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<title>VOIP Tech Chat forum - dslreports.com community</title>
<link>http://www.dslreports.com/forum/voip</link>
<description>VOIP Tech Chat forum current topics</description>
<language>en</language>
<copyright>Copyright 2007, dslreports.com</copyright>
<pubDate>Wed, 08 Feb 2012 18:44:20 EDT</pubDate>
<lastBuildDate>Wed, 08 Feb 2012 18:44:20 EDT</lastBuildDate>

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<item>
<title>Recommendation for internet fax service</title>
<link>http://www.dslreports.com/forum/remark,26848570</link>
<description><![CDATA[We use MyFax right now and have been happy for 2 years but lately they've been really bad.  I'll upload a 7 page PDF for example, and hit send and then it will say the fax is only 3 pages and only send 3 pages.  Will try other PDFs and it's the same thing, then will retry it later and it works.  I did live chat yesterday, it said 6 people were in front of me (this was 1pm eastern time), had to wait about 15 minutes then it said i'm next, then it says no one is available to assist me send an email!  I was furious so I want to get rid of it.

My main reason though is because when I look at our volumes, last month there were 7 sent pages and 9 received.  Month before was 12 send, 19 received.  The highest in the past 6 months was one month where there were 49 received and something like 39 sent.  Even if it was 10 cents a page that's still cheaper than myFax.

Wondering if anyone has any cheaper recommendations, or a service where I preferably load up some funds and then it deducts per fax?  I'm in Canada so preferably a service that's not all US focused.  Thanks!]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,26848570</guid>
<pubDate>2012-02-03 08:42:24</pubDate>
</item>

<item>
<title>[CallCentric] [HT-286] outbound faxing problems</title>
<link>http://www.dslreports.com/forum/remark,26868768</link>
<description><![CDATA[I used to be able to fax outbound with minimal trouble; fax speed would drop all the way from 14400 to 4800, but they'd go through. Now, however, the connection will drop during T.30 setup.

The HT-286 settings are all default (including Fax Mode = T.38), except that the LEC (Line Echo Canceller) is disabled. The HP 640 fax machine's settings are the following (no idea which ones might be truly pertinent):

Forgot this tidbit: I did try to set Fax Mode to Passthrough (PCMU/G.711&mu;), but that didn't work either.

Also, as noted below, I had turned the Modem Speed all the way down to 2400, with similar results.

Come to think of it: I had a temporary DID through CallCentric while I waited for the Death Star to allow a port-out. That temporary DID was issued by Earthlink Business (Choice One), but the permanent DID was ported to Peerless Network. The faxing problems started around that time. Correlation isn't causation, but...

Modem Speed: 14400
Fax TX Level: -12dBm (can be between -15 and 0)
Fax RX Level: -43dBm (can be between -43 and -48)
CNG Count: 2 (can be between 1 and 4)
T1 Time: 55 (can be between 30 and 150)
Fax Error Rate: 10% (can be 5, 10 or 20)
Flash Time: 600 ms (can be 100, 280 or 600)
Pause Time: 3s (can be between 1 and 9)
Make/Break: 40/60 (can be 40/60 or 33/66)
Ring On Time: 15x10ms (can be between 0x10 and 99x10)
Ring Off Time: 60x10ms (can be between 0x10 and 99x10)
DTMF High Level: -6dBm (can be between -99 and 0)
DTMF Low Level: -8dBm (can be between -99 and 0)
USB Mode: Fast (can be Fast or Slow)
PWM Value Setup: 168 (can be between 100 and 208)
--
Politics, noun. A strife of interests masquerading as a contest of principles. The conduct of public affairs for private advantage. - Bierce]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,26868768</guid>
<pubDate>2012-02-08 12:01:20</pubDate>
</item>

<item>
<title>PBX call-backs</title>
<link>http://www.dslreports.com/forum/remark,26861358</link>
<description><![CDATA[Hey, 
I have a question about PBX call-backs to give a dial tone to call out on. 

A while ago when I was looking into VOIP products I think I came across people doing this.  My phone bill is through the roof now because of work, yet work refuses to support my expenses.  So I am looking for alternatives.  

I never have stable enough data connections to go full VOIP, which would be ideal!  But my provider does offer unlimited incoming calling for 15 bucks a month. 

Essentially I would like to figure out how to setup a PBX call back scheme to allowing me to dial out on an incoming call.  Will this even work if I am all over the country side and often don't have data connections?  

Can someone please point me in the right direction to getting this setup?  (I am Canadian btw)

Muchly appreciated!]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,26861358</guid>
<pubDate>2012-02-06 18:47:49</pubDate>
</item>

<item>
<title>Zyxel PK5000Z QOS</title>
<link>http://www.dslreports.com/forum/remark,26866062</link>
<description><![CDATA[I'm trying to get acceptable sound quality from my Nettalk VOIP device. It does ok, but as soon as another machine uses the Internet, all goes to hell. Here is the config page. I need some help with QOS settings to give this device the highest priority. I've already statically assigned the IP to the one you see here, but the part I don't get, is it won't accept the config unless I set the TOSbit in option 3 to 6 or 7. I'm not clear on what this is, or how it works, but I know that this doesn't not help my sound quality issue.
--
I'm with the Central Government. I'm here to help you. Now bend over, really, I'm helping you, just, just stay still. You'll feel better in a moment.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,26866062</guid>
<pubDate>2012-02-07 18:34:23</pubDate>
</item>

<item>
<title>[Future9] Jitter problems</title>
<link>http://www.dslreports.com/forum/remark,26869097</link>
<description><![CDATA[Anyone else having jitter problems on F9?  My Comcast speeds are 35mb/5mb, so should not be a bandwidth problem.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,26869097</guid>
<pubDate>2012-02-08 13:06:43</pubDate>
</item>

<item>
<title>Building global voice network design</title>
<link>http://www.dslreports.com/forum/remark,26869846</link>
<description><![CDATA[For the sake of discussions, let's assume the following environments

* Call center
* Corporate/Enterprise 

Based on what I gathered, following is list of exploration

* Having Avaya phone system to support call center since I heard that Avaya is best solution
* Having Cisco UCM to support corporate as Cisco is best solution
* Having SIP trunks to one or more providers
* Possibility of maintaining private MPLS backbone/cloud to support voice, video, and data with multicast and one VRF for each network

I'd like to hear what everyone's opinions and thoughts regarding the following.

1. Avaya vs Cisco
* What are the financial and technical reasons of having Avaya system to support call center and of having Cisco UCM to support corporate?
* How should technical intercommunication in place be assuming there are both Avaya and Cisco voice systems? Should there be SIP trunks between the two systems?

2. One vs Multiple providers
* Beside redundancy and cost saving, are there any other benefits of running SIP trunks with multiple providers compared to just run SIP trunk with single provider?
* What are the financial and technical challenges of integrating local provider and global providers specially when companies as client do not manage the providers' network?
* What are the financial and technical challenges of integrating two or more global providers specially when companies as client do not manage the providers' network?
* Which factors to look in selecting local or global SIP providers?

3. Voice Firewall
* What are benefits and challenges of having voice firewall?
* Is it wise to use Cisco ASA as voice firewall? I never see any real-life Juniper SRX firewall implementation as voice firewall so I'm unsure if Juniper SRX firewall is better option as voice firewall
* Any thoughts of Acme Packet firewall? Is it any good or else?

Thanks]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,26869846</guid>
<pubDate>2012-02-08 15:21:26</pubDate>
</item>

<item>
<title>Reliable Outgoing Toll Free Calls without Reg</title>
<link>http://www.dslreports.com/forum/remark,26834692</link>
<description><![CDATA[I have searched high and low for reliable toll free voip providers but I have found only Future-Nine to be reliable. I need a backup to Future-Nine and would appreciate if anyone could point me in the right direction.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,26834692</guid>
<pubDate>2012-01-31 11:55:50</pubDate>
</item>

<item>
<title>Australian 13xx numbers</title>
<link>http://www.dslreports.com/forum/remark,26868559</link>
<description><![CDATA[Are any of you guys using a VoIP provider outside of Australia to call 1300 phone numbers down umner?

When dialed from within Australia, they cost between 35c to 45c, however, I used the rate calculator at Future-Nine for +6113451731 and it told me it was a Special Services number chargeable at a rate of $0.0173

I do not have an F9 account so cannot prove or disprove if that rate is achievable. Anyone care to let me know if there is a cheap way to call 13xx numbers through a VoIP provider?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,26868559</guid>
<pubDate>2012-02-08 11:20:14</pubDate>
</item>

<item>
<title>new vonage smartphone app</title>
<link>http://www.dslreports.com/forum/remark,26868201</link>
<description><![CDATA[http://www.vonagemobile.com/

i just tried the android version. free to US for now.

any # you can receive sms on can be set as the caller ID.

nice backup for outbound calls showing my GV # as caller ID.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,26868201</guid>
<pubDate>2012-02-08 09:55:10</pubDate>
</item>

<item>
<title>Multiple ATA&#x27;s/DID&#x27;s and one way audio</title>
<link>http://www.dslreports.com/forum/remark,26868195</link>
<description><![CDATA[Hello,

I am hoping to get some suggestions, Thanks!

Voip provider is voip.ms,

SPA3000 - one DID(line1), using proxy toronto.voip.ms:5060, SIP Port 5060, and STUN stun.sipgate.net

SPA112 - two DID
first DID(line1) using proxy toronto.voip.ms:5080, and SIP Port 5061, and no STUN
second DID(line2) using proxy toronto.voip.ms:5080, and SIP Port 5062, and no STUN

Both ATA's are behind my home router running DD-WRT.

I have no problem with only SPA3000 running, no one way audio, no problem registering.

But I am having problems when both ATA's are running. When I am on the phone using SPA3000, if someone wants to dial out using SPA112 and either line, one way audio occurs on SPA112's lines. With the following message in syslog,

02-08-2012&#9;22:45:21&#9;Kernel.Error&#9;192.168.1.36&#9;*** ICMP error on RTP ch 0. Err count 1.

I have tried using three different ports that voip.ms provides, and 3 different SIP port (5060,5061,5062) on each line, and it doesn't work. Also tried using a different STUN on SPA112 and no success there either.

Can anyone please give me suggestions on this set up? am I even doing this right? (I am pretty new to this).

Thank you.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,26868195</guid>
<pubDate>2012-02-08 09:53:07</pubDate>
</item>

<item>
<title>[Ooma] ooma increasing taxes and fees 354%!</title>
<link>http://www.dslreports.com/forum/remark,26644951</link>
<description><![CDATA[After only a year of charging $11.75 per year, I now got an email announcing an increase to $3.47 per month. That's a whopping 354% increase. 

Have taxes really gone up that much? I doubt it. 

Alternatives to ooma? :mad:]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,26644951</guid>
<pubDate>2011-12-11 12:49:56</pubDate>
</item>

<item>
<title>[General] Why won&#x27;t my phone number work???</title>
<link>http://www.dslreports.com/forum/remark,26861691</link>
<description><![CDATA[I have a phone number, 1-800-637-xxxx with Callcentric. When I try to call the number from an AT&T small business line in Chicago or from an AT&T residential line (2 miles away from the small business line) it gives me that automated "Either the area code or telephone number is not valid" error message. The same thing happens with Comcast Xfinity Voice phone service.

If, however, I try to call the number through Skype, Google Voice, or my Verizon cell phone it works just fine.

I've contacted Callcentric and they cannot figure it out. I'm on hold with AT&T but I'm worried they won't be able to figure it out.

Can anyone think of a technical reason why my phone number can't be reached from some phones.

I've made sure over and over that I'm calling the right number, tried calling other local, long distance, and toll-free numbers and I can't seem to figure anything out.

If you wish to give it a try and report your service  and whether or not it went through that might help too.

I've had this number for over a month but have only noticed the problem today. I also have a second Callcentric account with a different 800 number and am having the same problem.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,26861691</guid>
<pubDate>2012-02-06 19:59:16</pubDate>
</item>

<item>
<title>fongo</title>
<link>http://www.dslreports.com/forum/remark,26866357</link>
<description><![CDATA[looks interesting:

http://www.fongo.com/]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,26866357</guid>
<pubDate>2012-02-07 19:42:30</pubDate>
</item>

<item>
<title>[General] VoIP business phone line service in the GTA</title>
<link>http://www.dslreports.com/forum/remark,26865766</link>
<description><![CDATA[I'm looking at dropping my (Toronto-area) Bell business land line, which is costing close to $100/month. I'm switching my Internet service to TekSavvy (after 20 unhappy years as a Robbers customer!), who offer phone service. However TSI don't mention anything about business phone services on their website. I want to transfer my business number, either to a VoIP service plan or some other home phone service. My only requirements are that:

1. I can keep my current phone number.
2. The Caller Name Display for incoming/outgoing calls lists my business name.

The barrier I'm facing is with Bell not allowing me to have a residential phone line registered to a company name. I want a business line, just not at the prices they're offering. What are my options? Any help would be greatly appreciated :)]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,26865766</guid>
<pubDate>2012-02-07 17:31:05</pubDate>
</item>

<item>
<title>(Resolved) Outage at Junction Networks [OnSip] service</title>
<link>http://www.dslreports.com/forum/remark,26866301</link>
<description><![CDATA[Noticed this after my phone's status light started flashing, relating to Junction Networks [OnSip] service.  They are a good outfit and I am sure will be back up soon....

6:15 Network Alert
We are aware of many customer reports of an inability to make calls and access admin.onsip.com as of 6:10 PM ET. We are treating this situation with the highest priority and will update this network alert with more information as we have it.

6:31 ET Update
Services appear to be back for some customers. Please feel free to update us here with your status. We will add more RFO detail shortly.

6:45 ET Update
Services were up briefly at 6:30 ET, but we have confirmed network issues in our NY datacenter at this time. We greatly apologize for this interruption in service. We are working to resolve this situation and will continue to update.

7:20 ET Update
We have engineers on-site in our NYC datacenter and should be able to update with more specific information soon. Once again, we greatly apologize for this inconvenience and understand this is causing an interruption in your work day. Please continue to refer to this Network Alert.
http://www.onsip.com/blog/nicole/2012/02/07/reports-of-inability-to-make-calls-and-access-adminonsip]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,26866301</guid>
<pubDate>2012-02-07 19:30:21</pubDate>
</item>

<item>
<title>[Equipment] RTP300 internal call trasnfer??</title>
<link>http://www.dslreports.com/forum/remark,26845036</link>
<description><![CDATA[Hello,

I have been a silent reader of this forum for a long time now and learned a lot tricks from here so first like to thank every one for that and now I have questions may sound stupid to few here but I will ask.

Here what I am trying to achieve I have my RTP300 configured for freephoneline.ca service on line 2 which is working fine now I have callcentric account where I do I have a US DID which I like to configure on line 1 and get that transferred to line2 internally is it possible ?? as I know I can use PBXES and sipsorcery etc but that thing is just not working for me and I believe freephoneline.ca block all incoming sip calls, so my best bet is to registered both of them in RTP300 and get them transferred any help would be appreciated thanks in advanced.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,26845036</guid>
<pubDate>2012-02-02 13:17:47</pubDate>
</item>

<item>
<title>$2.10: Android: GrooVe IP - Google Voice VoIP</title>
<link>http://www.dslreports.com/forum/remark,26858757</link>
<description><![CDATA[On sale today until 11PM EST: https://market.android.com/details?id=com.gvoip&rdid=com.gvoip

Normally $4.99]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,26858757</guid>
<pubDate>2012-02-06 08:06:24</pubDate>
</item>

<item>
<title>2 VOIP Lines with SpeedTouch 780WL ?</title>
<link>http://www.dslreports.com/forum/remark,26862951</link>
<description><![CDATA[I am thinking of getting a 2nd voip line, but, ?

Can anyone of you voip-expert people tell me if, in fact, my Speedtouch 780WL (DSL/Voip-ATA/Router/Wireless/Combo...) can handle "2" unique VOIP lines. ?

What I mean is, (voip-wise) is, can I have a:
613-xxx-xxxx voip service/number on Phone1(fxsport1) jack on my 780WL
as well as a,
613-yyy-yyyy voip service/number on Phone2(fxsport2) jack on my 780WL

And no, I do NOT use the PSTN(analog)Pots port, on my 780WL, aka No analog Bell service, other than a plain Dry-Loop service that I pay monthly for.
BTW, I have been, as of up to now, sucessfully using a VOIP service/line on my existing "Phone1" port for the last 4 years.

If you need any other details, I will be glad to try to provide them for you.

Thanks all.

Rick.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,26862951</guid>
<pubDate>2012-02-07 03:35:41</pubDate>
</item>

<item>
<title>[Voip.ms] Inbound DID issue</title>
<link>http://www.dslreports.com/forum/remark,26848382</link>
<description><![CDATA[At about 0530, I received an e-mail from someone trying to call my DID saying its been busy when trying to call. I did a few checks from cell and another voip provider and all were dead air or busy. No call logs where shown in voip.ms. Changed servers, no resolve. At this time 0715, all seems to be back to normal. Anyone else notice any issues early morning?. I left my ticket open with voip.ms to find out what may have happened.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,26848382</guid>
<pubDate>2012-02-03 07:18:31</pubDate>
</item>

<item>
<title>[Asterisk] Trying to configure my Linksys PAP2 with Asterisk</title>
<link>http://www.dslreports.com/forum/remark,26826119</link>
<description><![CDATA[I am running FreePBX and I am trying to incorporate my PAP2 into the mix. Currently one of the lines is configured with a Broadvoice account, and I plan on keeping it separate from the Asterisk deployment. The second line is configured as an extension of the PBX. Line 1 is connected to a cordless phone, and my issue is I dont have a second phone to connect to Line 2. Is there any way for me to either:

1) Have one line of the PAP2 associated with both Asterisk and the VoIP provider?

2) Have any inbound (internal) calls from other extensions of the PBX to Line 2 ring to Line 1? This is really my preferred situation so I am hoping its possible! 
--
"No you won't" -The American people to President Obama (11/2/2010)



]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,26826119</guid>
<pubDate>2012-01-28 21:22:26</pubDate>
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