EMAIL PROTECTED]
Hi Andrew -
We would be happy to facility this. T.38 on inbound
What did you expect?
On Jan 10, 2008 2:08 PM, Andres Paglayan [EMAIL PROTECTED] wrote:
Hi
Looks like all (or most) of DIDs through Teliax are down,
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asterisk-biz mailing
No.. because I never requested any information about termination.
It seems to me if you were asking about a service and someone responds
with some sort of offer the message was solicited which is contrary
to the definition of SPAM.
On Dec 30, 2007 3:50 PM, C F [EMAIL PROTECTED] wrote:
Anybody
On 9/26/07, Miles Keaton [EMAIL PROTECTED] wrote:
- Each work-at-home person should only need a headset + softphone
software on their PC, right? (Is the software audio quality as good
as separate hardware SIP phones now, if run on a modern-speed PC?
Recommendation for best?)
When you say
Hi:
I am looking for a provider that can port numbers in the above
prefixes. They are all in the GAINESVL, FLORIDA ratecenter. Don't care
how its delivered, IAX, SIP, Ulaw, G723. It would be great if you can
support T.38 faxing but if you don't thats fine.
I need a guaranteed date the LNP will
On 9/5/07, Mike Hammett [EMAIL PROTECTED] wrote:
Does anyone have recommendations for T.38 fax machines? The ones I've found
by Oki and Ricoh were quite expensive.
There is no T.38 support in Asterisk and no plans to add it. So even
if there was a T.38 fax machine -- what's the point?
On 8/22/07, Seysan [EMAIL PROTECTED] wrote:
Hi Folks,
about 1 week ago I asked a question about using EXTERNAL Fax/Modems (RS-232)
that can be used as an FXO or FXS.
But I couldn't get to a clear answer yet!
No, that is not possible.
Has anyone used EXTERNAL Fax/Modems to SEND/Recv FAXES?
On 8/6/07, Matthew Rubenstein [EMAIL PROTECTED] wrote:
According to Xorcom's press release advertising their reliable
Asterisk
Fax/ZAP solution
( http://www.xorcom.com/documentation/white_papers/fax_modem ), the
problem Asterisk has with mediating faxes is Zaptel timer desync
Take a look at www.911enable.com I've had very good experiences with them.
On 8/4/07, Ivan Kovacevic [EMAIL PROTECTED] wrote:
Hi Everyone,
Can anyone recommend a reliable E-911 provider in Canada?
Thanks,
Ivan
___
--Bandwidth and
Jeff:
We can do the Linksys products and are an authorized Linksys reseller so we
can provide you all the provisioning docs as well. We don't have SNOM but
have you looked at the Linksys SPA series of phones? Take a look here:
On 7/17/07, lists [EMAIL PROTECTED] wrote:
Does tis mean vtwhite and viatalk are going away too since Sun Rocket
supposedly owned them?
No. ViaTalk is a registered trademark of HostRocket LLC
___
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On 7/12/07, randulo [EMAIL PROTECTED] wrote:
any comments?
They were probably mad of the Astri name... You know how Digium
can be when it comes to the use of their names such as Asterisk.
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On 7/13/07, W. Glenn Campbell [EMAIL PROTECTED] wrote:
Polycom (In my opinion) has the best quality.
Polycom have great sound quality and I would say Aastra are very close
as well. The only problem with these two phones is their poor NAT
support. Polycom just doesnt really have it (maybe their
sip debug is your friend.
On 7/3/07, Ryan M. Colbert [EMAIL PROTECTED] wrote:
This may be the wrong forum for my question, and if so, please forgive my
error.
I have been working on setting up a SIP trunk from Bandwidth.com for
almost a week now. Outgoing calls are working fine but I can't
service works better than there web site :)
Best regards,
Al Bochter
Bochter Services
Did you check your US Greenbacks for GOLD Today?
http://www.bochterservices.com/?t=USbill_email
Andrew Joakimsen wrote:
Does anyone know of a decent E-911 provider? I would use Intrado or
similar
On 5/19/07, Luki [EMAIL PROTECTED] wrote:
There's http://spc.pifiu.com but for the SPA-phone the program is
just a bit old, it does work but the phone constantly will download
the config because some parameter is not right.
Double check your parameters then. We provision 921/922/941/942
Audiocodes blatantly violates the GPL under which key components of
the software that runs their product is licensed to them. So there's
no way an Audiocodes product can be considered fine or acceptable in
any way.
On 5/8/07, Gregory Boehnlein [EMAIL PROTECTED] wrote:
Gregory Boehnlein wrote:
On 5/16/07, C F [EMAIL PROTECTED] wrote:
1) They will not officially give out the provision details.
What do you mean by that? I have them provisioned using http and it works great.
Well none of the sample configfiles from the internet every work for
me. Linksys has an spc.exe program it
Does anyone know of a decent E-911 provider? I would use Intrado or
similar but the problem is we need just a low-volume solution.
I posted this a while back and a man by the name of Paul Watts with
a company called US-Nap (www.us-nap.com) contacted me and has done
nothing but waste my time for
Yes its a scam. Couldnt even get someone to answer the phone or an
email BEFORE trying to order service.
On 5/19/07, Oli Gunnarr Håkansson [EMAIL PROTECTED] wrote:
I'm wondering about this site, trouble tickets are not responded to and
information on this website seems not to have been updated
SprintPCS phones with the ReadyLink feature have a built in SIP
client, but I couldnt tell you if there is a way you could use it.
On 5/2/07, Begumisa Gerald M [EMAIL PROTECTED] wrote:
Hi,
Is anyone aware of a device capable of talking SIP over a CDMA phone
network - apart from the combination
On 4/10/07, Glenn Campbell [EMAIL PROTECTED] wrote:
Hello,
I have a customer that would like to know what were good resources for phone
lists to insert into a predictive dialer system. What they are looking for
is a list that has been DNC scrubbed and also able to search for lists based
on
Is there any PSTN provider in the world that can offer you say a PRI
which uses G722
On 4/9/07, Guilherme Góes [EMAIL PROTECTED] wrote:
Actually the CODEC name is ITU-T G.722.2 which is identical to the
GSM-AMR Wideband CODEC. The HD part of the CODEC is its sampling rate
at 16 kHz, where the
with IP T1 and VoIP services and can assure you the
QoS between the T1 and us for VoIP. If you wish to have a quote,
please email me your address and the main BTN at that location.
Andrew Joakimsen
[EMAIL PROTECTED]
___
--Bandwidth and Colocation
I have seen with DIDx issues where sometimes a DID will just stop
working, maybe even someone else answers it. Its not often but its
just very odd.
Also they take too long to respond to any questions. Their billing
system is very odd. Say you have 5,000 DID and have fully paid for
those service,
I dont see that you would have issues but I would suggest that you
dont use Fedora, much less version 4.
On 4/4/07, Michael Munger [EMAIL PROTECTED] wrote:
I was thinking about using the following intel board for some Asterisk
installations with FC4, has anyone had any issues with it? Anyone
You can download the schematics for tormenta boards at zapatatelephony.org
The main issue I see is as follows:
1) The design is not correctly licensed under the GPL license
2) I do not believe that the GPL can be applied to hardware
3) You cannot copyright a circuit.
So I really see no
90% of the NANPA ratecetners are the city name. I think NYC is the
largest part of the list you'll find without it.
On 3/31/07, Linda Goldsmith [EMAIL PROTECTED] wrote:
Thank You,
I Need the Rate center Names and corrsponding City and State Names also.
Date sent: Sat, 31 Mar 2007
-biz-
[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen
Sent: Tuesday, March 27, 2007 4:39 PM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [asterisk-biz] VoIPSupply Q1 Clearance
The quality of service does show. A few weeks to answer email... HALF
A YEAR TO PROCESS AN RMA
The quality of service does show. A few weeks to answer email... HALF
A YEAR TO PROCESS AN RMA.
On 3/22/07, Hermann Wecke [EMAIL PROTECTED] wrote:
Cory Andrews wrote:
Cory Andrews
voice direct - 716.250.3402
Does anyone knows if Cory is still alive - or if he is and/or if
Voipsupply is out
Look here: http://www.askcalea.net/docs/20030129.ssd.cgvop.pdf
Most of the events needed are there, what is needed more is a system
to actually log events by subscriber. Realistically it should be very
easy to implement if you are familiar with asterisk. Hell hack a bit
off the cdr modules, a
Yep so a 10 second calll at 1/2 cent/minute actually costs 1 cent.
That's going to be the least of your worries with VoicePulse. Forbid
that you have a servive issues, they will tell you to install Wine and
run Win32 programs on your asterisk machines.
On 3/1/07, Matt Dunkin [EMAIL PROTECTED]
Since you have the best Cuba route I assume you can offer some assured
call completion ratio? Or at least tell me what that has been in the
past?
On 3/1/07, Lee Wasser [EMAIL PROTECTED] wrote:
All,
We have a TIER 1 Cuba route. This is the best quality out there - 72
cents / min.
This is more like freepbx configuration question
Then you are on the totally wrong mailing list, perhaps FreePBX has a
mailing list where you can discuss their software? I'm sure their
homepage would contain those details, you can find their homepage if
you search FreePBX on a search engine,
You wan to use a keypress other than pound to exit voicemail in
order to retain the ability to exit voicemail by pressing pound.
I think you answered your own question, no changes are needed to
retain the current funcinality.
Perhaps you can post your question on the correct list next time
Yes, but it was originally T-Mobile (Omnipoint). It think it depends
more on your area than anything, if the provider can LNP a number from
the ratecenter it shouldnt be an issue who the carrier is.
On 11/13/06, Peter Beckman [EMAIL PROTECTED] wrote:
On Fri, 10 Nov 2006, Peter Beckman wrote:
Horrible horrible horrible support
On 12/27/06, Mark C [EMAIL PROTECTED] wrote:
I was wondering what peoples experiences with www.DiDx.net. I have been
thinking about offering Hawaii DIDs (AC 808) via the exchange. Good
Idea? Bad Idea?
Your input would be greatly appreciated.
Mark C.
I'm going to have to take back all the nice things I said about Isphone,
they are total crap.
On 8/29/06, Andrew Joakimsen [EMAIL PROTECTED] wrote:
So you want another company to manage the TDM - SIP conversion but you
still want to be in charge of the trunks.
I think probably your best bet
Well it must be your setup because I have used the three phones you mention
and each one pretty much worked out of the box with no issues, no problems
with NAT. The GXP-2000's aren't that bad, their provisioning is pretty
straightforward audio quality seems fine, unlike many phones you can
Yes, I have I am wondering if you have you thought of the potential
liability if something did not function right.
You would probably need a server to pool all the access to the 911 network,
who would maintain that?
On 11/22/06, Marnus van Niekerk [EMAIL PROTECTED] wrote:
Anybody ever
Yes it can, they even have examples on their web site.On 11/14/06, ismail loo [EMAIL PROTECTED] wrote:
It seems trxtel's service can't accept iax2 call.
One communications has a similar service,
URL please?
2006/11/15, Mike Hammett [EMAIL PROTECTED]:
One communications has a similar
If anyone can get some Cuba DIDs I would like to know as wellOn 11/12/06, JOAO CARLOS MOURA [EMAIL PROTECTED]
wrote:
Hi
i need DIDs for
Honduras
Guatemala
Nicaragua
El Salvador
Cuba
Thank's
João Carlos Moura[EMAIL PROTECTED]
NINETEL TELCO,Inc7382 NW 35 Terrace (33122)
Miami - FL -
I've thought of the same thing in the past and would be open to discuss how it would work. Please contact me off list if you have serious interests.On 11/10/06,
Bart Fisher [EMAIL PROTECTED] wrote:
If you have ever wanted to get into the VOIP Biz, but found thatCompanies that offer such
You should have read that link you posted.Some people confuse pyramid and Ponzi schemes with legitimate multilevel marketing.
Multilevel marketing programs are known as MLM's,(4) and
unlike pyramid or Ponzi schemes, MLM's have a real product to sell. More importantly,
MLM's actually
First you need a team that is highly familiar with VoIP, Asterisk, SER, CCM, SIP, IAX, H323, etc. That's not something a few posts on a mailing list or even reading some books can give you. You need people who are very familiar with these technologies and have hands on experience. If you are
Sorry for posting this! I overlooked this being posted on the -biz listEhsan: My consulting services are avaliable at a rate of $125/hour if you need further assistance with your business, please contact me off-list.
On 11/2/06, Andrew Joakimsen [EMAIL PROTECTED] wrote:
First you need a team
All the standalone WiFi phones I have used are crap, and all the ones I have wanted to use others have told me are crap.However, I've never heard bad things about SpectraLink products
On 10/31/06, Alex [EMAIL PROTECTED] wrote:
Steve
The best solution will be a Dect phone
]] On Behalf Of
Andrew Joakimsen
Sent: Tuesday, 31 October 2006
12:48 PM
To: Commercial
and Business-Oriented Asterisk Discussion
Subject: Re: [asterisk-biz]
Wireless SIP Solution
All the standalone WiFi
phones I have used are crap, and all the ones I have wanted to use others have
told me are crap
it is my understanding from about two years ago that they do support SIP and the information on their website seems to suggest the same: http://www.spectralink.com/resources/faq.jsp#netlink6
However, I've only worked with their older 900mhz gear.On 10/31/06, Cory Andrews [EMAIL PROTECTED]
wrote:
I've used ISPhone for a while and no complaints here. Their website is www.isphone.net They charge us $50 per toll free number for setup, but besides that the rates are competetive.
On 10/27/06, George Masgras [EMAIL PROTECTED] wrote:
Hello everyone, I've been having a lot of headaches with my
Looking for numbers in the 352 area code ratecenter Gainesville. You must be able to assign us new numbers and be able to port 352-416 numbers.
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To UNSUBSCRIBE or update
Just wanted to give you guys a heads-up. I usally don't cross post but thought this is an important warning in case anyone was considering to use VoicePulse service.-- Forwarded message --From: Andrew Joakimsen
[EMAIL PROTECTED]Date: Oct 22, 2006 4:28 PMSubject: Re: [asterisk
I thought they were using Asterisk..
They are still down No messages may be taken for this mailbox
ring ring No messages may be taken for this mailbox and it goes on
forever.
On 10/16/06, C F [EMAIL PROTECTED] wrote:
BTW, Last time I called them they weren't using Asterisk, IIRC it was
What sort of volume are you expecting? What prices can you offer for
lower volumes?
On 10/4/06, Vijay Shan [EMAIL PROTECTED] wrote:
Mitul,
Conditions mean: volume or potential volume which we will determine after
reviewing the client and the application the trunk is used for.
Thank you,
And like it or not politics and business have a very close relationship.
On 9/28/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I thought this group is about Asterisk Business.
Just for the record, and what are YOU doing now?
And what is this:
What about mobile/PCS phones? Usually the carriers DO provide the data
and it is delivered to RBOC PSTN users but it seems someone along the
line does not do the PROPER lookups, be it for whatever reason.
How close to RBOC PSTN CNAM can you get?
On 9/8/06, Gregory Giagnocavo [EMAIL PROTECTED]
- for CallerID
To: Andrew Joakimsen [EMAIL PROTECTED]
The email address [EMAIL PROTECTED]
is protected by EarthLink spamBlocker. Your
email message appeared to be spam, and has been redirected to a
suspect email folder and Gregory Giagnocavo hasn't seen your email
yet.
In order for your message
Lookup sample.call what we do is have a PHP script that generates
those files for us. It is very flexable, you could for example run
it on the asterisk server and have your webservers post the value to
it.
On 9/8/06, Francesco Basili [EMAIL PROTECTED] wrote:
Hi,
does exist any open
So you are saying Digium has never distributed G723 and/or G723.1
codec in any form?
Best regards,
Andrew Joakimsen
On 9/7/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
- Andrew Joakimsen [EMAIL PROTECTED] wrote:
What about the G723 code that used to be on the Digium CVS server
Kevin:What about the G723 code that used to be on the Digium CVS server?Best regards,Andrew JoakimsenOn 9/6/06, Kevin P. Fleming
[EMAIL PROTECTED] wrote:On September 4, 2006 an anonymous poster sent a message to these mailing lists containing a link to a package of source code claiming that it
I don't know about G729, but you used to be able to get G.723 from the Digium CVS site! cvs get g723. I just looked at the makefiles from 1.2.9.1 and they still have the g723 stuff in them FWIW
.On 9/4/06, Kannaiyan Natesan [EMAIL PROTECTED] wrote:
Hey,Is this code released by Digium?Looks like
So even if we license from Intel the code, it is illegal to use it with Asterisk because Asterisk is GPL? I still don't get that partOn 9/4/06, Justin Newman
[EMAIL PROTECTED] wrote:
Kannaiyan,It may be helpful to
Try globalsources.com, I found a bunchOn 9/2/06, Go4Calls [EMAIL PROTECTED]
wrote:Hi All,I am looking two port IAX ATA similler to Linksys PAP2.
Could you please give some URLS where i can find this kind of ATA?Regards, __Do You Yahoo!?Tired of
http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/On 9/2/06, Kannaiyan Natesan
[EMAIL PROTECTED] wrote:STOP !!
I'm least bothered whether g729 works or not or what the developer didto made it to work.I'm bothered how it works and what are the details about it. As youare
Then if you control said DID number by defention any use of that DID as CID/ANI is NOT spoofing. Spoofing would be using something you don't have and IMO there is no valid reason for spoofing.Any method of caller identification SHOULD NOT rely on the Caller ID or ANI because we all know they are
So you want another company to manage the TDM - SIP conversion but you still want to be in charge of the trunks.I think probably your best bet is to find a managed co-location provider -- you obtain the TDM trunks and they will manage the servers and SIP or whatnot. I am sure such a thing exsists
You can publish whatever you want with whatever terms you want. Just remember if you USE GPL code in your code then read the GPL and understand what the requirements are.If you want the possibility of having your patch applied/included with Asterisk then of course you need to meet Digium's
Does anyone know of a system that will allows us to accept tickets as email from the customer, and also allows our CSRs to input calls and then send the customer an email when the issue is updated?
On 8/27/06, Bob Smith [EMAIL PROTECTED] wrote:
what was the one with all in one.. SIP callback +
Bellsouth filters outgoing (and probably inbound) port 25, which is for sending mail as a measure to reduce spam I think its BS, but you can get around by having your mailhost use port 587 which is the standard for mail submission.
Honestly, Comcast has the worst service and they are total
Where in the US can we buy them?On 8/23/06, GlobalOfficePhone [EMAIL PROTECTED] wrote:
What is the price in the US?
On 8/23/06, Zoa [EMAIL PROTECTED]
wrote:
They are pretty good, very good price quality ratio.Brian Franklin wrote:Anyone ever use Thomson VoIP sets?They look nice but how do they
Please see http://www.dslreports.com/forum/remark,15941943 -- a thread
titled Nufone shut down -- all toll free disconnected! for further
refrences regarding NuFone.
They fired us when we were complaining (asking them to fix)
gmabout an issue and the message was somethign along the lines of it
Any business. For the smaller ones 5 employess they are probably
looking for a hosted solution Remember the Bells charge after
taxes and such at least $40/month for a business phone line and that
includes no features and calling to just the local area. With a hosted
VoIP solution you can
1) This is the incorrect forum to ask for that information. The
correct one is the regular asterisk mailing list. This is
asterisk-biz for business related discussions of asterisk users.
2) The way caller id name works is the number is transmitted with the
call and at some point a lookup is done
On your site it says that the cost for LNP is $15 Is that per
month? Per year? One time fee?
What is the exact pricing for a DID that was ported? Setup fee? Per
month fee? Included minutes? Per minute fee? Minimums? Term
agreements?
On 8/14/06, Linda Goldsmith [EMAIL PROTECTED] wrote:
Hello
You should contact UPS and inform them of the fradulent use of your
account, I'm almost sure they cannot charge you for a package that was
not shipped by you.
What this has to do with Asterisk or any sort of VoIP is beyond me...
On 8/9/06, Bob Smith [EMAIL PROTECTED] wrote:
Garret , we got a
Keep in mind, that at least in the United States, if you are providing
Asterisk PBX or using Asterisk PBX it could be considered fair use
to promote that fact. If you were selling Digium PCI cards there would
be nothing stopping you from marketing them as Digium PCI cards
because that's what they
Again, at least in the United States, what Digium says doesn't
necisarrily have to be the truth It is what Digium WANTS, but you
might have other rights granted to you by law, namely fair use.
Here's one site that outlines the basics:
http://www.publaw.com/fairusetrade.html
The Lanham Act
In my ATA there is an option Suppress voice packets during RFC2833
Telephone Event packet transmission Enabling this option fixes the
dual DTMF issue.
If you are using G711 codec you could also disable RFC2833/SIP INFO
DTMF which would cause asterisk to do the detection.
Basically the issue is
I have noticed the same issue on SIP ATA --Asterisk SIP provider.
Also noticed, when I set a voicemail recording and press # at the end,
the DTMF for the # is recorded -- this did not happen before.
ATA and SIP.conf are set for DTMF RFC2833.
Also this is low latency, maybe 35 ms between
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