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<title>Topic &#x27;Re: Alternative to voip.ms ?&#x27; in forum &#x27;VOIP Tech Chat&#x27; - dslreports.com</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27870072</link>
<description></description>
<language>en</language>
<pubDate>Sat, 25 May 2013 20:00:51 EDT</pubDate>
<lastBuildDate>Sat, 25 May 2013 20:00:51 EDT</lastBuildDate>

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<title>Re: SIP forwarding</title>
<link>http://www.dslreports.com/forum/Re-SIP-forwarding-27884820</link>
<description><![CDATA[SCADAGeo posted : <div class="bquote"><said>said by <a href="/profile/1572525" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=1572525');">PX Eliezer</a>:</said><p>If it is done by creating a Virtual SIP number and using that as an inbound SIP, there is a nominal fee:<br><div class="bquote"><p>Monthly fee is $0.25 (25 cents) / month per active Virtual SIP number.  Incoming calls are billed $0.001 (one tenth of a cent) per minute.<br></p></div>&raquo;<A HREF="https://www.voip.ms/m/virtualnumbers.php" >www.voip.ms/m/virtualnumbers.php</A><br><br>HOWEVER, a method that is quite similar (using a subaccount to make an internal extension, and using that internal extension as an inbound SIP, appears to be [free].  Their website PREVIOUSLY talked about a charge of one-tenth of a cent per minute for THIS method, but now the language is gone.<br>&raquo;<A HREF="https://www.voip.ms/m/subaccount.php" >www.voip.ms/m/subaccount.php</A><br> </p></div>I used sub-accounts for testing, and I wasn't aware of the first method.  Thanks for the clarification. :)]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/Re-SIP-forwarding-27884820</guid>
<pubDate>Sun, 06 Jan 2013 01:27:44 EDT</pubDate>
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<title>Re: SIP forwarding</title>
<link>http://www.dslreports.com/forum/Re-SIP-forwarding-27884585</link>
<description><![CDATA[PX Eliezer posted : <div class="bquote"><said>said by <a href="/profile/1847958" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=1847958');">SCADAGeo</a>:</said><p>When I tested VoIP.MS's inbound SIP, I did it without funding my account, so it appears as if it the inbound SIP calls are free - don't know if this is still true...<br><br> </p></div>If it is done by creating a Virtual SIP number and using that as an inbound SIP, there is a nominal fee:<br><div class="bquote"><p>Monthly fee is $0.25 (25 cents) / month per active Virtual SIP number.  Incoming calls are billed $0.001 (one tenth of a cent) per minute.<br></p></div>&raquo;<A HREF="https://www.voip.ms/m/virtualnumbers.php" >www.voip.ms/m/virtualnumbers.php</A><br><br>HOWEVER, a method that is quite similar (using a subaccount to make an internal extension, and using that internal extension as an inbound SIP, appears to be [free].  Their website PREVIOUSLY talked about a charge of one-tenth of a cent per minute for THIS method, but now the language is gone.<br>&raquo;<A HREF="https://www.voip.ms/m/subaccount.php" >www.voip.ms/m/subaccount.php</A>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/Re-SIP-forwarding-27884585</guid>
<pubDate>Sat, 05 Jan 2013 22:55:08 EDT</pubDate>
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<title>Re: SIP forwarding</title>
<link>http://www.dslreports.com/forum/Re-SIP-forwarding-27884455</link>
<description><![CDATA[SCADAGeo posted : <div class="bquote"><said>said by <a href="/profile/1572525" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=1572525');">PX Eliezer</a>:</said><p>There CAN be small Voip.MS charges for [inbound] SIP calling.  <br>&raquo;<A HREF="/forum/r21533299-What-is-the-voip.ms-sip-url-">What is the voip.ms sip url?</A><br> </p></div>When I tested VoIP.MS's inbound SIP, I did it without funding my account, so it appears as if it the inbound SIP calls are free - don't know if this is still true...<br><br>Outbound calls, even to extensions on sub-accounts, requires a funded account, but there is no cost for the call.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/Re-SIP-forwarding-27884455</guid>
<pubDate>Sat, 05 Jan 2013 21:54:51 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27881715</link>
<description><![CDATA[Arne Bolen posted : <div class="bquote"><said>said by <a href="/profile/1805984" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=1805984');">ConstantineM</a>:</said><p>P.S. It's surprising noone ever mentions pbxes.org on this forum.  I wish I would have found them earlier. </p></div><A HREF="http://www.pbxes.org">PBXes</A> has often been mentioned on this forum by many members including myself. ;)<br><small>--<br><A HREF="http://myvoipnews.com/tagged/Anveo-Support">My VoIP News</a></small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/Re-Alternative-to-voipms-27881715</guid>
<pubDate>Fri, 04 Jan 2013 20:39:51 EDT</pubDate>
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<title>Re: SIP forwarding</title>
<link>http://www.dslreports.com/forum/Re-SIP-forwarding-27875414</link>
<description><![CDATA[PX Eliezer posted : <div class="bquote"><said>said by <a href="/profile/755055" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=755055');">OZO</a>:</said><p>"SIP URI" routing is billed using usual incoming rates and not free. <br> </p></div>No, it is billed at a [very small] charge.<br><br>cf:<br>&raquo;<A HREF="/forum/r21533299-What-is-the-voip.ms-sip-url-">What is the voip.ms sip url?</A>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/Re-SIP-forwarding-27875414</guid>
<pubDate>Wed, 02 Jan 2013 22:14:38 EDT</pubDate>
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<title>Re: SIP forwarding</title>
<link>http://www.dslreports.com/forum/Re-SIP-forwarding-27875391</link>
<description><![CDATA[grand total posted : <div class="bquote"><said>said by <a href="/profile/755055" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=755055');">OZO</a>:</said><p>I was under the impression, that you have mentioned "Call Forwarding", while you actually meant "SIP URI" routing...<br> </p></div> :) I thought you were confusing the two and went along with your vocabulary.<br><small>--<br>DPC3825 (bridged mode) - WRT610N + Tomato - Panasonic KX-TGP500 - Asterisk 11.0.2 on Virtual Server <br>Anveo - FreePhoneLine - Voxbeam - Numbergroup - Callcentric - VoIP.MS - Localphone - UKDDI</small>]]></description>
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<pubDate>Wed, 02 Jan 2013 22:02:58 EDT</pubDate>
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<title>Re: SIP forwarding</title>
<link>http://www.dslreports.com/forum/Re-SIP-forwarding-27875378</link>
<description><![CDATA[OZO posted : I was under the impression, that you have mentioned "Call Forwarding", while you actually meant "SIP URI" routing...<br><small>--<br>Keep it simple, it'll become complex by itself...</small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/Re-SIP-forwarding-27875378</guid>
<pubDate>Wed, 02 Jan 2013 21:56:34 EDT</pubDate>
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<title>Re: SIP forwarding</title>
<link>http://www.dslreports.com/forum/Re-SIP-forwarding-27875375</link>
<description><![CDATA[grand total posted : <div class="bquote"><said>said by <a href="/profile/755055" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=755055');">OZO</a>:</said><p>It's not "Call Forwarding". They called it "SIP URI" routing. <br>"SIP URI" routing is billed using usual incoming rates and not free. <br> </p></div>Yes, I don't think anything you have written here conflicts with anything I have posted in this thread.<br><small>--<br>DPC3825 (bridged mode) - WRT610N + Tomato - Panasonic KX-TGP500 - Asterisk 11.0.2 on Virtual Server <br>Anveo - FreePhoneLine - Voxbeam - Numbergroup - Callcentric - VoIP.MS - Localphone - UKDDI</small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/Re-SIP-forwarding-27875375</guid>
<pubDate>Wed, 02 Jan 2013 21:54:48 EDT</pubDate>
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<title>Re: SIP forwarding</title>
<link>http://www.dslreports.com/forum/Re-SIP-forwarding-27875358</link>
<description><![CDATA[OZO posted : It's not "Call Forwarding". They called it "SIP URI" routing. <br>"SIP URI" routing is billed using usual incoming rates and not free. <br><small>--<br>Keep it simple, it'll become complex by itself...</small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/Re-SIP-forwarding-27875358</guid>
<pubDate>Wed, 02 Jan 2013 21:48:50 EDT</pubDate>
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<title>Re: SIP forwarding</title>
<link>http://www.dslreports.com/forum/Re-SIP-forwarding-27875350</link>
<description><![CDATA[grand total posted : DID Numbers|Manage DID|Edit DID shows DID routing. One option is to route the DID to SIP URI.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/Re-SIP-forwarding-27875350</guid>
<pubDate>Wed, 02 Jan 2013 21:44:33 EDT</pubDate>
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<title>Re: SIP forwarding</title>
<link>http://www.dslreports.com/forum/Re-SIP-forwarding-27875327</link>
<description><![CDATA[OZO posted : How to configure "Call Forwarding" to SIP URI in VoIP.MS?<br><br>They have "<a href="https://www.voip.ms/m/callforwarding.php">Call Forwarding</a>" page, which allows to add a new forwarding rule. It has "Phone Number" field where it accepts only digits and not any SIP URI...<br><small>--<br>Keep it simple, it'll become complex by itself...</small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/Re-SIP-forwarding-27875327</guid>
<pubDate>Wed, 02 Jan 2013 21:34:02 EDT</pubDate>
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<title>Re: SIP forwarding</title>
<link>http://www.dslreports.com/forum/Re-SIP-forwarding-27875145</link>
<description><![CDATA[PX Eliezer posted : <div class="bquote"><said>said by <a href="/profile/1836359" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=1836359');">zapattack</a>:</said><p>V.MS does not appear to offer SIP CFWD. <br> </p></div>As [grand total] said, that is incorrect.  They do offer free outbound SIP forwarding.  <br><br>There CAN be small Voip.MS charges for [inbound] SIP calling.  <br>&raquo;<A HREF="/forum/r21533299-What-is-the-voip.ms-sip-url-">What is the voip.ms sip url?</A><br><br><div class="bquote"><said>said by <a href="/profile/1836359" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=1836359');">zapattack</a>:</said><p>The menu further warns that BOTH incoming and outgoing charges apply.<br> </p></div>Outbound charges would apply only to calls forwarded out to phone numbers, not to SIP URI.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/Re-SIP-forwarding-27875145</guid>
<pubDate>Wed, 02 Jan 2013 20:23:09 EDT</pubDate>
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<title>Re: SIP forwarding</title>
<link>http://www.dslreports.com/forum/Re-SIP-forwarding-27875101</link>
<description><![CDATA[grand total posted : <div class="bquote"><said>said by <a href="/profile/1836359" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=1836359');">zapattack</a>:</said><p>V.MS does not appear to offer SIP CFWD. The menu further warns that BOTH incoming and outgoing charges apply.<br> </p></div>Huh? Of course VoIP.MS offers SIP forwarding. If you forward a call to a SIP URI you only pay for the incoming call (as usual) the forwarding is free.<br><small>--<br>DPC3825 (bridged mode) - WRT610N + Tomato - Panasonic KX-TGP500 - Asterisk 11.0.2 on Virtual Server <br>Anveo - FreePhoneLine - Voxbeam - Numbergroup - Callcentric - VoIP.MS - Localphone - UKDDI</small>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/Re-SIP-forwarding-27875101</guid>
<pubDate>Wed, 02 Jan 2013 20:07:55 EDT</pubDate>
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<title>Re: SIP forwarding</title>
<link>http://www.dslreports.com/forum/Re-SIP-forwarding-27875090</link>
<description><![CDATA[zapattack posted : V.MS does not appear to offer SIP CFWD. The menu further warns that BOTH incoming and outgoing charges apply.<br>For my light usage, it is easier to let PBXes and V.MS deal with DDoS problems. As PBXes does not offer DIDs, it is a simple matter to change PBXes servers if one is subject to an attack.<br>The last problem was NY area congestion on the route to Germany during Sandy, so I changed to Seattle.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/Re-SIP-forwarding-27875090</guid>
<pubDate>Wed, 02 Jan 2013 20:03:37 EDT</pubDate>
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<title>Re: SIP forwarding</title>
<link>http://www.dslreports.com/forum/Re-SIP-forwarding-27874640</link>
<description><![CDATA[ConstantineM posted : Basically, yes.  But, again, you first have to make sure that VoIP.ms has a SIP-forwarding server that is more-or-less immune to DDoS; I think it's relatively easy to establish such a server or two, and I think I would be more surprised if they didn't have one already, but, on the other hand, I'm not certain they specifically have such a server, since they didn't seem to have suggested such a solution yet as one of the alternatives.<br><br>You can still try it to see if it might work for you.  CallWithUs lets you provide any +1 number as Caller ID (non-+1-numbers don't seem to be sent correctly to at least +1 destinations, see &raquo;<A HREF="/forum/r27772922-CallWithUs-Caller-ID-to-T-Mobile-USA-prepends-a-1-">CallWithUs Caller ID to T-Mobile USA prepends a "1"</A>), and Anveo also lets you provide +1 or +44 numbers as Caller ID (probably all other codes work, too, but you have to verify numbers first).<br><br>pbxes.org has a feature where you can integrate it with external providers.  You would integrate it with VoIP.ms only from within the VoIP.ms interface itself, for your incoming calls (e.g. just do SIP forwarding from VoIP.ms to pbxes.org), and then integrate CallWithUs and/or Anveo from within pbxes.org interface for outgoing calls.  (Integrating voip.ms from within pbxes.org would be kinda pointless; since the whole exercise is to avoid the external SIP registration servers of VoIP.ms that may be prone to DDoS.)  Then register your phones with pbxes.org.  They, pbxes.org, do seem to have some minute-based usage limits on free accounts, but you can still see if something like that fits your needs.<br><br>Also, CallWithUs lets you have multiple sub-accounts for free, too, and they also support TCP, so you might also try going with CallWithUs and do forwarding onto CallWithUs, avoiding the pbxes.org route.  The only negative about CWU accounts is that the usernames are random numbers with no aliases, so, SIP calling with a CWU account is not user-friendly.  But since they let you specify your own Caller ID, the integration can be quite nifty for PSTN calling.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/Re-SIP-forwarding-27874640</guid>
<pubDate>Wed, 02 Jan 2013 17:09:56 EDT</pubDate>
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<title>Re: SIP forwarding</title>
<link>http://www.dslreports.com/forum/Re-SIP-forwarding-27874026</link>
<description><![CDATA[Goliath13 posted : So, based on this type of setup, how would you configure this?<br>Create a new account with another carrier and just have it accept forwarded SIP URIs from voip.ms?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/Re-SIP-forwarding-27874026</guid>
<pubDate>Wed, 02 Jan 2013 14:07:57 EDT</pubDate>
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<title>SIP forwarding</title>
<link>http://www.dslreports.com/forum/SIP-forwarding-27873926</link>
<description><![CDATA[ConstantineM posted : The whole purpose of avoiding VoIP.ms DDoS is to _not_ register with any of their servers (neither directly nor through pbxes.org).  Don't they let you forward calls to external SIP addresses for free?  Provided it is so, such an approach is supposed to be more reliable than manually registering, especially during DDoS (again, with a good architecture, forwarding is supposed to be immune to DDoS and such, but I cannot say whether VoIP.ms forwarding is or is not).<br><br>In theory, it's impossible to DDoS a server that's only doing the forwarding, since the only way to initiate connections is through PSTN.  But if VoIP.ms only has one server at each location, and with only one interface and one IP-address, and their forwarding is distributed to a location where you would otherwise manually register through SIP, then, of course, it's a different question, and traffic shaping would probably not be there or may not have much effect if the interfaces are completely saturated.<br><br>I would assume that if they don't confirm that it's more reliable, then maybe in the end it is not. :&#45;)]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/SIP-forwarding-27873926</guid>
<pubDate>Wed, 02 Jan 2013 13:40:22 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27872757</link>
<description><![CDATA[zapattack posted : PBXes.org with V.MS is problematic. It refuses to let me register to most of the American servers. Fortunately LosAngeles does work for me and is close enough. <br>Also the V.MS portal shows me as unregistered (to PBXes) at times, even though DID calls complete.<br>The free PBXes package doesn't give you any way of submitting a real trouble ticket. <br>But, for the amount of working (FREE) features, it is still a good deal and, strangely, more reliable than V.MS.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/Re-Alternative-to-voipms-27872757</guid>
<pubDate>Wed, 02 Jan 2013 03:30:52 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27872374</link>
<description><![CDATA[ConstantineM posted : Depending on your usage, and if Anveo addons are deemed expensive, you might also integrate pbxes.org into the mix (although it also has its own usage limits), and, for example, configure pbxes.org to use Anveo for incoming calls (or, hey, for starters, you can even try it with VoIP.ms!), and CallWithUs for outgoing etc.<br><br>And, yes, you might as well try to use pbxes.org with VoIP.ms itself; in theory (and depending on the specific architecture of their setup), they're supposed to be able to forward incoming calls just fine even when under a <em>relatively</em> severe attack (how it is in practice, I don't know, plus it seems that some of their colocation providers (e.g. the one in Atlanta) don't provide them with "reasonable", whatever that would be, DDoS protections/handling).<br><br>P.S. It's surprising noone ever mentions pbxes.org on this forum.  I wish I would have found them earlier.]]></description>
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<pubDate>Tue, 01 Jan 2013 22:00:58 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27872122</link>
<description><![CDATA[PX Eliezer posted : I've never needed to do that but there are lots of folks here who can advise you.<br><br>Note that there are a lot of choices besides Asterisk, such as YATE.<br><br>You may want to start a new thread asking for advice on PBX setup.]]></description>
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<pubDate>Tue, 01 Jan 2013 20:16:10 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27872113</link>
<description><![CDATA[Mango posted : It's easy to set up Asterisk.  But it's not easy to build a fully-featured, high-reliability business phone system.  It's certainly possible, and we'll be delighted to give you advice if you need it, but I wouldn't call it easy.<br><br>There is some value in using a PBX which allows you to use multiple unreliable service providers, but I would rather use a reliable service provider with PBX features.  In this case there are typically fewer points of failure.<br><br>My suggestions would be for you to use Anveo if you want to DIY, or otherwise hire a company such as AcroVoice to set up and maintain your system.]]></description>
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<pubDate>Tue, 01 Jan 2013 20:13:06 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27872111</link>
<description><![CDATA[SCADAGeo posted : <div class="bquote"><said>said by <a href="/profile/1657543" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=1657543');">Goliath13</a>:</said><p>It seems to me that if we can get an Asterisk set up going, we would have the most flexibility to go with whichever provider we want.  I'm just not super confident in my ability to set it up nicely.  How hard would this be to setup for a noob?<br> </p></div>It's very easy: <a href="http://nerdvittles.com/?p=4525">The 5-Minute PBX: It&#146;s Incredible PBX 11 Virtual Machine for VirtualBox</a>. :)]]></description>
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<pubDate>Tue, 01 Jan 2013 20:12:59 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27872095</link>
<description><![CDATA[Goliath13 posted : I've looked into PBXes as well which is another solution.<br><br>It seems to me that if we can get an Asterisk set up going, we would have the most flexibility to go with whichever provider we want.  I'm just not super confident in my ability to set it up nicely.  How hard would this be to setup for a noob?]]></description>
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<pubDate>Tue, 01 Jan 2013 20:04:47 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27872081</link>
<description><![CDATA[PX Eliezer posted : <div class="bquote"><said>said by <a href="/profile/1657543" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=1657543');">Goliath13</a>:</said><p>I need to have 3 different DIDs because they handle different parts of our businesses basically.  <br> </p></div>OK, that's good to know.   So it [would] be easiest for you to have a provider with subaccounts or the equivalent.<br><br>Sometimes we see folks who ended up having multiple DID's in the POTS days even though they didn't need them.  Your setup clearly does need them....]]></description>
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<pubDate>Tue, 01 Jan 2013 19:59:50 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27872053</link>
<description><![CDATA[Goliath13 posted : Sorry, I didn't understand that question PX Eliezer.<br><br>We might have a situation where 2 calls could be coming in simultaneously into 2 different DIDs.  I need to have 3 different DIDs because they handle different parts of our businesses basically.  One DID for business A, one DID for business B & C and toll free for business C.]]></description>
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<pubDate>Tue, 01 Jan 2013 19:49:05 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27872003</link>
<description><![CDATA[PX Eliezer posted : Do you need/use your 3 DID's or do you just need the ability to get 3 incoming calls at a time (which could for example be [one] DID with 3 channels)?<br><br>With a modern IP phone plus extra handsets (such as a Panasonic or late-model Gigaset as often discussed here) a [single] DID with multiple channels can do for you what you used to need multiple POTS DID's.]]></description>
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<pubDate>Tue, 01 Jan 2013 19:31:32 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27871973</link>
<description><![CDATA[Goliath13 posted : Thanks for the help connor79.<br>The lines are for a business environment but our level of usage is very low.  We probably have 400 minutes of usage a month between the 3 lines.  While we have low usage, it's definitely important to not miss calls from clients or have some other issues with the calls.<br><br>I thought about doing some kind of Asterisk setup but I'm not a networking specialist or anything and voip.ms allowed us to achieve PBX functionality without complicating the setup with another piece of equipment.]]></description>
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<pubDate>Tue, 01 Jan 2013 19:19:09 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27871718</link>
<description><![CDATA[Trimline posted : Check out &raquo;<A HREF="http://www.vitelity.com/services_voip/" >www.vitelity.com/services_voip/</A><br><br>They also offer "sub-accounts" for multiple sip registrations on a DID.  <br><br>They can do outbound CNAM for a one time charge as well.   There are discounts available, so check around.]]></description>
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<pubDate>Tue, 01 Jan 2013 17:11:36 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27871663</link>
<description><![CDATA[Davesnothere posted : <div class="bquote"><said>said by <a href="/profile/1606481" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=1606481');">Mango</a>:</said><p><div class="bquote"><said>said by <a href="/profile/1567602" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=1567602');">MartinM</a>:</said><p>I'm currently on a yacht in Mexico for New Year, and that's how much I care about our customers: While everyone is sipping champagne, I'm surfing DSLReports.</p></div>You have no idea how envious I am.  I've always thought this to be an excellent combination.  :)<br> </p></div>&nbsp;<br><b>Yacht-ernet ?</b> :p<br><br>And to paraphrase/bork an old song :<br><br>&#9835; &#9836; <i>"Don't it make my Blue eyes Green !"</i> &#9835; &#9836;<br><br><small>--<br><br>We have only 2 things about which to worry :<br>(1) That things may never get back to normal<br>(2) That they already HAVE !<br></small>]]></description>
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<pubDate>Tue, 01 Jan 2013 16:37:40 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27871287</link>
<description><![CDATA[connor79 posted : We just ditched voip.ms and got all cell phones for our staff.  The problem with what you're looking for is that voip.ms isn't just a termination provider but they provide extra features.  There's literally hundreds of providers that do what they do, but they require you use your own device or phone system/pbx even a hosted one etc as that's where you typically would set up ring groups etc not from the provider directly.  Voip.ms is great in the sense they have all the same features that Asterisk does, ring groupes, queues, IVR etc.  Anveo does this as well, but that's why it's a bit hard to find other providers that work the same since most just provide the access.  I tested RingCentral recently not sure how their pricing is as I just used their one plan where it acts like a PBX and forwards calls to cell phones but it worked great, with a great interface, but I think the pricing is probably higher than voip.ms but what we have learned the hard way is that you get what you pay for.]]></description>
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<pubDate>Tue, 01 Jan 2013 13:45:08 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27871161</link>
<description><![CDATA[VexorgTR posted : My opinion on a voip provider will depend on your usage (in minutes) and reliability needs.    If your DID's are for a business that gets TONS of calls,   I have one idea...  if its for a home office that gets moderate calls,  it's another thing.<br><br>Please add that detail.]]></description>
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<pubDate>Tue, 01 Jan 2013 12:37:54 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27870740</link>
<description><![CDATA[Arne Bolen posted : <div class="bquote"><said>said by <a href="/profile/1572525" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=1572525');">PX Eliezer</a>:</said><p>I look forward to 2013 and hope that it will be good for all of us. </p></div>(20)<b>13</b> = The Lucky Number. ;)<br><small>--<br><A HREF="http://myvoipnews.com/">My VoIP News</a></small>]]></description>
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<pubDate>Tue, 01 Jan 2013 06:44:56 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27870591</link>
<description><![CDATA[JJ_GTA posted : <div class="bquote"><said>said by <a href="/profile/1232816" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=1232816');">Stewart</a>:</said><p>Panasonic cordless IP phone system, you could have DID A ring handset A, DID B ring handset B, etc., with only one user.  <br> </p></div>How do you have multiple DIDs ringing different handsets with only one user/subaccount? I have setup separate users/subaccounts to handle the DIDs ringing to different stations.  <br><br>I am happy with VoIP.ms and Anveo. They work differently but both have the pros/cons just like any provider.  ]]></description>
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<pubDate>Tue, 01 Jan 2013 01:15:29 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27870513</link>
<description><![CDATA[Mango posted : <div class="bquote"><said>said by <a href="/profile/1567602" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=1567602');">MartinM</a>:</said><p>I'm currently on a yacht in Mexico for New Year, and that's how much I care about our customers: While everyone is sipping champagne, I'm surfing DSLReports.</p></div>You have no idea how envious I am.  I've always thought this to be an excellent combination.  :)]]></description>
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<pubDate>Tue, 01 Jan 2013 00:08:47 EDT</pubDate>
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<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27870511</link>
<description><![CDATA[OZO posted : Happy New Year to you too :)]]></description>
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<pubDate>Tue, 01 Jan 2013 00:06:56 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27870497</link>
<description><![CDATA[MartinM posted : Hello Goliath13,<br><br>If there's any way I can help you with the issues you've experiencing, I would be glad to help. Please send me details via PM. <br><br>I'm currently on a yacht in Mexico for New Year, and that's how much I care about our customers: While everyone is sipping champagne, I'm surfing DSLReports.  <br><br>Happy New Year<br><small>--<br>Martin - VoIP.ms</small>]]></description>
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<pubDate>Mon, 31 Dec 2012 23:47:43 EDT</pubDate>
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<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27870492</link>
<description><![CDATA[PX Eliezer posted : This has been a tough year all around, and I don't just mean VoIP.<br><br>I look forward to 2013 and hope that it will be good for all of us.  <br><br>¡Feliz Año Nuevo!   :)<br> ]]></description>
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<pubDate>Mon, 31 Dec 2012 23:46:00 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27870462</link>
<description><![CDATA[SCADAGeo posted : <div class="bquote"><said>said by <a href="/profile/1657543" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=1657543');">Goliath13</a>:</said><p>Voip.ms' sub-accounts feature got the job done really nicely but their reliability is a different story.<br> </p></div>Voip.ms had a rough month.<br><br>MartinM mentioned that the new Toronto location is coming online in January, and it's supposed to have more capacity, too.]]></description>
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<pubDate>Mon, 31 Dec 2012 23:26:51 EDT</pubDate>
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<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27870381</link>
<description><![CDATA[crazyk4952 posted : <div class="bquote"><said>said by <a href="/profile/1281719" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=1281719');">grand total</a>:</said><p>If I have one complaint about Anveo it is that they can easily nickel and dime you if you're not careful...<br> </p></div>^^^this! This is exactly how I feel whenever I consider using Anveo as my primary provider!]]></description>
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<pubDate>Mon, 31 Dec 2012 22:40:21 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27870367</link>
<description><![CDATA[Goliath13 posted : We have 3 Polycom VVX 500s and 1 Polycom VVX 600.<br><br>We're sharing 3 DIDs between the 4 phones. <br><br>Two lines are shared between 3 phones.  1 Did is for a standalone phone and isn't shared.  <br><br>I don't have a PBX setup and Polycom unfortunately doesn't have a line-seize setting to allow it to choose Line * on handset or speaker phone pickup.  What I've done is make the CallWithUs line the top most line.  It's a crude solution but it does the job.  Whenever the handset or speakerphone is picked up it goes to line 1.<br><br>I'd like all 3 phones to ring whenever a call is made to the 2 Dids.  Whoever picks up would answer and if need be can transfer the call between the other phones.<br><br>Voip.ms' sub-accounts feature got the job done really nicely but their reliability is a different story.]]></description>
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<pubDate>Mon, 31 Dec 2012 22:33:28 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27870308</link>
<description><![CDATA[Stewart posted : <div class="bquote"><said>said by <a href="/profile/1657543" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=1657543');">Goliath13</a>:</said><p>So, if I have 3 telephones with 3 DIDs, I would need 3 users ... </p></div>Not necessarily.  "Users" correspond (roughly) to SIP registrations.  If you have three stand-alone IP desk phones, without a PBX or other consolidation hardware at your end, you would need three users, even if you have only one DID.  However, if you have e.g. a Gigaset or Panasonic cordless IP phone system, you could have DID A ring handset A, DID B ring handset B, etc., with only one user.  You could do the same thing with two OBi202s, though I'd recommend one user per ATA, so a failure of one device won't affect the other(s).<br><br>If you have two or more DIDs, e.g. one toll-free and one local, with incoming calls routed in the same phone(s), the extra DIDs would not require additional users.<br><br>What VoIP hardware do you have?  What behavior do you want when each of your DIDs are called?]]></description>
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<pubDate>Mon, 31 Dec 2012 22:04:41 EDT</pubDate>
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<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27870274</link>
<description><![CDATA[grand total posted : <div class="bquote"><said>said by <a href="/profile/1657543" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=1657543');">Goliath13</a>:</said><p>So, if I have 3 telephones with 3 DIDs, I would need 3 users which would necessitate getting the additional $5 upgrade of 6.55 plan?<br>Does the plan encompass all DIDs though or is it just for a single DID?  If I had to get a 6.55 plan for each DID I guess it wouldn't be worth it.<br>Ordering the pay as you go plan is I think $1.95 per DID a month with Anveo right?  Plus a certain per minute rate plan.<br><br>I'm just trying to get a handle on what the final total is going to be.<br>Forgot to mention that one DID is a toll free number.<br><br>I'm going to be using CallWithUs for the outgoing calls from now on though.  I've never had any issue at all with them over years of using them.<br> </p></div>3 telephones = 3 users, so 'free' plan + $5 upgrade or $6.55 plan. Either way it covers all 3 numbers. You should check that Anveo can port all your numbers.<br><br>Incoming unlimited DIDs are $2 per month, toll-free is $1.49 per month plus 1.9 cents per minute.<br><br>How will you use CWU for outgoing? Do you have some sort of PBX? In that case maybe you wouldn't need a third user.<br><small>--<br>DPC3825 (bridged mode) - WRT610N + Tomato - Panasonic KX-TGP500 - Asterisk 11.0.2 on Virtual Server <br>Anveo - FreePhoneLine - Voxbeam - Numbergroup - Callcentric - VoIP.MS - Localphone - UKDDI</small>]]></description>
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<pubDate>Mon, 31 Dec 2012 21:53:42 EDT</pubDate>
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<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27870245</link>
<description><![CDATA[Goliath13 posted : So, if I have 3 telephones with 3 DIDs, I would need 3 users which would necessitate getting the additional $5 upgrade of 6.55 plan?<br>Does the plan encompass all DIDs though or is it just for a single DID?  If I had to get a 6.55 plan for each DID I guess it wouldn't be worth it.<br>Ordering the pay as you go plan is I think $1.95 per DID a month with Anveo right?  Plus a certain per minute rate plan.<br><br>I'm just trying to get a handle on what the final total is going to be.<br>Forgot to mention that one DID is a toll free number.<br><br>I'm going to be using CallWithUs for the outgoing calls from now on though.  I've never had any issue at all with them over years of using them.]]></description>
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<pubDate>Mon, 31 Dec 2012 21:41:05 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27870232</link>
<description><![CDATA[grand total posted : <div class="bquote"><said>said by <a href="/profile/1657543" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=1657543');">Goliath13</a>:</said><p>Thanks for the reply.<br>When you say "users" does that mean sub-accounts or channels under each DID?  I would need to eventually port 3 DIDs.  Would that mean that I would need to have 3 different users?<br>CallCentric provides 3 channels per DID but they can't be used with a different device I guess or at least they didn't recommend that.<br>I'm just trying to figure out how it would work with Anveo as it seems like the most probable option at this point.<br> </p></div>Users are sub-accounts in VoIP.MS speak. You need as many users as you have telephone instruments. You can have any or all DIDs ring any or all telephones.<br><small>--<br>DPC3825 (bridged mode) - WRT610N + Tomato - Panasonic KX-TGP500 - Asterisk 11.0.2 on Virtual Server <br>Anveo - FreePhoneLine - Voxbeam - Numbergroup - Callcentric - VoIP.MS - Localphone - UKDDI</small>]]></description>
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<pubDate>Mon, 31 Dec 2012 21:36:20 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27870164</link>
<description><![CDATA[Goliath13 posted : Thanks for the reply.<br>When you say "users" does that mean sub-accounts or channels under each DID?  I would need to eventually port 3 DIDs.  Would that mean that I would need to have 3 different users?<br>CallCentric provides 3 channels per DID but they can't be used with a different device I guess or at least they didn't recommend that.<br>I'm just trying to figure out how it would work with Anveo as it seems like the most probable option at this point.]]></description>
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<pubDate>Mon, 31 Dec 2012 21:02:24 EDT</pubDate>
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<title>Re: Alternative to voip.ms ?</title>
<link>http://www.dslreports.com/forum/Re-Alternative-to-voipms-27870152</link>
<description><![CDATA[grand total posted : <div class="bquote"><said>said by <a href="/profile/1657543" onClick="this.blur(); return popup(event,'/uidpop?ajh=1&uid=1657543');">Goliath13</a>:</said><p>I've heard about Anveo here but would I be able to achieve the same thing as voip.ms?  Any other companies out there that provide dependable service?<br> </p></div>I switched from VoIP.MS to Anveo. Very happy with the service. I still have a balance in the VoIP.MS account, just in case, but at this point I can't see using it. <br><br>You can have multiple sub-accounts (users in Anveo), but not as cheaply as VoIP.MS. The 'free' Anveo package gives you only 2 users you can add a 3rd for $5 per month or upgrade to the 'starter' package for $6.55 per month which gives you 3.<br><br>If I have one complaint about Anveo it is that they can easily nickel and dime you if you're not careful, but I can put up with that because the service has been faultless and, more importantly, my wife agrees!<br><small>--<br>DPC3825 (bridged mode) - WRT610N + Tomato - Panasonic KX-TGP500 - Asterisk 11.0.2 on Virtual Server <br>Anveo - FreePhoneLine - Voxbeam - Numbergroup - Callcentric - VoIP.MS - Localphone - UKDDI</small>]]></description>
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<pubDate>Mon, 31 Dec 2012 20:58:31 EDT</pubDate>
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<link>http://www.dslreports.com/forum/Alternative-to-voipms-27870072</link>
<description><![CDATA[Goliath13 posted : Hi,<br>I'm looking for an alternative to voip.ms.  We currently have 3 dids with them and reliability has been an issue on and off and I do have issues with taking a long time to hear ring back on calls once in a while.<br><br>Anyway, I'm experimenting with callcentric but they don't have the same sub-account system that voip.ms does.  With voip.ms I can create a sub-account for another phone and create a ring group with 2 phones for a single phone number.  With Callcentric, for the same thing to work, each phone has to have its own separate account, with separate billing and all the hassle associated with it.<br><br>I've heard about Anveo here but would I be able to achieve the same thing as voip.ms?  Any other companies out there that provide dependable service?<br><br>Thx & Happy New Year!]]></description>
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<pubDate>Mon, 31 Dec 2012 20:25:17 EDT</pubDate>
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