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Old 2010-03-21, 06:31 PM   #781
Mango
 
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Call Waiting can be configured in your device. Look for "Call Waiting Serv" on the Line tab.
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Old 2010-03-21, 07:37 PM   #782
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I do have a fax machine I occasionally use, so I am curious at to what optimal settings I would need to put into the PAP2 to ensure the best possibly connection for success? Obviously the other post implies that premium routing would be required, which is no issue for me.
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Old 2010-03-21, 07:43 PM   #783
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Faxing with VoIP.ms is not guaranteed to be reliable because neither VoIP.ms nor the PAP2T support T.38. For low volume faxing, it might work, as Merve has discovered. Here are some good tips:

http://www.future-nine.com/faq/index...d=5&artlang=en

Note that they suggest disabling ECM. This can sometimes make faxes work with VoIP when they otherwise wouldn't. However, this has the added side effect of the fax machine sometimes reporting successful send when the send was not successful. If you must turn off ECM, then you should call the recipient of the fax to verify that they received it.

m.
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Old 2010-03-22, 09:18 AM   #784
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Yes premium routing makes a world of difference, I have set my ATA to strickly use g711u and my device supports t.38
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Old 2010-03-22, 04:17 PM   #785
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Default DISA feature

Voip.ms have added the DISA feature. It seems to work OK, but does not handle calling a number in my phone book (*75xx).
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Old 2010-03-22, 04:20 PM   #786
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You could setup an IVR to answer when you call. Then program short codes to access your numbers or DISA.
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Old 2010-03-22, 08:08 PM   #787
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voip.ms sure likes to add features, but i dont understand the point of disa?!?!?

does anyone think voip.ms will one day have the option for customers to list their name and number in major white pages?
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Old 2010-03-22, 09:05 PM   #788
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Quote:
Originally Posted by merve04 View Post
voip.ms sure likes to add features, but i dont understand the point of disa?!?!?
It's useful when you want to call some place but pay VoIP.ms rates. Hotels for example usually charge a lot for international calls. Now, I'll be able to call a VoIP.ms number and pay VoIP.ms rates instead. It will also be useful for people who have cell phones - a (better) alternative to the Callback feature.

Quote:
Originally Posted by merve04 View Post
does anyone think voip.ms will one day have the option for customers to list their name and number in major white pages?
Put in a ticket about that...I seem to remember that being possible though I've never actually done it. If you do it, please post back and let us know what was involved

m.
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Old 2010-03-23, 08:28 AM   #789
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Default Any experience in the flat-rate DID plan?

I currently am on the $1.49/mo and $0.0149 rate for outbound minutes.

It appears I could save a few dollars/mo by switching to $4.95/mo flat-rate with a restriction of two simultaneous calls.

My question is this: Is the two-call restriction for inbound calls only? If I have two inbound calls, could I still place a third/fourth outbound call?
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Old 2010-03-23, 09:02 AM   #790
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Yes, the restriction is for inbound calls only. It's intended for residential use, where someone might want to be speaking on the phone with one person and using Call Waiting or letting another person leave a voicemail.
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Old 2010-03-23, 02:58 PM   #791
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OK, thanks, Mango. It looks like I get cut my usage charge in half for just an extra $3.50 per month. Hard to beat that deal!

I don't suppose you have any insight on my URI question from a couples pages back, do you?
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Old 2010-03-23, 03:07 PM   #792
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How did you create jones@sip.voip.ms and bob.jones@sip.voip.ms? I thought that SIP URIs always started with numbers, with 11 + Accountcode + 3 digits of your choice for a total of 11 digits. No?

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Old 2010-03-23, 04:24 PM   #793
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Yes, but there is also an option to create URI addresses in the voip.ms portal.
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Old 2010-03-23, 04:29 PM   #794
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Are you talking about https://www.voip.ms/m/sipuri.php? If so, that won't create a SIP URI. That will allow you to route calls to an existing SIP URI.

To create a SIP URI, you either order it from the Order DIDs page, or use the default SIP URI for a subaccount.

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Old 2010-03-25, 10:44 PM   #795
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Mango,

I often experience echo when on my voip line. Is there any way that you may have found to eliminate that? I've tried several different settings, and nothing seems to work well.
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