Looks like Vonage uses a new Grandstream ATA for their service, model HT802.
It's Grandstream newest generation of ATAs, with 2 ports.
Currently available at Walmart and BestBuy for $10.
If unlockable, this could be a nice, inexpensive 2 port ATA. Interesting that they are using a MicroUSB port for power.
I don't have a way of getting any of these soon to test as I am located in Canada. However, I am sure there are similarities between unlocking those and the BasicTalk HT701 (though probably they will not accept Mackey's firmware without some modifications).
Edit: seems like some new members are jumping into unlocking these adapters without going through the thread and reading all the comments. The proper unlocking method, tested by many already is posted here.
I still recommend reading through all the posts to learn from other members experience before starting anything.
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[Unlock] New Grandstream ATA used by Vonage [Unlocked]
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[Anveo] Anveo Direct dual setup guide on PAP2T and other Cisco device
First I want to thank everyone in here, others and Cisco for helping me figure this out. There is no doubt in the end this was not worth the effort but there are times where for whatever reason the will to make something work just takeover. I hope in sharing this it will at least help others.
The background is I first look at the Obi 200 device and had been juggling how to do this without having to spend about 60$ to save a few dollars per years in my use case. Of course the irony is I spent many hours on this PAP2T "upgrade" so there goes this logic... In addition, I had a voip provider for many years but it seems with time premium line switch to value line and very knowledgeable support switch to "Is the power on" type of support. All of this got me more and more interested in the Obi 200 device and wondering why I could not have multiple provider on my old PAP2T. This is where I started to spend time researching... I got this to work on the PAP2T but since I took much of the info from other Cisco device Iâm assuming it would work on those newer one as well. Much of this info is in many thread or PDF so I thought having it all in one place might be useful.
- The static IP
You need a static IP or one where the effort and the inconvenience of having to update your Anveo Direct account is less then the payoff of using Anveo Direct. In my case, while my IP is not static it rarely changes so it works for me. I did not try to make it work with a service like Dyndns so I cannot say if it would work but I can say it does not work out of the box.
There seem to have been two choices to get this to work I chose the following:
Set Handle VIA received, Handle VIA rport, Insert VIA received, Insert VIA rport to YES
The other option seem to have been to set EXT IP instead. I chose the former since I only have to update my Anveo Direct account when my IP change and not the ATA as well.
- The ATA dial plan
This is where the first part of the magic comes in thanks to a not very well documented at all feature.
011xxxxxxxxxxxxxx
This is pretty self explanatory the trick is to add the outbound trunk prefix from Anveo Direct in there, for example this dial plan rules says to route international call to Anveo Direct instead of my regular voip provider which is more expensive for international call. Please note that this feature allows to pass a different user id and password then the regular one defined for the line. However, while playing with to try and adjust the caller id, since Anveo Direct seem to use the provided user id as the caller id, I was having some problem. Therefore, I left it out since it is not required to make calls with Anveo Direct.
Note that there is a little bug in the firmware and if we write
011.
instead of routing everything after 011 to Anveo Direct it routes it to the regular voIP provider hence why it needs to be spelled out as above.
- The RE-INVITE problem and solution
I donât like to have people sending random invite and having my phone ring for no reason so I use "Restrict Source IP" the problem with that is Anveo Direct will send a RE-INVITE after a while and if there is no response it will drop the call. The solution to this is another not very well documented feature that "Restrict Source IP" letâs through the proxy defined in Line 1 or Line 2. Therefore, I enabled the line 2 which I was not using and at the same time move the fax to this dedicated line since Anveo premium line are better and cheaper then my regular provider value line. Setup line 2 with "sbc.anveo.com" as proxy and set make call without registering to YES and registering to NO. Leave everything else blank as it is not needed. Adjust dial plan if you are actually going to use the line as stated above.
That about sums it up I hope it is useful to someone else.
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Next time someone says they use POTS because it's more secure...
...I'm going to link them to this. There were half a dozen on this street in similar condition.
[att=1]
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[CallCentric] CallCentric new SMS messaging service
Don't know if anyone noticed, but CC is now offering SMS messaging service for a $1.00/month and $0.01/message.
More information here (you have to be logged-in):
https://my.callcentric.com/sms.php
This URL is public:
https://www.callcentric.com/sms/
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Banned from Obi Forum
Since my Obi 110 stopped working a several days ago, I had been posting on the Obi forum for a solution. Like many, I had gone through the sudden loss of service before. Many on the Obi forum were quick to suggest that I buy a new Obi 2XX device as the solution. NONE from that forum mentioned the quick easy fix I used from www.obifirmware.com. One response did mention 3rd party firmware which caused me to search and find the solution from www.obifirmware.com. I felt as if Obi was trying to hide the solution and I was right! After using the Obi forum to respond to others with 100 or 110 devices who lost service, and suggesting the firmware, I was BANNED from the forum! Clearly Obi does not want others to know about this solution. I am certainly glad I did not invest more in Obi products! Has anyone else been banned by Obi for this? I bet my posts have also been deleted.
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[CircleNet] SIP server appears to be down
Good day.
For about an hour now, my OBi302 has been unable to register on sip.circlenet.biz.
The status web page shows the following (emphasis added):
SP1 Service Status
Parameter Name Value
Status Register Failed: No Response From Server
(server=23.25.121.90:5060; retry in 99s)
I was monkeying around with the OBi302, but didn't change any server settings for the CN SP.
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[Equipment] Android phone vs. dedicated wireless headset
Dedicated wireless phone headsets are priced in the range of $100, why not use $50 android phone and bluetooth with a SIP client on Wifi?
The phone is additional hardware to carry around but here are the benefits that I see.
1.Attach to cheap headset 2. Increase the range. 3. With some cheap cellular service, it can also act as a fail-over line.
Any thoughts? Any SIP client that you would recommend that will do call waiting/hold/transfer (Standard SIP desk phone features)?
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FreePBX, Asterisk, and PJSIP
I'd be interested to know how many FreePBX users are actually using PJSIP rather than Chan SIP. A couple days ago I tried setting up a new install of FreePBX using the instructions at https://wiki.freepbx.org/display/FOP/Installing+FreePBX+14+on+Debian+8.8 in part to try PJSIP. My initial impression is that it is not the easiest thing to set up; things that are easy in Chan SIP become much more difficult (if not impossible) in PJSIP, at least initially.
I tried configuring a PJSIP extension and at first it did not work at all. I wound up tweaking several settings under Asterisk SIP Settings and I am not sure exactly which worked, but this is what I wound up with:
General SIP Settings
Security Settings
Allow Anonymous Inbound SIP Calls - No
Allow SIP Guests - No
Default TLS Port Assignment - unset
Chan SIP PJSip
NAT Settings (used detected network settings which are correct)
RTP Settings
RTP Port Ranges = Start: 10000 - End: 20000
RTP Checksums - Yes
Strict RTP - Yes
RTP Timeout - 600
RTP Hold Timeout - 900
RTP Keep Alive - 20
Media Transport Settings (all blank)
ICE Blacklist (all blank)
ICE Host Candidates (all blank)
WebRTC Settings (all blank)
Audio Codecs
T38 Pass-Through - No (for now, wondering what best setting is)
Codecs
ulaw
alaw
gsm
g726
g722
Video Codecs
Video Support - Disabled
Chan PJSIP Settings
Misc PJSip Settings
Allow Reload - Yes
Show Advanced Settings - Yes
TLS/SSL/SRTP Settings (left at defaults/unset)
Transports
udp
udp - 0.0.0.0 - All - No
udp - xx.xx.xx.xx - eth0 -Yes
tcp
tcp - 0.0.0.0 - All - No
tcp - xx.xx.xx.xx - eth0 - No
tls
tls - 0.0.0.0 - All - No
tls - xx.xx.xx.xx - eth0 - No
ws
ws - 0.0.0.0 - All - No
ws - xx.xx.xx.xx - eth0 - No
wss
wss - 0.0.0.0 - All - No
wss - xx.xx.xx.xx - eth0 - No
xx.xx.xx.xx (udp)
Port to Listen On - 5060 (BE SURE TO SET THIS, IT IS NOT SET BY DEFAULT)
Domain the transport comes from - left blank
External IP Address - left blank
Local network - left blank
Note that xx.xx.xx.xx is the server's IP address, also the client I was using was on a different IP address block behind a NAT router (if that's the correct terminology). In the Chan SIP extension setup, I would always have to use NAT Mode - Yes (force_rport,comedia) to get two-way audio in this situation. In this case, when setting up the PJSIP extension, I found I could pretty much leave all the settings at the defaults.
Also note that until I set Show Advanced Settings - Yes in the PJSIP settings, I could not see the options that included the specific IP address of the server. I think setting this may have helped. But the biggest thing that seemed to help resolve issues was restarting Asterisk, or in one case, completely rebooting the system. For some reason, even if you set the Allow Reload - Yes option, just applying the configuration alone sometimes seems to leave the system in a weird state where things don't work as expected.
Where everything went to hell was when I tried to set up a PJSIP trunk to a remote FreePBX system (that uses Chan SIP only) at the same external IP address as my test extension. After a lot of messing around I finally got it to sort of work, but the minute I did then my test extension could no longer connect. Does PJSIP not like two separate things trying to connect from the same IP address? But then again, I really was not sure what trunk settings to use in the first place. If anyone know any tricks for making a PJSIP trunk on a newer system talk to a Chan SIP trunk on an older one, that would be useful information.
Honestly I'm not sure why I'm even messing with PJSIP, since Chan SIP has always worked well for me, but to me it does not seem like PJSIP is fully baked yet. I found a couple of other weird things that worked fine with a Chan SIP extension, but not with a PJSIP one. Right now I am just poking at this trying to see if I can get it to work. The PJSIP test extension does seem to work fine now, as long as I don't try make a PJSIP trunk to the Asterisk server at the same IP (making a Chan SIP trunk is fine, though). I do find it interesting that a PJSIP extension doesn't seem to need specific NAT settings, once you finally get it working.
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[Equipment] UBNT UVP-Pro unmutes every 15 minutes
So I also have a thread going on the UBNT forums on an issue where my UVP-Pro is unmuting itself every 15 minutes into a call when on speakerphone. Have not tested with using handset or bluetooth.
So I am using the their controller for configuration of the SIP config and running the latest versions. I am using the phone on a FreePBX server, don't have the version at the moment, but it has been updated to the latest.
My question for this forum in not related to the UBNT side but the SIP side as to if something on the SIP side can signal an unmuting of the phone. Just want to pin the blame on where it needs to go.
There was another person that posted on my thread at UBNT who is having the same issue for the past few years but apparently never reported it. So I am leaning that it is a UBNT issue.
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[Anveo] How to Disable Anveo "Voicemail To Text" transcription?
Anveo's Call Flow seems not at all easy to configure... quite confusing.
Arne Bolen mentioned plans to make a Users Manual a few years ago. Sure would be nice if that ever happened. :o
I think this should be a very simple fix, but I cannot find it.
PBX > Voicemail Boxes> Edit > Transcription >
and the "Enable Voicemail Transcription" is NOT checked.
So, Why are some voicemails still being transcribed?
I Do Not want that service, and I do not want to pay for it.
HOW to disable that service?
Thanks, in advance.
Mark484
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[Equipment] "CallForwardOnNoAnswer" not working on OBi1022 SIP Phone
Good day.
I set up my OBi1022 SIP Phone to forward calls to an outside number if a call is not answered after five rings, as follows:
IP Phone/Phone Settings/
CallForwardOnNoAnswerEnable [Ticked]
CallForwardOnNoAnswerNumber 1617XXXXXXX
CallForwardOnNoAnswerRingCount 5
After "submitting the new settings and rebooting the phone, I made four test calls with the same result: The phone screen flashed after the fifth ring, then continued to ring locally.
The following are two redacted entries for the same call from the phone's call log:
Call 1 06/20/2018 08:43:27
08:43:27 From 'BOSTON MA' SP1(1617XXXXXXX) To PH1
08:43:27 Ringing
08:43:42 Call Ended
Call 2 06/20/2018 08:42:59
08:42:59 From 'BOSTON MA' SP1(1617XXXXXXX) To PH1
08:42:59 Ringing
08:43:27 Call Ended
The origination SIP provider's CDR lists only one call at a time.
The CDR for the provider that would have received a forwarded calls does not list any forwarded calls.
This is not a Google Voice number.
Any ideas?
Thank you.
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Can SPA 112 firmware work on PAP2?
I'm talking about the old PAP2 not NA or T. If not, where can I get the latest firmware for it?
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Obihai OBi200/202/302 + OBi1022/1032/1062 firmware mods
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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Manual LRN Routing
Is there are any VoIP provider out there that allows to route calls into Specified LRN/Switch data without Auto-Lookup?
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[Equipment] Current OBi firmware...
Given the June 2018 Polycom/Obihai/Google Voice XMPP-to-SIP transition https://www.dslreports.com/forum/r31938501-Google-Voice-XMPP-support-will-go-away-in-June, and their poor documentation, maybe it would be helpful to track current OBi firmware and availability here, if possible...
[Updated 5/22/2018]
Official/Factory/Stock OBi firmware released on the OBiTALK forum
http://www.obitalk.com/forum/index.php?topic=9.0
Stock OBi 100 Series adapter firmware 1.3.0 (Build 2886) (end-of-life product; invalid link).
Stock OBi 200/300 Series adapter firmware 3.2.1 (Build 5757EX).
Stock OBi 500 Series adapter firmware 4.0.1 (Build 5235).
Stock OBi 1000 Series phone firmware 5.0.0 (Build 3497).
Official/Factory/Stock OBi firmware NOT released on the OBiTALK forum
https://www.obitalk.com/images/OBi202-3-2-2-5857EX-19283837473.fw (misnamed; invalid link, 5/22)
Stock OBi 200/300 Series firmware 3.2.2 (Build 5859EX) adds support for the new GV SIP platform and is being pushed out to devices added to the OBiTALK Provisioning portal. This version can NOT be downgraded (or user-modified), similar to the past GV OAuth2 transition firmware 3.0.1 (Build 4550).
OE
Other references:
User-Modified OBi firmware released on a dedicated user website
http://www.obifirmware.com
User-Modified OBi firmware discussion on this forum:
https://www.dslreports.com/forum/r31792784-Obihai-OBi200-202-302-OBi1022-1032-1062-firmware-mods
User-Modified legacy OBi firmware discussion on this forum:
https://www.dslreports.com/forum/r31741105-ObiHAI-Obi100-Obi110-Firmware-Mod-Discussion
Simon Telephonics Google Voice Gateway
https://simonics.com/gw
Google Voice Gateway SIP interoperation discussion on this forum:
https://www.dslreports.com/forum/r31966059-Google-Voice-Gateway-beta-test-for-SIP-interop
Google Voice Gateway legacy XMPP interoperation discussion on this forum:
...
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Problem Incoming calls from mobiles
I am with Vonage and discovered 3 days ago I can't receive calls from Mobile numbers or international numbers.
Calls from landlines work as do outgoing calls.
I have been trying to research what could be causing specific calls to fail. The calls can't be diverted either, so it seems that are not even reaching the network ?
As some calls can come through - i don't think it can be a registration error
Any ideas?
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VoIP Testing Software
Howdy,
Does anyone have a recommendation on VoIP quality testing software?
Using a provider's echo test or maybe two accounts on the same or different VoIP providers?
I'm not looking for a speedtest type service, but something that can actually test line quality to 'my' location, using 'my' network..
For example giving the application one (or two) of my voip.ms sub accounts and the 4443 echo test number, and it can do some testing for a period of time and give me some results.. Or two sub accounts and a number that this testing application can call between them
Does this make sense? Anybody know of anything?
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Nerd Vittles hit piece on 'Unlimited' SIP Trunks
I see that the Nerd Vittles site has posted an article entitled, VoIP’s Dirty Little Secret: Why ‘Unlimited’ SIP Trunks Are a Very Bad Deal and in this case I happen to think it's quite a bit off base. There are times I agree with Ward and other times I disagree, but in this case the problem is that the data he's using to support his argument seems cherry-picked to make the strongest argument possible, and in the process to perhaps cause readers to think that the metered service he's prompting is a great deal. He does fully disclose this:
Full Disclosure: Vitelity is a Platinum Sponsor of Nerd Vittles™ and our open source projects including Incredible PBX®. We also happen to like their business practices and recommend them without hesitation.
The problem is that he's comparing Vitelity to SIPStation™, but if what he is saying about Sipstation is correct, almost anything other than Ma Bell would compare favorably to SIPStation. So, let's take the worst example you can find to make the comparison, sure it will make your sponsor look better! And considering that he says that his sponsor's rates are at most "a little less than a penny and a half a minute", from posts I have seen recently in this forum I would guess that's far from being the least expensive VoIP deal out there.
However, the part that made me take notice was this:
Third, if you believe these one-call-at-a-time unlimited trunks provide truly unlimited calling, we’ve got some swamp land in Florida that may be of interest. Leave your trunks off-hook for 2 weeks playing music on hold and see how long your account lasts.
And in that he misses the whole point of why people WANT unlimited service. If I have a trunk that is stuck open and playing hold music (or anything else) for two weeks, I want my provider to kill that f---ing connection with extreme prejudice, NOT leave the thing open and just continue billing me by the minute! If they feel they need to terminate my account and put me on a blacklist of people to never do business with again, that's fine, I'd rather they do THAT than just hand me a huge bill!
Those of us who lived through the bad old days of Ma Bell can remember people who accidentally left a modem connected to a BBS and fell asleep waiting for a download to finish (at 1200 bps), or otherwise got distracted and forgot they were online, and got a huge phone bill as a result. While we don't have dialup modems anymore, crap does happen, and I have always said that the whole point of unlimited service is that it's insurance against unexpected high charges if there's a technical malfunction, or someone hacks your account, or whatever. It also encourages your provider to help you out by killing ridiculously long connections, though I don't know if any actually do that.
Personally I am not in the slightest impressed by either of the services he's comparing, and while he doesn't frame it as an advertisement for his sponsor, it sort of smells like one. I would not be surprised if Vitelity uses that article in some promotional manner.
I really have nothing else to say about this, other than that I doubt we'd have seen such a heavily promotional article in the early days of Nerd Vittles.
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Trouble with GV and OBi200
Hi I just bought a OBi200 and have it installed it's calling out and receiving calls just fine but Faxing is another issue. I use a Epson WF-7520 all in one machine and also using Google Voice so when I send a fax it's working fine but receiving them it's getting to the fax machine it says connecting on the screen of my Epson and when the fax is done it says Fax received but then it does not print the fax instead for some reason it goes to GV and leaves a voice mail and when I play it there are the fax tones for the fax coming in you know the beeps that you hear when faxes are being transmitted. Does anyone know or have had this issue?
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Problems installing naf's Asterisk
I'm using a Shuttle XS35 PC with an Intel Atom processor.
I installed Ubuntu Server 18.04
Downloaded naf419-asterisk-a9e5e33 this morning
Did following:
./configure
make
make install
make config
make samples
Only saw these warnings during Make:
[CC] irc2pc.c -> irc2pc.oirc2pc.c: In function ‘irc2pc_’:irc2pc.c:145:12: warning: ‘temp[9]’ may be used uninitialized in this function -Wmaybe-uninitialized] pc[j] = temp[j - 1]; ~~~~~~^~~~~~~~~~~~~irc2pc.c:145:12: warning: ‘temp[8]’ may be used uninitialized in this function -Wmaybe-uninitialized]irc2pc.c:145:12: warning: ‘temp[7]’ may be used uninitialized in this function -Wmaybe-uninitialized]
Then started Asterisk without any config changes.
service asterisk start
Asterisk log:
[Jun 25 14:14:58] Asterisk UNKNOWN__and_probably_unsupported built by root @ shuttle on a x86_64 running Linux on 2018-06-25 13:36:17 UTC[Jun 25 14:14:58] NOTICE[27111] loader.c: 295 modules will be loaded.[Jun 25 14:14:58] WARNING[27111] loader.c: Error loading module 'res_pjsip_sdp_rtp.so': /usr/lib/asterisk/modules/res_pjsip_sdp_rtp.so: undefined symbol: ast_sip_session_media_get_transport[Jun 25 14:14:58] WARNING[27111] loader.c: Error loading module 'res_pjsip_pidf_digium_body_supplement.so': /usr/lib/asterisk/modules/res_pjsip_pidf_digium_body_supplement.so: undefined symbol: ast_sip_presence_xml_create_node (Several more warnings for pjsip modules) [Jun 25 14:14:59] NOTICE[27111] cdr.c: CDR simple logging enabled.[Jun 25 14:14:59] WARNING[27111] res_phoneprov.c: Unable to find a valid server address or name.[Jun 25 14:14:59] NOTICE[27111] chan_skinny.c: Configuring skinny from skinny.conf[Jun 25 14:14:59] ERROR[27111] ari/config.c: No configured users for ARI[Jun 25 14:14:59] NOTICE[27111] confbridge/conf_config_parser.c: Adding default_menu menu to app_confbridge[Jun 25 14:14:59] NOTICE[27111] cel_custom.c: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.[Jun 25 14:14:59] ERROR[27111] loader.c: Failed to resolve dependencies for func_aes[Jun 25 14:15:00] WARNING[27111] res_hep_rtcp.c: res_hep is disabled; declining module load[Jun 25 14:15:00] ERROR[27111] asterisk.c: Module initialization failed. ASTERISK EXITING!
What did I do wrong?
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