So I want to add the ability to configure these devices for GV using oauth without obitalk, similar to the changes for the obi100 (and add an ssh server, for grins).
I think I have the MD5s in the firmware file worked out (its the same "Goodbye! Reboot Now" garbage as the 100), and I see where the oauth refresh token code is, so it should be pretty straightforward unless there is code signing that I missed.
The only hiccup is... I don't actually have an OBi20x :-(
Anyone have one of these devices that wants to be a guinea pig? You should definitely have a way to SPI the flash back *when* i brick the thing the first couple tries...
[or if someone has one sitting in the closet, you could just send it to me. ill name the fw after you :-)]
EDIT: speaking of flash, its supposed to have a w25x128 on board, but is it the SOIC package or some BGA madness?
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Obihai OBi200/202/302 + OBi1022/1032/1062 firmware mods
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Bandwidth Developer Platform
Has anyone worked with this platform? It appears to be a powerful tool and if scripted properly, an easy and cheap way to work with a set of parked numbers if there is going to be relatively minimal use.
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[General] Crazy VoIP Idea...
Hello, I'm looking at trying to setup a home phone system for next to no cost...
Is it possible to configure some kind of software for Windows that will communicate with a dial-up 56k modem and Google voice to have incoming calls come in through that modem to either an analog phone system and make calls out from it? If not, what would be my best option?
Thanks,
Eric
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Google Voice XMPP support will go away in June
http://www.obitalk.com/forum/index.php?topic=13824.0
Will naf be able to fix the OBi100?
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[Future9] Calls marked as completed but all could hear is ringing tone
We got some new phone numbers to call in Europe and they dial correctly but all I hear is never ending ringing tone.
In Asterisk portal, all calls are marked as answered and have duration cost attached.
Other party reported phone ringing but no response on other end (while all we got is ringing tone)
Anybody experienced that in the past?
Using calling-card option with no pin-dialing
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Choosing DIDs
I have two young children I'd like to get some simple-to-remember numbers (that are close to each other, ex. XXX-XXX-5400 and 5500) for when it's time for them to own a cellphone. Aside from CallWithUS, are there any other providers who make their available DIDs searchable? Is there another way to go about achieving this?
As usual, many thanks for any insight.
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[Anveo] Anveo Voicemail Problem
Over the past few days, the "play button" for my Anveo emails is no longer visible for new voicemails appearing in my dashboard/inbox. It continues to appear for older voicemails that were present prior to the problem surfacing. Is anyone else having this issue? The problem doesn't appear to be browser-specific.
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[General] Firewall block list may prevent your voip
from working.
[EDIT corrected address] Today I implemented a very extensive blockklist from https://iplists.firehol.org/
and all my voip.ms registrations failed ….. low and behold some of the servers that voip.ms use are for some reasons caught in this blocklist trap …. so I had to revert to another service for my blocklist until the firehol list is fixed.
Just a heads up for those that use Firehol and voip.ms :-)
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auto-listen, time-and-temperature with VOIP?
Two offbeat questions for this crowd...
1. Is it possible to set up VOIP on a computer so that it auto-answers with live audio? I have an old RadioShack answering machine that allows you to call in and punch a code and hear the sound in the room. (A crude version of home-monitoring.) Is there an easy way to set up VOIP so that it will auto-answer and allow you to hear audio from a device or microphone? The RS answering machine allows you to hear 30 seconds of audio before hanging up, longer or unlimited call time would be nicer.
2. Is there a simple way to set up an automated-voice "time-and-temperature" service on a computer that would auto-answer a VOIP line?
I'm interested in any very low-cost hobbyist solutions out there...
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[Anveo] [Retail] Did Anveo Retail stop charging for "Platform Minutes"?
Hello.
It used to be that that, if a call came in to an Anveo Retail SIP URI and it was answered, the duration of the call was rounded-up to full minutes and there would be a daily and monthly tally of so-called "Platform Minutes" at the bottom left of the account Dashboard.
It appears that the tally has disappeared in the last few days.
Not sure whether all Platform Minutes are now free of charge (doubt it) or the tally widget was inadvertently omitted from the latest web site update (more likely).
If there is still a charge for Platform Minutes in excess of forty per day, it would be nice to have the tally back.
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[Other] Lookup North American number for SMS support
Is there a definitive source to query (presumably for a fee) to determine if a north american number supports SMS (cellular or otherwise)?
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My number w/Callcentic was illegally ported. Can I get it back?
Hi,
I moved my business number to Callcentric. Last night I realized my number was not working and filed a report. I heard back this morning that, first, they investigated and found it was not working. Then a second message later saying they received notice from their "underlying carrier informing (them) that (my) number ported out earlier this morning to another carrier."
Now I know going forward there is some extra protection I can put on accounts that may not have been there - although I've had this number for years prior with no problem like this. But can I get it back? I'm not hearing anything back from Callcentric. I have no way to call them. For me this is an emergency as this is my business number and all of my marketing supplies and online presence and more are tied to it.
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[Asterisk] android native SIP Client - SIPAUA/0.1.001
This is the version on Android 8.1.0:
Useragent : SIPAUA/0.1.001
Where can I find more info on this SIP Client? It seems does not support TLS (only UDP and TCP supported). Also trying to find what codecs are supported etc, search-fu does not return much.
Would anyone know?
Thanks!
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Skype to SIP - any working solutions left?
Can someone suggest a currently working Skype-to-SIP solution, preferably free? Sometimes there's a need to redirect an incoming Skype call to a SIP number to get a ring alert, especially calls from older folks with limited packages knowledge (Skype only). Microsoft currently offers Skype-to-SIP Gateway https://support.skype.com/en/faq/FA10578/what-is-a-sip-profile on business accounts, but setting it requires a paid Skype sub, which many don't have a regular use case for.
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[Anveo] Bug In Voicemail Notification? No VMWI After a New Message
I missed an important message because the Visual Message Waiting Indicator did not turn on.
I only noticed because the VMWI turned on two days later!
Has anyone else on Anveo been having problems?
Contributing factors would be that I'm using the Shared Voicemail function. The VMWI that finally turned on was on the shared subaccount. The mailbox is associated with the main account where the VMWI never turned on.
Also, the affected message happened to have a free sample of the Voicemail Transcription featured applied. It would be terrible if software written for an advertising offer happened to introduce a bug in the voicemail operation!
I just tried a test voicemail and the VMWI of the shared subaccount seem to work. The main account VMWI is still off. Now I have to test to rule out a hardware problem.
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Security heads-up for FreePBX (and really, all) users
In recent weeks, I've started seeing an interesting method of trying to compromise SIP credentials on FreePBX systems. I'll preface this with the fact that the method used here could easily apply to systems other than FreePBX, but seeing as how FreePBX is widely-deployed, it seems they're at least targeting FreePBX systems.
Instead of the usual attempts to brute-force/guess SIP credentials (which are quickly blocked by Fail2ban/responsive firewall/etc), the provisioning server is now being used as an attack vector.
Most phone vendors have a recommended file naming scheme for phone configuration files. I'll use Linksys as an example here, although many Cisco phones use the same scheme:
For your Linksys devices (SPA-2102 ATAs, SPA942 IP phones, etc.) it seems the "recommended" naming scheme is spaXXXXXXXXXXXX.xml -- with XXXXXXXXXXXX being the MAC address of the device
Thus, attackers are walking the entire MAC address range, requesting config files from FreePBX servers, hoping to eventually "hit" a MAC address that's provisioned on that particular server.
This is made easier by the fact that vendors' MAC addresses are predictable. For example, most of the aforementioned Linksys devices have MAC addresses beginning with 00:0e:08 ... thus, it is known all Linksys devices' MACs fall between 00:0e:08:00:00:00 and 00:0e:08:ff:ff:ff. If my calculations are correct, that's 166 (or about 16.7 million) combinations.
Once you "hit" a valid MAC address on the PBX in question, you'll be fed an unencrypted XML file with the SIP credentials in it -- no brute-forcing or guessing necessary. Just configure up your client and start calling!
Lather, rinse, repeat for other phone vendors' config file naming schemes and MAC address ranges.
I have personally seen this activity with TFTP (I know, I know... TFTP over the Internet is a bad idea... try to convince your customers/techs) however it's possible with HTTP (or even HTTPS) as well.
If you must use TFTP for a device, I'd recommend forcing your customer to get static IP service from their ISP, and then only allowing TFTP traffic from said IP(s). The same would probably be prudent with any type of HTTP/HTTPS provisioning as well.
There is the ability (if you are running FreePBX Distro and commercial modules) to secure the provisioning server using HTTP authentication, but this also requires the purchase of the Sysadmin Pro module. I haven't messed around with this yet (may do so in the next few days), but that may also be an option to securing things while still allowing provisioning to happen from anywhere.
Just thought I'd put a heads-up out there; I'd thought of this concept months ago but hadn't mentioned it to anyone. However, I can now say I am definitely seeing it in the wild, so it's definitely an issue at this point.
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Vonage VDV22 + VOIPFAN
Followed instruction on VOIPFAN webpage to unlock a Vonage VDV22
All went well, provisioned with my own SIP settings, both Line1 & Line 2 is up and working
However, when read "Messages", it looks like that regularly, the device still looking for and try to connect to a Vonage server.
Is there any CLI command to stop this?
Thanks!
May 04, 07:43:12 am Ready to make calls
May 04, 07:43:14 am Ready to make calls
May 04, 07:49:34 am Vonage Connect Message [003] Unable to connect to the Vonage network. Please unplug power connector from Vonage phone adapter, wait 60 seconds, and plug it back in.
May 04, 08:23:22 am Vonage Connect Message [003] Unable to connect to the Vonage network. Please unplug power connector from Vonage phone adapter, wait 60 seconds, and plug it back in.
May 04, 08:49:43 am Vonage Register Message [004] Line 1 unable to connect to the Vonage network. Please unplug power connector from Vonage phone adapter, wait 60 seconds, and plug it back in.
May 04, 08:49:48 am Vonage Register Message [005] Line 2 unable to connect to the Vonage network. Please unplug power connector from Vonage phone adapter, wait 60 seconds, and plug it back in.
May 04, 08:50:04 am Vonage Register Message [005] Line 2 unable to connect to the Vonage network. Please unplug power connector from Vonage phone adapter, wait 60 seconds, and plug it back in.
May 04, 08:50:04 am Ready to make calls
May 04, 08:50:09 am Ready to make calls
May 04, 08:56:52 am Vonage Connect Message [003] Unable to connect to the Vonage network. Please unplug power connector from Vonage phone adapter, wait 60 seconds, and plug it back in.
May 04, 09:31:56 am Vonage Connect Message [003] Unable to connect to the Vonage network. Please unplug power connector from Vonage phone adapter, wait 60 seconds, and plug it back in.
May 04, 09:56:38 am Vonage Register Message [004] Line 1 unable to connect to the Vonage network. Please unplug power connector from Vonage phone adapter, wait 60 seconds, and plug it back in.
May 04, 09:56:43 am Vonage Register Message [005] Line 2 unable to connect to the Vonage network. Please unplug power connector from Vonage phone adapter, wait 60 seconds, and plug it back in.
May 04, 09:57:05 am Internet Connect Message [002] Not reaching the Internet. Please check if your Internet service is down. Try restarting modem. For DSL: Refer to DSL Setup instructions and check your PPPoE setup.
May 04, 09:57:13 am Vonage Register Message [204] Line 1 and 2 unable to connect to the Vonage network. Please unplug power connector from Vonage phone adapter, wait 60 seconds, and plug it back in.
May 04, 09:57:13 am Connected to Internet
May 04, 09:57:13 am Vonage Register Message [005] Line 2 unable to connect to the Vonage network. Please unplug power connector from Vonage phone adapter, wait 60 seconds, and plug it back in.
May 04, 09:57:15 am Ready to make calls
May 04, 09:57:18 am Ready to make calls
May 04, 10:02:00 am Retrieving Profile
May 04, 10:06:09 am Vonage Connect Message [003] Unable to connect to the Vonage network. Please unplug power connector from Vonage phone adapter, wait 60 seconds, and plug it back in.
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How does one do a Direct IP call with Cisco SPA5xx phones?
There are options for enabling direct IP dialing, both initiating and receiving calls..
But how does one actually dial an IP address?
I tried 173*243*2*68 as I have done on other phones but no-go, Invalid. Tried creating a contact but I couldn't get the . to show either.
It seems simple but I can not figure it out. Not even YouTube videos on it.
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Caller ID Name lookup question
I just began with voip.ms and have questions regarding the CallerID Name Lookup.
What's the difference between their paid query service (LIDB/CNAM) and the free default one? I inquired and the default uses CNAM.
Does this include outbound caller ID? My outbound caller ID doesn't work. Even though I set it in my spa112.
Isn't this suppose to be included in the service? Don't most providers offer this free of charge?
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[Equipment] SPA9000 + Shared Line Apperances
Good morning everyone,
So, I've aquired a SPA9000 + a few SPA942s cheaply. I'm trying to setup a shared line appearance system to half-a emulate a key system. However, I'm running into a few things.
Yes, I understand it's EOL so I may just be SOL.
One of my main issues is that it seems that it does not enforce one call per shared line. I can call the shared line appearance, pick up the line on one of the phones, and then using the analog phone attached to the SPA, call the same extension and it rings the same extension on the other registered phone.
I've turned off call waiting service on both the SPA9000 and the SPA942s. There has to be something simple I'm missing. This seems to be quite a grave oversight/bug when it comes to SLA if you can "Share" two calls on one appearance.
Firmware on everything is 6.1.5. 6.1.5(a) on the phones.
TIA!
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