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#781 |
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Join Date: Nov 2008
Location: Alberta
Posts: 797
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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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#782 |
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Join Date: Aug 2006
Location: Ottawa, ON
Posts: 10
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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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#783 |
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Join Date: Nov 2008
Location: Alberta
Posts: 797
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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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#784 |
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Join Date: Jul 2007
Location: Edmonton, AB
Posts: 967
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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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Panasonic PT-AE4000|Pioneer VSX-1021K|Soundstage STAGE3D4, STAGE3D2C, STAGE3D2, Velodyne VDR-12|Sony BDP-S350|WDTV Live Plus|Telus CIS430|Harmony One LG 55LE5500|H\K AVR 347|Yamaha NS-8390|Polk Audio PSW-10|Samsung BD-P1600|WDTV Live|Telus CIS330|Harmony 600 LG 42LC7D|ASUS O'Play|Telus CIS330|Harmony 600 LG 32LS3400|Telus CIS330 |
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#785 |
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Rookie
Join Date: Nov 2009
Posts: 3
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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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#786 |
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Join Date: May 2006
Location: Gatineau, QC
Posts: 1,165
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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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#787 |
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Join Date: Jul 2007
Location: Edmonton, AB
Posts: 967
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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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Panasonic PT-AE4000|Pioneer VSX-1021K|Soundstage STAGE3D4, STAGE3D2C, STAGE3D2, Velodyne VDR-12|Sony BDP-S350|WDTV Live Plus|Telus CIS430|Harmony One LG 55LE5500|H\K AVR 347|Yamaha NS-8390|Polk Audio PSW-10|Samsung BD-P1600|WDTV Live|Telus CIS330|Harmony 600 LG 42LC7D|ASUS O'Play|Telus CIS330|Harmony 600 LG 32LS3400|Telus CIS330 |
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#788 | ||
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Join Date: Nov 2008
Location: Alberta
Posts: 797
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Quote:
Quote:
m.
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#789 |
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Veteran
Join Date: Jun 2007
Location: /dev/null
Posts: 2,686
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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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#790 |
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Join Date: Nov 2008
Location: Alberta
Posts: 797
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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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#791 |
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Veteran
Join Date: Jun 2007
Location: /dev/null
Posts: 2,686
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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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#792 |
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Join Date: Nov 2008
Location: Alberta
Posts: 797
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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?
m.
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#793 |
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Veteran
Join Date: Jun 2007
Location: /dev/null
Posts: 2,686
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Yes, but there is also an option to create URI addresses in the voip.ms portal.
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#794 |
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Join Date: Nov 2008
Location: Alberta
Posts: 797
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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. m.
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#795 |
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Join Date: May 2006
Location: Gatineau, QC
Posts: 1,165
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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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