I am using fpl with obi200.
the device was working fine until yesterday.
starting today can't make incoming or outgoing calls.
power cycled the device few times.
no changed on network/router/modem etc.
checked the SIP settings on obi200. all looks good.
I have even updated the sip server to
voip4.freephoneline.ca:6060
anyone know what is going on or what can i try?
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[Other] freephoneline with OBI200 dead today
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[Equipment] VVX 400, 802.3af and Cisco 3750G POE?
Any reason that combo shouldn't work?
I suspect some ding-dong did some bad cabling, but anyone have that combo of switch and phone running?
An older Polycom Soundpoint conference unit is working fine, the VVXs not powering up, and the cisco not seeing and CDP or LLDP traffic on the ports.
3750G POE version does standards-based 802.3af, and as best I can tell the phones do too...
Remote cable tests off the switch show me two things of note, but I can't find good info on the standard that says whether POE should still work:
- The two main data pairs are intact, but crossed (I suspect someone bought a bunch of x-over patch cables)
- The two extra pairs show shorted, but not sure if I'm seeing the POE transformers on the phones as a short
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[Asterisk] No ring back tone from Google Voice outbound calls
I have been experimenting with Google Voice on freePBX on and off since last year. Unfortunately my PC's hard drive died a couple of weeks ago and I had to re-install everything on a new one. I installed freePBX 14 (Asterisk 13/Motif/OAuth2) on a Debian Jessie. Everything seems to be working except that I don't have the ring back tone from outbound calls. The phone, hooked up to Asterisk through an HT802 ATA, is dead silence until the other end answers. I have found a solution to get the ring back tone by changing Trunk Dial Options (Connectivity->Trunks) from "T" (Asterisk default) to "Tr". But I remember I did not have to do anything like this last time. So I am wondering if this is the correct solution and if I have missed some necessary configurations when I set up the system. Thanks in advance for any help I can get.
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[General] Ring.To/RingTo ends on 31 March 2018
I didn't get this in an e-mail, just happened to notice when I went to their web site tonight. Service ends at the end of March. The first part of their post is below. They're recommending SideLine but I don't know if you can port in a number to SideLine much less bringing in multiple numbers.quote:RingTo is saying goodbye
It's been a pleasure working with customers like you these last few years on RingTo. While we've learned a lot over the course of our journey with RingTo, we're focusing our ongoing development elsewhere. So with that said, we'll cut to the chase here: we are sunsetting RingTo on 3/31/2018.Edited to add: Yes, SideLine does allow porting in at least one number; unsure about multiple numbers.
Port-in info: https://sideline.zendesk.com/hc/en-us/articles/214936063-iOS-Porting-FAQs
Port-in from RingTo specifically: https://sideline.zendesk.com/hc/en-us/articles/360000127846-Porting-your-Number-from-RingTo
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Its been a while
And now that my company is switching to Switchvox, Asterisk and I cross path again... :D
What a small world... Now that I am back at working with all this, I cross path with Anav and I am reminded of dslreports...
Allot has happen sense I revisited all mys VoIP stuff... Is good to see that Gigaset came back to life....
Anybody heard of NexVortex?
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VOIP.ms Rotten Porting Experience
I've used www.voip.ms for years with at max a dozen numbers (DIDs). They've done well the vast majority of the time until now. I recently attempted to port out a number. It’s only reasonable that voip.ms use automation as the primary means of fulfilling ports. Computer automation is FAST and I expect 99.9% of ports succeed via automation.
In this case, despite having voip.ms provide me the necessary authorization info via email (matched my account records as well) and emailing it to the receiving carrier, it’s been 9 days, hours of frustration, and a dozen unsuccessful attempts.
Account Name **** ****
Address *** **** ****
City ****
State **
ZIP Code *****
No pin
Account number and BTN- **********
Voip.ms assures me if only the receiving carrier would share the PON (port order number) identifier with them they'd investigate and resolve whatever the hangup is. I find it fishy voip.ms doesn't already see the PON when each port attempt takes place (it seems they're outsourcing that part of their business to another vendor and collaborate poorly)
It's equally suspect as to why the receiving carrier doesn't see the PON identifier either. The receiving carrier may outsource as well.
When the automation doesn’t succeed, my experience has been that voip.ms shirks assuming ownership for the problem and an effective secondary means to allowing these to succeed.
The receiving carrier has been willing to conference call, chat, email chain etc. Especially because the receiving carrier could be communicating with voip.ms the very moment the request is initiated, voip.ms acknowledge and promptly resolve any hitch in the process. Voip.ms has stonewalled such reasonable efforts.
It seems rotten a person should have to go so far as to contact the FCC/NPAC.
Before I give Google Fi a shot, is it easy to get help from the FCC/NPAC??
Is anyone aware of an escalation contact at voip.ms that is good about getting things done??
Hopefully fleshing out possible gaps in their business process, and channels to work through the gaps, can help folks in a similar situation from falling through the cracks.
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Plantronics to buy Polycom (and Obihai)
https://www.cnbc.com/2018/03/28/plantronics-to-buy-video-conferencing-gear-maker-polycom-for-2-billion.html
tl;dr: Thank goodness for naf and friends.
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SIP No outbound audio/media between pbx and ATA on diff vlans
I posted this in the utm forum too, but maybe you guys can point me in the right direction.
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PBX ip (freepbx but called wazo in the pic above) = 10.10.1.76 (internal lan)
ATA (obi200) = 10.1.3.102 (vlan 3)
The ata is configured on its own vlan to isolate it from the main lan. The ata is able to connect to external sip provider (on the internet) without any signaling or media issues.
PBX is configured on the local network and it too works well with its configured trunks and local lan clients.
Goal: An extension on the pbx has been configured exclusively for the ATA's use. Firewall rule below has been set up to allow the ATA to register successfully to the PBX.
Problem: That's as far as I get. The signaling sets up correctly but RTP/media is only getting passed inbound. That is, when a call is made using the extension configured above, I can hear the other party, but they can't hear me. Firewall has been disabled in freepbx to rule that out.
I'm not seeing anything in the firewall log or IPS. Ideas? Thanks!
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[Equipment] RPi 0 and Rpi 0 W on sale US $3.14 @MicroCenter
I just posted here about the subject. You can grab them from your local MicroCenter store before the offer expires and/or out of stocks. I am sure you can use the Rpi 0 as a JTAG tool and probably Rpi 0 W as a PBX, etc.
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[Equipment] Someone Was Looking for a DECT Desk Phone
I can't find the thread to reply to.
Someone was looking for a SIP DECT cordless desk phone. At the time the only ones found were Panasonic's.. Vtech also makes them, VSP608 for example.
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google aiy voice kit $3.14, a way to SIP door phone?
microcenter offers $3.14 for google aiy voice kit (including voice hut, speaker, mic, button), seems qualified as a sip door phone. possible to turn this kit to sip door phone?
http://www.microcenter.com/site/content/Google_AIY.aspx?ekw=aiy&rd=1
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[Equipment] Tomato Shibby QoS configuration note
Recently I was configuring a Tomato router for QoS. I was confused to discover all the VoIP equipment started dropping registration. I checked the tracked connections and verified the VoIP traffic was correctly classified as High. That should have worked, but when I disabled QoS, they immediately re-registered.
The problem was that I had set the maximum speed for the High class to No Limit. On older builds of Tomato I've used, this exempted that class from throttling. This time I did not observe that behaviour.
I reset the maximum speed to 100% and now things are working properly. Instead of No Limit, now I use No Ingress QOS for UDP which is a setting not present in older builds. This seems to produce the same behaviour.
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Any Alternative to CallCentric?
Any Alternative to CallCentric free number wise? I need a did that accepts incoming calls only, outgoing not needed. today as an example I only got 1 call out of the 40 that called today. Most of the time users get a CallCentric error message. This has been an on going thing for 2 years now. I would use Google Voice BUT calls pick up the instant it starts ringing instead of only when voicemail picks up by phone system.
Thanks. I'm up for paying a couple of dollars a month. But I'm done with call centric.
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"Off the shelf" VOIP (anveo, twilio, etc) sms iPhone/Android app
With today's (well-discussed) close of Ring.to I moved most of my RT numbers to Anveo -- both consumer and Anveo Direct.
Ring.to had a good SMS text messaging component. But now I'm looking for an easy way to send (and receive, though I have "receive" forwarding set up) text messages from those numbers for the very rare occasions I would need to (rare enough I don't want to make a big investment of time and effort to do so).
Textable was brought up. https://textable.co/ Plug in your API info for Anveo, Twilio, Flowroute, voip.ms, or Bandwidth.com (Telnyx and Nexmo "coming soon") and it seems to work well with only maybe 10 seconds of lag. But it currently only works with one provider/one number per Texable account.
I use Acrobits softphone app for iPhone, and https://smsglue.com/ provides some insight on tweaking Acrobits to handle texting for voip.ms, thus making it a texting app. I assume (perhaps wrongly) the same could be done for my Anveo numbers as well as the couple I have Twilo, Tropo and Voxbeam.
I want something easy from my phone that doesn't require logging via a browser into anveo's or others' dashboard to send a text message. And while it might be an interesting intellectual exercise to figure, I know I don't have the programming experience/patience/time combo to write my own program.
Does anyone else have suggestions for quick and easy apps (or tweaks of apps or other work-arounds) to make texting work easily for multiple existing VOIP numbers/accounts at the low bar, of say, texting on par with the quality of Google Voice, and without porting them to something else?
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Cloudatcost adds annual fee.
Crooks at cloudatcost has added an annual fee without informing anyone. Noticed an invoice on my dashboard and then found this in terms:
9.18 Customers with a onetime payment service is subject to an annual maintenance fee of $9 which will be invoiced 12 months after using our service. This does not apply to users that have a monthly paid service. This Maintenance fee will ensure proper hardware upgrades and maintenance to reduce degradation of onetime payment services.
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CircleNet has returned.... queue the ominous music
Hello DSLReports team,
I'm the owner of CircleNet LLC and previously was active on dslreports. I've been away from the board here and development on new CircleNet features has slowed to a crawl. But now we're back :-)
Why were you gone?
I've been taking care of a relative of ours that's suffered a brain aneurysm/stroke . It's been an uphill fight but she's 90% recovered now. SOOO much thanks to the patient folks at UVA.
Is stuff still running?
Yep, I've been care and feeding CircleNet.
So your back what's new?
Well we're now taking customers from the US (Sorry still no Ca).
Our porting cost has dropped to $10
Our interface has improved and FINALLY we have voice mail.
Our URL http://www.circlenet.biz/
What's the same?
Our call quality.
The awfulness of our website (Yep it's terrible).
Our great rates (Check out our simulator).
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[Asterisk] asterisk - outbound dtmf
Have been using asterisk to do pin-less international calling for a few months now:
exten => _+44.,1,NoOp("Receiving UK call to ${EXTEN} id: 8011") same => n,Dial(Motif/gvoice_a/${CALLING_CARD}@voice.google.com,,rD(011${EXTEN:1}))
This used to work, however it stopped since the last few days. It appears the DTMF being sent is falling on deaf ears:
-- Sending DTMF '01144XXXXXXX4' to the called party.As if the other end never gets the digits.
If just ${CALLING_CARD} is dialed and rest of the dest is punched in by hand, it does work.
DTMF[7637][C-0000001c]: channel.c:3888 __ast_read: DTMF begin '5' received on SIP/203-0001b DTMF[7637][C-0000001c]: channel.c:3899 __ast_read: DTMF begin passthrough '5' on SIP/203-0001b DTMF[7637][C-0000001c]: channel.c:3802 __ast_read: DTMF end '5' received on SIP/203-0001b, duration 140 ms DTMF[7637][C-0000001c]: channel.c:3843 __ast_read: DTMF end accepted with begin '5' on SIP/203-0001b DTMF[7637][C-0000001c]: channel.c:3872 __ast_read: DTMF end passthrough '5' on SIP/203-0001b DTMF[7637][C-0000001c]: channel.c:3888 __ast_read: DTMF begin '4' received on SIP/203-0001b DTMF[7637][C-0000001c]: channel.c:3899 __ast_read: DTMF begin passthrough '4' on SIP/203-0001b DTMF[7637][C-0000001c]: channel.c:3802 __ast_read: DTMF end '4' received on SIP/203-0001b, duration 140 ms
This made me experiment with macros such as :
[macro-thib]exten => s,1,Playtones(941/200, 1477/200) ; my version of sending a "#" exten => s,2,wait(4)exten => s,3,SendDTMF(${NUMBER}#) [macro-dwait]exten => s,1,Playtones(425/50,0/50) ; whatever this tone is, I am not sure.. exten => s,1,Wait(${ARG1})exten => s,3,SendDTMF(${NUMBER}#)
with "thib", the DTMF digits make it to the called party, it dials the destination - one big caveat, the called end cannot hear anything I say.
Has anyone experienced this before?
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[VOIPo.COM] Down?
Is VOIPo down? Vpanel shows both my home phone and our American Legion in devices connected as being OK but neither phone can receive incoming calls (goes straight to voicemail). My home number can make outgoing calls OK. (Cannot test outgoing at the Legion right now).
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Cheapest Route to Port Landline to VoIP (Apologies in advance)
.. I know this question has been asked many times, but the optimal route at any particular point often change. I was wondering if anyone knew of the most cost-effective method currently? T-Mobile no longer sells their $.99 SIM and while some other MVNOs have cheap SIMs, their minimum activation amounts offset any savings.
Would a port to Anveo work? I have an existing Anveo Consumer account. Also does a successful port to a provider like Anveo successfully recharacterize the number as not a landline number which can then go to GV (the ultimate destination for this number)?
I'm planning to Port from Verizon FiOS, if that helps. Many thanks in advance to all the veterans here!
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Google Voice begins WiFi calling beta
Hello Google Voice users,
Be the first to try out calling over Wi-Fi and mobile data directly from the Google Voice apps!
Wi-Fi calling lets you:
Reduce roaming charges when you’re on the go
Keep in touch even when you have poor cell service
Make calls from any device, not just your phone
Supported platforms: Android & Web. iOS support coming soon!https://productforums.google.com/forum/#!topic/voice/aI7m1PrV21Y/discussion
Known Issues:
Obihai devices won’t work if you enable calling over Wi-Fi (e.g. Incoming calls won’t ring your Obihai phone if you turn on calling over Wi-Fi and mobile data)
You won’t be able to use incoming call options (call recording & call transfer) when you enable calling over Wi-Fi and mobile data
On your computer, Wi-Fi calling only work with Chrome. Support for Firefox, Safari and Edge coming soon!
Android bluetooth support:
You won't be able to use the buttons on your bluetooth device to answer calls or hang up
Unable to use earpiece mode when a bluetooth device is paired
Depending on your Android version, your calls might drop if you try to switch between Bluetooth and speakerphone.
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