Minerva Hosting - Soluciones de telefonía
Foros
Entrar...

Google



Boletín de novedades

Apúntate si quieres ser el primero en enterarte de las novedades


 
 

VoIP World - Forum Voice over IP


AVISO: Contenidos ajenos a nuestra web obtenidos mediante métodos de sindicación (RSS) facilitados por la web a la que pertenecen.



[CaRP] php_network_getaddresses: getaddrinfo failed: Name or service not known (0)
VOIP Tech Chat forum - dslreports.com
VOIP Tech Chat forum current topics

$20 Orange Pi has onboard 8GB eMMC flash,would RaspPi PBX's run?

I was just reading a small blurb about the new "Orange Pi PC Plus Quad Core Development Board" which supposedly will sell for $20 and will include 8GB eMMC flash memory on the board itself. Since I'm not that familiar with memory types I just wondered if eMMC memory is more reliable than SD cards. The problem with running a small PBX on a Raspberry Pi is that if you don't take extreme measures to minimize SD card writes, in particular by disabling most of the logging performed by Asterisk and FreePBX, the SD cards have a tendency to die prematurely. I have an original Raspberry Pi and SD card that's been working for about three years now but I have turned down the logging to ignore all but critical entries. I would just wonder if eMMC memory is any more robust. I think 8GB would be enough to run a small PBX with just a few extensions. The larger problem would be whether any existing software would run on an Orange Pi. I suppose it's just different enough from a Raspberry Pi that if you tried to install RasPBX, or the XiVO build for the Raspberry Pi, it would probably crash and burn. But it sure would be nice to be able to build an inexpensive, fast, and hopefully reliable PBX that would not depend on an SD card for storage. This is the article I was reading, and for the record I have nothing whatsoever to do with this company that makes these, in fact I have never even seen an actual Orange Pi: http://www.cnx-software.com/2016/06/09/orange-pi-pc-plus-quad-core-development-board-with-1gb-ram-8gb-emmc-flash-sells-for-20/ -- I refuse to believe that corporations are people until Texas executes one.

[Unlock] Discovery on the Linksys SPA-2102 that may allow an unlock.

I recently purchased an SPA2102 ATA by Linksys/Cisco, hoping that I could find a permenant unlock for it. I have tried the Sunrocket Unlock ISO, but that doesn't work on my ACNDIGITAL locked box. I happened to open the case (via the 4 screws hidden under the foot pads), and noticed not only the 48-pin TSOP NOR flash chip, but I also see what I believe to be an EEPROM. If my suspitions are correct, this may be where the parameters are set on locked devices. If so, and it is easily readible (ie data is not encrypted), then it should be fairly easy to read and manipulate. For those who don't have the means to read and write an EEPROM, I'm hoping that it is writable from the system software. I am planning to purchase a Teensy development board and an 360-clip to read both the system Flash ROM and the EEPROM. Before anyone starts going out and buying all of the locked devices hoping to flip them - I want to warn: this is not a sure thing, and even if I can do it, it could be years before I can get something going.

[Asterisk] OAuth 2.0 Support for Asterisk 13 or Asterisk 14

The attached package should provide OAuth 2.0 supoort to any Asterisk 13 or Asterisk 14 system: 1. Extract oauth2.tar.gz to the /root directory. 2. Edit oauth2.creds. The first line must contain your Oauth 2.0 Client ID and the second line must contain your OAuth 2.0 Client Secret. 3. Make oauth2 executable: chmod +x oauth2 4. Execute oauth2: ./oauth2 5. Use your OAuth 2.0 refresh token(s) as the Password(s) in FreePBX's Motif module (or as the secret(s) in xmpp.conf of plain Asterisk). 6. After clicking Apply Config in FreePBX, wait 15 seconds after the Reloading dialog box disappears before doing anything else. . If you don't already have an OAuth 2.0 Client ID, Client Secret, and refresh token(s): 1. Go to Google Developer Console: https://console.developers.google.com/project 2. Log in with your Google Voice username/password 3. Click CREATE PROJECT 4. Enter a Project name 5. Click Create 6. In the left pane, click Credentials 7. Click OAuth consent screen 8. Enter a Product name shown to users 9. Click Save 10. Click Create credentials 11. Click OAuth client ID 12. Select Web application 13. Enter a Name 14. Enter https://developers.google.com/oauthplayground at Authorized redirect URIs 15. Click Create 16. Record Client ID and Client secret 17. Go to https://developers.google.com/oauthplayground 18. Click the gear icon 19. Check Use your own OAuth credentials 20. Enter OAuth Client ID and OAuth Client secret 21. Click Close 22. Enter https://www.googleapis.com/auth/googletalk at Input your own scopes 23. Click Authorize API 24. Click Allow 25. Click Exchange authorization code for tokens 26. Reopen Step 2 27. Record Refresh token To create a refresh token for additional Google Voice accounts, log out, log in to the desired account, and go to step 17. . Credit to Ryan Tilton, dziny, carlb8, phonesimon, and others for the original res_xmpp.c modifications.

VOIPO To Tmobile Porting Issues

I've been trying to port my number out of voipo to T Mobile for the last few weeks and have run into nothing but trouble. I keep going back and forth and both providers seem to not know what to do leaving me stuck in the middle.. I am hoping someone here can help more. T Mobile requires an account number, PIN and billing zip code to port my number across. According to Voipo, the account number is the same as my phone number and there is no PIN. Attempt 1 - I tried with my phone number and my billing zip code. That failed. . Attempt 2 - Found some other users on the web saying, use their houston texas zip code 77092. Tried that and didnt work. Same error again. Attempt 3 - Tried putting in my cell number as the account number since that's what shows up on Voipo's invoices and the houston zip code. That didn't work either. Same error again. Tmobile support is saying this isn't their problem and the problem is with Voipo. They also keep saying they cant do it without a pin code. Has anyone gone through this process before? Is this normal for voipo? I found them easy to deal with all along but im not sure how to move forward from here.

Flowroute

So I got a trunk failure this am:quote:Warning! Your PBX 'pbx' has detected that a SIP FAILURE has occurred. TRUNK sip.flowroute.com:5060 Request Sent [Generated at 09:30:19 on March 20, 2017] And I cant reach their site ether.... Is it just me?

Some Skype services are down.

Microsoft is suffering from a second outage affecting its account services this month. Services like Xbox Live, Outlook.com, Skype, OneDrive, and Microsoft?s Windows Store are currently not allowing users to sign into accounts. The Verge has tested a number of accounts, and can confirm that services are experiencing widespread issues. Microsoft recovered from an hour-long outage earlier this month which affected Microsoft Accounts. The software giant never revealed the root cause of the issue, and it appears to be occurring again today. http://www.theverge.com/2017/3/21/15009940/microsoft-account-second-outage-march-2017 We are actively investigating problem, preventing users from signing-in and sending Skype-to-Skype messages! We'll let you know as soon, as issues are resolved. http://heartbeat.skype.com/2017/03/problems_with_sending_messages.html

call forwarding passing caller id information

trying to figure out how to resolve getting the right info to show up on my caller id screen. for example i have 2 clients that forward their phones to my phone number. i have 2 ways i tried in my anveo call flow to receive the calls. the first i had the caller id set to pass the number to my phone. so what happens is the person calling my client, their phone number is showing up on my screen (my clients call forwarding just passes the call straight through). which i do want to know who is calling my client. the problem is that because both my clients send to the same number i dont know which client is being called so i dont know how to answer the phone. so secondly i set the callerid to the receiving phone number but that doesnt help b/c its just showing my number (i wasnt sure if it was going to show me my clients number as thats the number that the original caller is dialing) now is there a way to distinguish between the 2 clients that forward to my number and get the caller id of the person that originally called my client? or is the only way to pull this off is to give every single client their own separate DID they forward to so i know the client that is ringing on my end?

Google Voice (New version)

I was doing some cleaning of my GV account and noticed I can no longer filter inbound calls based on my groups and circles with the new UI. Why would the do this? This is one of the things I liked about GV I would create an entry for the business and place it in business group and have the filters setup so that only people in family and business calls would forward. All remaining call would goto GV voicemail. At this point I can switch to old version and do it but I have not tested to see if the filters still actually work.

[Unlock] Unlocking the BasicTalk ATA

Important NOTE: a better unlocking method has been posted later in this thread. Please see this post for a permanent unlock developed by uid://1479488. My soft unlock may help in some cases if the ATA has "called home". I have some good news for those of you looking for an inexpensive ATA. I've just got my hands yesterday on a couple of BasicTalk ATAs (I've had my eyes on them for a few months but I live in Canada and don't go to US that often) and I put together a small tutorial for unlocking them. The ATA is a Grandstream HT701 with a customized firmware. I posted it on my website at http://voipfan.net/unlock/ht701bt.php I will leave the access open to everyone for a couple months then make it available to registered users (like my other unlocking tutorials). Enjoy and if you run into any trouble please post here. -- Providers (through asterisk): voip.ms, freephoneline, smartcall.ro, ipcomms, callcentric. Hardware: Vonage VDV21, Moto VT2x42, Linksys SPA series, Grandstream HT series, Panasonic KX-TGP5x0 http://www.voipfan.net

Crowdfunding for Freeswitch voice prompts

There's a crowdfunding campaign to have Allison Smith record new voice prompts for Freeswitch: https://www.gofundme.com/allison-prompts-for-freeswitch I haven't yet contributed, but plan on doing so shortly.

[Asterisk] Incredible PBX for $10 Raspberry Pi Zero W

We've released an updated version of Incredible PBX for the Raspberry Pi that now supports the $10 Pi Zero W. Setup is a 2-minute procedure. Finding a Pi Zero W is the trick at the moment. Some tips and tricks as well as download and setup tutorial is available here: http://nerd.bz/2mGpBxZ

Obi202 ATA On "DMZ" or LAN?

Crude ASCII "art" network diagram: LAN <-> | router | <-> DMZ <-> | Cable modem | Internet The DMZ is an interesting beast. The cablemodem is configured to present my public IP netblock, a /30, to my router's WAN interface, but it also presents an RFC1918 netblock on its "LAN" ports. Let us say it's 10.1.1.0/24. The CM can serve DHCP, though I currently have it turned off. The question is: Where to place the Obi202? I can put it on my LAN, or I can put it on the DMZ--kinda sorta along with my router's WAN interface. Assuming I have the choice, there are advantages to each approach. Placed on the LAN: The ATA would be well-protected against assault from the 'net, but it could be a threat to my LAN if compromised. Placing it on the CM's LAN side gets it off my secured LAN, but exposes it naked to the Internet. Thoughts? Suggestions?

[CallCentric] Obi202, CallCentric, Muliple Numbers and Outbound Number/CID?

So let us say I have an Obi202, CallCentric (which I do) and two phone numbers (which I eventually will: A number I bought from CallCentric and a number I port in). Is it possible to configure the ATA and CC so that calls going out on physical line #1 present one CID and calls going out on physical line #2 another? I presume it is. It is not immediately obvious to me how that would be accomplished, however. Thanks in advance!

[General] Porting questions

Hey guys, I signed up for a Telos account through the Google play store and purchased a number from them. Lo and behold, I must have missed the part where I had to pay $9.99/mo. to keep it. I swear I didn't see that little tid bit anywhere or I wouldn't have purchased it in the first place. Anyways, It's a nice number for the area where I live so I'm thinking of porting it out to another provider. Telos is OK with that and even provided me with an account number and pin, but most other providers want a signed invoice or LOA in order to port a number in. Ring.to at one time would process a port in request with just an account number and possibly a pin, but unfortunately they have suspended all porting for the time being. I tried voip.ms and callcentric but they need that signed invoice and there's no way that's possible. Apparently it's considered a mobile number so I tried porting it to Google Voice, but that didn't work either. Anyone know of a provider that would process a port in request with using just an account number and pin and without all the other paperwork? Thanks

[Anveo] Does Anveo Retail collect credit card fees from subscribers?

Good day. I understand that Anveo Direct discounts credit card fees from subscribers' payments. Although I deem it a nickel and dime practice, in theory, AD caters to business and its subscribers know what they are doing and accept it. Before I would open an Anveo Direct account, I would like to know whether or not Anveo Retail engages in the same practice. If it does, I probably would stay away from AR. Thank you



 


Gráfico pie página

Inicio | escríbenos | mapa | privacidad | webmasters | publicidad

http://www.recursosvoip.com
© Recursos VoIP
Alojados en Minerva Hosting - Obsesionados con la calidad

Gráfico derecho pie página

Inglés English (automatic translation)