Here is what i'm trying to accomplish:
Google voice does not allow to send international text messages...
But they can receive
My primary and only number people communicate with me is a google voice#
Problem is I cannot text internationally with my google voice #
Proposed solution send message through twilio showing my google voice ID https://support.twilio.com/hc/en-us/articles/223133967-Changing-the-sender-ID-for-sending-SMS-messages
Can this work ? maybe there's a much easier system to send SMS with my caller ID and not have to develop something new?
Thanks in Advance
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[Other] Twilio/ClickAtell: Anyone with experience?
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[Equipment] Obi Voice Gateway not going thru
I have been using Obi 100 for years and then switched to Obi 200 with pretty much the same configuration (or so I thought). I have several voice gateways configured to call out based on dial plan, but I use them very rarely. I have never had any issue calling out on those gateway on obi 100, but with the obi 200, I noticed if I haven't used any gateway for a long time, the first call using the gateway always failed with either dead air, or if I waited long enough, I'd get an error message that there's no response from the service provider.
However, once I hung up and dialed again, the call would go thru without any issue. And it'd be ok for a while until I stopped using it and forgot about it and one day I needed to use the voice gateway it'd fail again.
I have used 3 different providers and had the same issue and if I put them in one of the SP they always worked, so the problem is likely on my side, either with the settings on the Obi or my network.
I wonder if anyone has any idea what may be causing this issue. TIA.
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[Equipment] New Panasonic dect SIP phone KX-TGP600
I just noticed Panasonic has a new dect SIP phone:
http://panasonic.net/pcc/products/sipphone/products/kx_tgp600/index.html
In addition to the usual dect handsets, it looks like they have a dect desk phone for it, also.
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[Equipment] New firmware for Gigaset C610A IP
Despite Gigaset taking down the English version of their website (and pulling out of the North American market?), there seems to be a firmware for the Gigaset C610A IP phones (and possibly others). The new firmware version is 42.231 (422310000000 / V42.00).
I am not able to find any information about it, but I decided to install it anyway. I downloaded it through the web interface of the phone. Nothing appears to have changed visually.
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The end of the (IMO) best CLEC?
CenturyLink, a Network Provider, to Acquire Level 3, a Rival
http://www.nytimes.com/2016/11/01/business/dealbook/centurylink-a-network-provider-to-acquire-level-3-a-rival.html
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FlowRoute's published rates do not include USF taxes
Good day.
In my eternal search for excellent VoIP providers, I opened a FlowRoute account and received 25 cents credit to make test calls.
After figuring out how to set up an OBI200 to work with FlowRoute (took me a while because there is a "trick" to transmitting outgoing Caller ID) I made a few test calls and was impressed with the stability and quality of the calls.
Clicked the CDR page to find out how much of the 25 cents was gone. Most of it was still unused. Noticed an interesting thing, though, there were two totals. One for calls at the published $.0098/minute rate billed in one second increments (notwithstanding that the web page states that calls are billed in sixty second increments -- they probably mean sixty one second increments), and one for "USF Fees."
All of the other VoIP providers I do business with (CC, VoIP.ms, Localphone and one or two others) publish rates inclusive of all taxes and fees. Not sure why FR chooses to charge them separately. Having said that, unless there is a show stopper event between now and the time my quarter is gone, I will probably fund and use the FR account.
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[VOIPo.COM] Expanding SimuRing
Right now only out VOIPo phone and another phone can use SimuRing. Many of us have a cell, home, work, sat office, etc. Called the nice peeps up and they said there are no immediate plans to change this? Was hoping to get them support to put this on the dev plate. :)
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Any alt to 3CX?
I gotten used to it, But I have 2 voIP lines mostly managed by my mobile phone one is number based from the 80s I like people to hear the song for a while before going the voicemail and the other is what I use for my personal number for them to be forwarded to be put on hold till I answer there call.
The problem I'm having with 3cx is when someone calls, I almost NEVER get the call via the installed 3cx client app the caller just hears the hold music and I NEVER see the incoming call. Using the iOS iPhone app. Is there a better system thats free that can do the same abouts what I need with 3cx? Just to have a ringback tone till I answer with a good generic voIP app?
Thanks!
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[Asterisk] Don't set your hostname = 127.0.0.1 in /etc/hosts
For some reason, people have been doing this, and it's a bad idea.
The scenario is that in Asterisk sip.conf, the "externhost" is set to a certain hostname, or in FreePBX Asterisk SIP Settings, the external host name is set. The idea is to provide a hostname that points to the external IP for NAT mapping.
Then in /etc/hosts, that same host name is set to 127.0.0.1.
What would be the reason for this? So that Asterisk can always find itself? There is a bad side effect. When Asterisk registers to a remote SIP service, it uses the information it finds from that externhost to build a Contact: header. When the externhost is mapped to 127.0.0.1, the Contact: becomes sip:user@127.0.0.1.
Some services may ignore that but others, including GVGW, accept it since it is what the user has requested, but the end result is failed incoming calls.
The only name that should be mapped to 127.0.0.1 in /etc/hosts is localhost.
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[General] Who's responsible for removing 411 listing for a VoIP DID
I'm confused at this point: the number was listed while still owned by Bell Canada, now that I moved (thank God!) to Anveo, how can I remove the listing from 411? Who's on the hook: Bell or Anveo?
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[Equipment] KX-TGP600 Firmware Upgrade
For those of us in the North American Market, it's been a juggling act with this phone and using either the Canadian or US download links to get the latest and hopefully greatest versions.
Version 1.129 was released last June. To update from 1.129 to 2.200 (Canadian download but US in the filename), I had to go from 1.129 to 1.177 and from 1.177 (This adds web gui logout ability amongst others) to 2.104 and then finally to 2.200. It never would allow me to go straight from 1.129 to 2.200. Do not get discouraged if you have to reboot three times after each upgrade to get your accounts to SIP register. This excercise is not unheard of with SIP equipment upgrades. Just let the TGP600 download to all your handsets each step of the way.
The older firmwares are here: http://panasonic.net/pcc/support/sipphone/download/TGP6/old_firm_na1.html
The Latest is here: http://panasonic.net/pcc/support/sipphone/download/TGP6/na1.html
Do not forget to read the latest Manuals plus the supplementary Admin guide as there is considerably more you can do with this device now. The logout ability starting with 1.177 is a good starting point for the web gui users among us.
Latest Admin Guide: http://cs-im.psn-web.net/Global/SIPPHONE/sipphone_net/download/TGP6/manual/CE/TGP600_AG(en)_WA.pdf
Supplementary Admin Guide with recent changes and additions: http://cs-im.psn-web.net/Global/SIPPHONE/sipphone_net/download/TGP6/manual/CE/TGP600_AG_Supplement(en)_XA.pdf
On my registration with one of my VOIP providers it shows as User Agent: Panasonic-KX-TGP600/02.200 (Mac Address).
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[General] Caller ID Spoofing on VOIP
Thread got deleted -- this is a different topic moderators, but I would say they're not doing more as they see the PSTN as dying, and it's not being enforced by the FCC, plus, we're dead. And, it's money lost, so yes, you're right.
CPN Screening on DMS came out in 1998 -- maybe earlier.
DMS would send the following bits when CPN Screening was enabled for PRIs.
SI of 01 for user provided number, verified, passed
SI of 11 for network provided number
SI of 00 for user provided number, not screened for spoofing
SI of 10 for user provided, verified, and failed (may be a spoof)
10 could be set to swap to 11 without telling you, and throw out the spoof, or reject the call.
So, I can't spoof. I could spoof on VOIP though. We did a friendly spoof to test security.
So I'm asking VOIP providers why they aren't doing this. They can't be bothered. We did something like this with private and unknown callers too.
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[Anveo] Anveo DIAL BY NAME
Hello all !
Anveo offers Dial-by-name directory only in English. I am a reseller and many of my customers (French) want this function but I cannot because it's only in English.
Do you, no english users, have a solution for this ?
It would be GREAT to have the possibility to select Audio File to change default Auto-attendant voice. Furthermore, it would be great to have the possibility to select Audio File for the extention members.
Thanks and have a nice day !
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Is this UK number supposed to be charged at a premium rate ?
So a couple months ago someone in our office had a support remote session and were supposed to call into a UK conference number. The number is +448003134262. This is supposed to be a Toll Free call in UK.
Our VoIP provider charged that call at approx $1 / min (we were charged over $160 for that call). I opened a ticket with the provider but they insist it's a premium number.
I checked that number against Anveo Direct, Callcentric, Localphone, Phone Power, FlowRoute, circlenet, CWU and all showed the call free or under 2 cents/min. Other providers like voip.ms, ooma, voipo don't have that prefix at all in their list.
Does anyone have a definitive answer whether this is supposed to be a premium destination ?
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[VOIPo.COM] Porting Time Warner Cable Phone Number to VOIPO
I have signed up for VOIPO service on Friday October 28, 2016. I sent in all the paperwork by fax so VOIPO could process the number port order. I received confirmation that they received the paper work and I got a second email giving me the Firm Order Commit date of Friday November 4, 2016. It does not give a time frame when the port will be completed on that date. Does anybody know what time does VOIPO pull the ported numbers into their system?
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Sanyo/Pansonic HR03 Eneloop 1.2V 800mAh NiMH Recharg. AAA Battery
Use these batteries in my Gigaset C610 handsets.
Bought a bunch of the batteries at the same time, I think it's been less than 2 years (??) and now they are all failing around the same time.
Anyone else experience this, or have suggestion as to batteries?
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Changing Business From RingCentral to Anveo (HELP PLEASE)
Can someone please help me understand the service types I should order (with Anveo) to transition a business away from RingCentral?
I will make this as simple as possible:
(a) The company currently owns 7 Polycom VVX-410 SIP phones, which I'd like to reuse.
(b) The main number can receive 4 simultaneous calls, and calls can be placed from it.
(c) Each of seven (7) employees has their own phone number (can receive calls directly).
(d) The company probably uses 3,000 outbound and 1200 inbound minutes per month.
(e) The company has RingCentral auto-attendant and voice-mail.
I am trying to figure out what services to order from Anveo (retail), to duplicate this setup.
For example (and I may be totally wrong):
For the main company number:
=========================================
Inbound - Office Unlimited with 3 inbound channels - $8.95/mo
Inbound - Additional 1 inbound channel - $8.95/mo
(the above gives me 4 trunks for "inbound" calls)
** What do I do for outbound calling on these trunks (see my CLID concern, below)?
For each of the 7 employees:
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User 1/Inbound - Pay Per Minute - $1.95/mo
User 2/Inbound - Pay Per Minute - $1.95/mo
User 3/Inbound - Pay Per Minute - $1.95/mo
User 4/Inbound - Office Unlimited - $8.95/mo
User 5/Inbound - Office Unlimited - $8.95/mo
User 6/Inbound - Office Unlimited - $8.95/mo
User 7/Inbound - Office Unlimited - $8.95/mo
(low-volume users 1-3 will go on pay-per-minute)
** What do I do for outbound calling on these trunks (see my CLID concern, below)?
Some operational facts:
=========================================
(a) The business is located in the USA, and so are the clients (and calls).
(b) Calls to the main number will be answered by the Anveo auto-attendant, where callers are given the option of reaching a particular person, or a general department, or the front desk attendant, or else leaving a general message. Of course, if the person's extension doesn't answer, voice-mail is taken for that person's extension rather than defaulting to the general message box. I'm assuming this can all be done with Anveo IVR and call routing?
(c) I want each user to have an "outgoing" line appearance they can use to place calls with (so people they call only see the main company #).
(d) I want each user to have their "direct" line appearance they can use when they want clients to see their direct phone #.
Questions:
I'm still thinking in terms of trunks (which are tied to a DID number) in setting up the above example. Is there a better way to do this, where outbound trunks are used and the CLID (caller ID) can be varied depending on which line appearance initiated the call? In other words, a "shared" group of SIP trunks is used for outbound calling and the CLID that customers see depends on if our employee pressed the "outside line" button or "personal line" button to make the call.
Can someone please show me, using an example similar to the above, what Anveo (retail) services I should be ordering to get this all working... assuming that my example isn't really practical, that is. Other suggestions are HIGHLY welcome, as long as they don't require an in-house Asterisk (or similar) system. For example, CallCentric has very limited IVR/Auto-Attendant functionality compared to Anveo. Voip.ms does not make their IVR offering clear on their page (and I refuse to have to register an account for the privilege of getting to see details of their offerings).
Thanks again
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"all circuits are busy" message on my RentPbx / Incredible PBX
Hello:
The short of it:
My PBX users are periodically getting the "all circuits are busy" message on my RentPbx / Incredible PBX.
Can you suggest what might be wrong with my setup?
Thanks,
Rob.
The long of it:
I have Anveo Direct setup as per http://nerdvittles.com/?p=5680 so I have these items setup to support it:
Trunks:
AnveoOut (Custom)
Anveo1 (sip)
Anveo2 (sip)
Anveo3 (sip)
Anveo4 (sip)
Anveo5 (sip)
Outbound Routes:
AnveoSip (trunk sequence = AnveoOut)
Inbound Routes:
3 Anveo Direct DID's whose "Destinations" are separate DISA's
I also have voip.ms setup with the following items to support it:
Trunks:
vmsT1 (sip) (Register String: vms_sub-account 1)
vmsT2 (sip) (Register String: vms_sub-account 2)
vmsT3 (sip) (Register String: vms_sub-account 3)
OutBound Routes:
**** None ****
Inbound Routes:
DIDvms1 (Routing: sub-account 1) Destination: DISA_vms1
DIDvms2 (Routing: sub-account 2) Destination: DISA_vms2
DIDvms3 (Routing: sub-account 3) Destination: DISA_vms3
I also have one extension set up for one user who uses his Yealink to place out going calls only. (As a matter of fact, all the uses use the PBX to place out going calls only.)
There are only 2 main users: the Yealink user and one DISA user. The other 5 DISA users use the PBX very infrequently.
Usage of PBX:
Anveo Direct voip.ms Total
Inbound 100 1017 1117
Outbound 1700 0 1700
Total 1800 1017 2817
Anveo Direct informed me that I may have 10 incoming calls *** and *** 10 outgoing calls going on similtaneously.
Do I need to setup 10 Anveo Outbound routes? (Eg: AnveoSip, AnveoSip2, ... AnveoSip10.)
Any help with this would be appreciated.
Rob.
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Question about original IncrediblePBX
Hi, I have a question about the original version of IncrediblePBX, IOW not the XiVO or 3CX variations. I thought I had read some time back that during the install is is now possible to select from a menu which features you want, except on a Raspberry Pi. But I cannot find that version. Ideally I want to install it on a system already running either Ubuntu Server, Scientific Linux, or Centos.
I'm pretty neutral about the operating system, as long as it's a long term support version, but I would prefer to be able to install the operating system first, then run a script to install Incredible PBX, but only if it will give me the option to select the addon features I want and omit the ones I don't. If there's no script install that will do that, then an ISO will also work, it will just be a bit more effort for me. But I need to be able to select the features.
I'm not looking for a minimal install that simply omits everything, because I do want incoming and outgoing fax support, and also oauth support for Google Voice. I could have sworn I read a while back that it was possible to select features from a menu, but I can't find that post anymore. Can someone please point me in the right direction?
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[General] FreelyCall down?
As of 4:00 PM, Central Standard Time, U.S., none of my devices will register with the usually reliable FreelyCall VOIP service. Nothing in my configuration has changed. Is anyone else experiencing problems with this provider? Thank you.
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