After several hours of troubleshooting I decided to ask you for help.
I have an Obi200 and my VOIP provider is Freephoneline (FPL).
I used this Obi200 adapter for the past 3 years with no problem.
Few weeks ago I moved to a new condo (where I have a new ISP) and since then I have issues with my FPL phone. I can make outgoing calls fine but I cannot get incoming calls.
When someone calls me it rings once and then the call goes to voicemail.
After I searched on the internet I was thinking that the issue is caused by the SIP ALG parameter. As I had a router which didn't have the option to disable this SIP ALG (Dlink DI-64) I decided to change the router.
So now I use a Trendnet TEW-639GR. However this router also doesn't have the option to disable/enable ALG.
And the problem is not solved, I cannot get any incoming calls.
Also I did another test, from another Obi adapter (Obi100) I was able to call my device (Obi200) and it rings OK, just the ring tone is different.
What do you think I should do? Find a 3rd router?
Do you have any suggestions?
Thanks!
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[Other] Freephoneline - no incoming calls - pls help
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[Voip.ms] Using Voip.ms to bridge calls to my IP devices
The subject line is a little vague. Essentially, I am considering porting my cell number from at&t to Voip.ms, and using it as a DID whereby calls inbound to this number are directed to certain subaccounts on my voip.ms account. i.e. the subaccounts (or extensions, if you will) for my personal IP phone, soft phones, and of course, the IP phone client that'll have to run on my mobile phone.
Kind of like making my own google voice, without using google voice. The only thing I haven't solved yet is placing outbound calls. To even use data for anything, be it browsing the web or receiving calls over an app, I'd have to have service on the phone. Which means giving it a new number. For those who have done this, have you found it possible at all to get your wireless carrier to change the outbound caller ID number? I.e. being able to use the native app on a phone for it's intended purpose, but the caller ID show up as my actual number, not the number given to the SIM card in the phone.
Or would I be stuck only being able to place calls via the SIP app of my choosing?
Am I crazy, is this a bad idea w/ trouble written all over it? Or something that's fairly common?
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Google Voice is getting an upgrade
Google’s somewhat aging VoIP calling service, Google Voice, is preparing to roll out a significant update, the company has confirmed. Several Google Voice users this week reported seeing an upgrade link touting “the new Google Voice” on the web version of the service, along with a link that would let them try it out. Clearly, the message was posted too early because the upgrade didn’t come through when the link was clicked.
However, its existence has been an encouraging sign for Google Voice customers, many of whom have felt as if the service has been abandoned in favor of Google’s many newer efforts in the communications space, including Hangouts, and more recently, messaging apps like Allo and Duo....
Google’s failure to continue investing in the space has allowed other voice calling apps the ability to gain ground. Not only are voice messaging apps a popular category today, the idea to give users the ability share a public phone number that can ring them anywhere, on a line that has its own voicemail and filtering options, is now something other startups, like Burner or newcomer Listen, are handling instead.
https://techcrunch.com/2017/01/10/google-voice-is-getting-an-upgrade/
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OBi200 is $10 off at Amazon.com until end of day January 30.
https://www.amazon.com/dp/B00BUV7C9A?m=A4VT9X8JBKS16
Click "View promotion details", then "Redeem".
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[PBX] SIP Trunk Provider US
Hi,
I am looking for a GOOD Sip Trunk provider in the US, that meets the following criteria:
- Offers unlimited calling within US and Canada
- Allows to add single trunks on a need basis
- Reasonable price
Thanks,
Daniel
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[Equipment] HTek announces new UC923 IP phone
HTek has just announced a new smaller, less expensive Color Gigabit IP phone, the UC923. It should be available around late February from HTek's US distributors Telephony Depot and Teledynamics.
HTek is also again offering their $99 reseller bundle of their flagship UC926 and UC924 IP phones from both distributors for qualified resellers/ITSPs. If you're not happy with your current IP phone vendor, or are looking for something more value-oriented, perhaps you should check out HTek. (Full disclosure: I'm a consultant working with them in the US).
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[General] Calls Drop when placed on hold
I have an odd one that we have been dealing with. We have been having complaints from a customer that if they are on a conference call and place it on hold, when they try to go back to the call the call drops. Now I have been told that it has something to do with re-invites that we are sending, but I am not 100%. Has anyone else seen this or know what is going on? We are using Asterisk.
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[CallCentric] questions on callcentric
few questions i have on them....
1. do they have a cloud hosted pbx solution?
2. do they support RDNIS?
3. do they have automatic call recording that gets every call incoming and outgoing?
4. do they support patching 2 calls together? (example...i have an incoming call that i answer, then i pick up a line and dial a third party, tell them that they are connected and let them start talking to each other, then i hang up)
thanks much
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Voxbeam offering 10 free DIDs
The promo came by email; they granted permission to repost it in this forum: "We are giving away 10 free On-net USA or UK numbers. For US, Boston and New York are available. For UK, London 0208 / Manchester or Sheffield are available."
You can get up to 10 numbers in one city, or mix and match across the available locations. The service is completely free, with no setup or monthly charges and no charge for incoming calls. Each DID is hard-limited to two channels; a 10,000 minute monthly limit applies to the entire offer.
The promotion is not automated -- send an email to dids@voxbeam.com mentioning "FREE USA and UK DIDs" with your specific request. In my case, I asked for a 1-646-740-XXXX, a 1-617-372-XXXX and a 44-208-XXX-XXXX. Of course, you can only get numbers in blocks assigned to Voxbeam and cannot pick specific numbers. You also need to provide the address where they will initially send calls, e.g. myserver.mydomain.com:5060, though you can change this later from the customer portal. They provisioned the DIDs within a few hours and they all work fine.
I don't know whether an unfunded account (you get a small credit at signup) is eligible, but suspect that if you ask nicely your request will be granted.
If appears that you can do failover via DNS SRV (not tested) and you can of course do manual failover by changing the destination host on the portal or via DNS. You can specify multiple destinations on the portal, but none of the options are useful for failover, unless the servers are always 'hot'.
There is no incoming CNAM and I don't know whether outbound CNAM is available, or whether a charge applies.
Blocked caller IDs do get though, though you will need to look in both P-Asserted-Identity and Remote-Party-ID for the calling number.
RDNIS is supported but my attempt at testing it failed for reasons unrelated to the DID.
No SMS, though they may allow a split port.
Offer expires 12/31/2016, though you should probably enter your request well before the holidays.
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voip.ms vs Comcast porting battle - any way to win?
I'm helping a customer move to voip.ms from Comcast. They have 2 numbers they want to move, and the port got rejected because the info on the request doesn't match the CSR. Comcast tells the customer that the new provider can request a CSR to make sure the info matches, and that they don't provide it to end users. They said it a second time on a separate call, and checking online seems to suggest the same: https://www.google.com/search?q=comcast+port+out+CSR&ie=utf-8&oe=utf-8 (results 1-4)
"Comcast does not require a NLSP to obtain a CSR for simple port requests prior to submitting an LSR via the Trading Partners website. CSRs simple port requests can be requested via the “Create CSR” link on the Trading Partners website.
To obtain a CSR for complex service, the NLSP is directed to call Comcast’s voice toll free support number at 1-877-547-7230 and select option 2."
voip.ms says they need the CSR to proceed and cannot request it as they are not a CLEC (after the customer told them what Comcast told them, including the phone number to request the CSR) - but what about Onvoy or whoever voip.ms is using, can they request it, and how can the client tell them (voip.ms) this (to follow Comcast's guideline to get the CSR)? They've told them what Comcast told them, along with the phone number, but they don't want to budge.
I know porting sucks but this is pretty bad. I like voip.ms - but any recommendations for another provider that's easier with porting? Or is Comcast wrong?
Any suggestions would be appreciated.
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TelexFree VoIP pyramid scheme: Feds find $20 million in mattress
When federal authorities raided an apartment in Westborough, Mass., earlier this month, they found money hidden under the mattress — approximately $20 million.
Photos show a box spring stuffed with bricks of cash that were seized during a Homeland Security investigation linked to a pyramid scheme involving a company called TelexFree.
Agents found the money after they arrested a Brazilian national named Cleber Rene Rizerio Rocha, who was charged with one count of conspiring to commit money laundering. The 28-year-old admitted to federal agents that he was in the U.S. to facilitate a money transfer to a founder of the company, court documents state.
TelexFree, which sold voice-over-Internet phone service, "was really a massive pyramid scheme," according to court documents. "It make little or no money from selling VOIP, but took in millions of dollars from people signing up to receive financial bonuses from advertising and recruiting."
http://www.npr.org/sections/thetwo-way/2017/01/24/511451628/feds-find-20-million-hidden-under-a-mattress-in-massachusetts
Previous coverage in DSLR:
http://www.dslreports.com/forum/r29208477-VoIP-Ponzi-Scheme-shut-down-in-USA-sets-up-in-Canada
http://www.dslreports.com/forum/r29442013-VoIP-company-investigation-in-US-and-Brazil-pyramid-alleged
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[General] TollFreeProxy Taking 40 Seconds to Connect
Got TollFreeProxy as a trunk on AsteriskNow/FreePBX and when making a call it's dead silence for 40 seconds.
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Trigger call from Conference
I have a Snom320 connected to a base station radio to stream radio traffic to a conference room. It's a direct IP call which is connected 24/7.
I would like to trigger a call whenever there is audio on the conference.
I have turned on Talker optimization. "With talker optimization, Asterisk treats talkers who are not speaking as being muted, meaning that no encoding is done on transmission and that received audio that is not registered as talking is omitted, causing no buildup in background noise."
I would like to trigger a call at the same time the conference unmutes the caller, only if there was no call triggered within 1-2 minutes.
Is this possible?
I'm addition to this, does anyone know where I can pickup a radio to phone gateway?
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Does voip.ms support/enable the phone intercom feature?
Is anyone out there successfully using their phones' intercom feature while using voip.ms as their cloud hosted provider? Specifically, can this be done with Yealink T2 series phones? How about Polycom VVX phones? I tried asking tech support at voip.ms, but their best answer was that I'd need to sign up for service, buy a couple of phones, and try it out!
To clarify, I am interested in intercom -not paging. Thanks!
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"These are the most beautiful phones I've ever used in my life"
said President Trump.
https://www.nytimes.com/2017/01/25/us/politics/president-trump-white-house.html
I believe that the one on the right is a Cisco 7975G. Anyone recognize the left one?
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calls dropping seemingly at random
Looking for someone that might be able to help. I have a FreePBX system version 10.13.66-17 (asterisk 13.12.2). Calls go over an Anveodirect trunk.
Randomly I will get reports from users having their calls drop. Nothing seemingly in common although I'd say most are less than 10min into the call. I have someone reporting to me when the calls drop (someone I trust to provide me full details and not just "call drop"). She had a drop this morning while on a conference call after 7min 46sec into the call. She called back into the call without issue.
I pulled the sip log from Anveo, sanitized our WAN IP and Phone#, and was hoping someone might look at it and see if they see any issues.
If I've missed sanitizing anything let me know.
/*<<<|src_WAN_IP:5060 @ 2017-01-26 14:59:46 */INVITE sip:prefix+number@sbc.anveo.com:5060 SIP/2.0Via: SIP/2.0/UDP src_WAN_IP:5060;branch=z9hG4bK33548cbcMax-Forwards: 70From: <sip:src_phone#@src_WAN_IP>;tag=as58b7ff93To: <sip:prefix+number@sbc.anveo.com:5060>Contact: <sip:src_phone#@src_WAN_IP:5060>Call-ID: 65b4c9a825e67ac95b2146c066d49127@src_WAN_IP:5060CSeq: 102 INVITEUser-Agent: FPBX-13.0.190.7(13.12.2)Date: Thu, 26 Jan 2017 14:59:45 GMTAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timerContent-Type: application/sdpContent-Length: 252 v=0o=root 677597822 677597822 IN IP4 src_WAN_IPs=Asterisk PBX 13.12.2c=IN IP4 src_WAN_IPt=0 0m=audio 10064 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=maxptime:150a=sendrecv /*>>>|src_WAN_IP:5060 @ 2017-01-26 14:59:46 */SIP/2.0 100 TryingVia: SIP/2.0/UDP src_WAN_IP:5060;branch=z9hG4bK33548cbcFrom: <sip:src_phone#@src_WAN_IP>;tag=as58b7ff93To: <sip:prefix+number@sbc.anveo.com:5060>Call-ID: 65b4c9a825e67ac95b2146c066d49127@src_WAN_IP:5060CSeq: 102 INVITEServer: Anveo CallcontrolContent-Length: 0 /*>>>|src_WAN_IP:5060 @ 2017-01-26 14:59:47 */SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP src_WAN_IP:5060;branch=z9hG4bK33548cbcFrom: <sip:src_phone#@src_WAN_IP>;tag=as58b7ff93To: <sip:prefix+number@sbc.anveo.com:5060>;tag=fc5806494c4b9b2b3ed7c2507dda75eeCall-ID: 65b4c9a825e67ac95b2146c066d49127@src_WAN_IP:5060CSeq: 102 INVITEServer: Anveo CallcontrolContent-Type: application/sdpContent-Length: 223 v=0o=IRISMSC2 12772 12850 IN IP4 209.58.101.249s=sip callc=IN IP4 74.120.95.52t=0 0m=audio 28562 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=sendrecva=maxptime:20 /*>>>|src_WAN_IP:5060 @ 2017-01-26 14:59:47 */SIP/2.0 200 OKVia: SIP/2.0/UDP src_WAN_IP:5060;branch=z9hG4bK33548cbcFrom: <sip:src_phone#@src_WAN_IP>;tag=as58b7ff93To: <sip:prefix+number@sbc.anveo.com:5060>;tag=fc5806494c4b9b2b3ed7c2507dda75eeCall-ID: 65b4c9a825e67ac95b2146c066d49127@src_WAN_IP:5060CSeq: 102 INVITEServer: Anveo CallcontrolContact: Anonymous <sip:50.22.101.14:5060>Content-Type: application/sdpContent-Length: 223 v=0o=IRISMSC2 12772 12850 IN IP4 209.58.101.249s=sip callc=IN IP4 74.120.95.52t=0 0m=audio 28562 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=sendrecva=maxptime:20 /*<<<|src_WAN_IP:5060 @ 2017-01-26 14:59:47 */ACK sip:50.22.101.14:5060 SIP/2.0Via: SIP/2.0/UDP src_WAN_IP:5060;branch=z9hG4bK3df434b2Max-Forwards: 70From: <sip:src_phone#@src_WAN_IP>;tag=as58b7ff93To: <sip:prefix+number@sbc.anveo.com:5060>;tag=fc5806494c4b9b2b3ed7c2507dda75eeContact: <sip:src_phone#@src_WAN_IP:5060>Call-ID: 65b4c9a825e67ac95b2146c066d49127@src_WAN_IP:5060CSeq: 102 ACKUser-Agent: FPBX-13.0.190.7(13.12.2)Content-Length: 0 /*>>>|src_WAN_IP:5060 @ 2017-01-26 15:04:47 */INVITE sip:src_phone#@src_WAN_IP:5060 SIP/2.0Via: SIP/2.0/UDP 50.22.101.14:5060;branch=z9hG4bK59544a33b94f76fc9822642b7f1475a4;rportMax-Forwards: 70From: <sip:prefix+number@sbc.anveo.com:5060>;tag=fc5806494c4b9b2b3ed7c2507dda75eeTo: <sip:src_phone#@src_WAN_IP>;tag=as58b7ff93Call-ID: 65b4c9a825e67ac95b2146c066d49127@src_WAN_IP:5060CSeq: 100 INVITEContact: Anonymous <sip:50.22.101.14:5060>Expires: 300User-Agent: Anveo Callcontrolcisco-GUID: 4107689790-475060837-1947940984-413399427h323-conf-id: 4107689790-475060837-1947940984-413399427Content-Type: application/sdpContent-Length: 223 v=0o=IRISMSC2 12772 12850 IN IP4 209.58.101.249s=sip callc=IN IP4 74.120.95.52t=0 0m=audio 28562 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-15a=sendrecva=maxptime:20 /*<<<|src_WAN_IP:5060 @ 2017-01-26 15:04:47 */SIP/2.0 100 TryingVia: SIP/2.0/UDP 50.22.101.14:5060;branch=z9hG4bK59544a33b94f76fc9822642b7f1475a4;received=50.22.101.14;rport=5060From: <sip:prefix+number@sbc.anveo.com:5060>;tag=fc5806494c4b9b2b3ed7c2507dda75eeTo: <sip:src_phone#@src_WAN_IP>;tag=as58b7ff93Call-ID: 65b4c9a825e67ac95b2146c066d49127@src_WAN_IP:5060CSeq: 100 INVITEServer: FPBX-13.0.190.7(13.12.2)Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timerContact: <sip:src_phone#@src_WAN_IP:5060>Content-Length: 0 /*<<<|src_WAN_IP:5060 @ 2017-01-26 15:04:47 */SIP/2.0 200 OKVia: SIP/2.0/UDP 50.22.101.14:5060;branch=z9hG4bK59544a33b94f76fc9822642b7f1475a4;received=50.22.101.14;rport=5060From: <sip:prefix+number@sbc.anveo.com:5060>;tag=fc5806494c4b9b2b3ed7c2507dda75eeTo: <sip:src_phone#@src_WAN_IP>;tag=as58b7ff93Call-ID: 65b4c9a825e67ac95b2146c066d49127@src_WAN_IP:5060CSeq: 100 INVITEServer: FPBX-13.0.190.7(13.12.2)Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGESupported: replaces, timerContact: <sip:src_phone#@src_WAN_IP:5060>Content-Type: application/sdpContent-Length: 252 v=0o=root 677597822 677597822 IN IP4 src_WAN_IPs=Asterisk PBX 13.12.2c=IN IP4 src_WAN_IPt=0 0m=audio 10064 RTP/AVP 0 101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20a=maxptime:150a=sendrecv /*>>>|src_WAN_IP:5060 @ 2017-01-26 15:04:47 */ACK sip:src_phone#@src_WAN_IP:5060 SIP/2.0Via: SIP/2.0/UDP 50.22.101.14:5060;rport;branch=z9hG4bKf3c098d6f2716f96259b2aa8be87d4b6Max-Forwards: 70From: <sip:prefix+number@sbc.anveo.com:5060>;tag=fc5806494c4b9b2b3ed7c2507dda75eeTo: <sip:src_phone#@src_WAN_IP>;tag=as58b7ff93Call-ID: 65b4c9a825e67ac95b2146c066d49127@src_WAN_IP:5060CSeq: 100 ACKUser-Agent: Anveo CallcontrolContent-Length: 0 /*>>>|src_WAN_IP:5060 @ 2017-01-26 15:07:33 */BYE sip:src_phone#@src_WAN_IP:5060 SIP/2.0Via: SIP/2.0/UDP 50.22.101.14:5060;branch=z9hG4bK26cd743bb2387c549466eb80dcf59343;rportMax-Forwards: 70From: <sip:prefix+number@sbc.anveo.com:5060>;tag=fc5806494c4b9b2b3ed7c2507dda75eeTo: <sip:src_phone#@src_WAN_IP>;tag=as58b7ff93Call-ID: 65b4c9a825e67ac95b2146c066d49127@src_WAN_IP:5060CSeq: 101 BYEContact: Anonymous <sip:50.22.101.14:5060>User-Agent: Anveo Callcontrolcisco-GUID: 4107689790-475060837-1947940984-413399427h323-conf-id: 4107689790-475060837-1947940984-413399427Content-Length: 0
This particular conference room number is something that our users are calling into a LOT in the last few months and it isn't uncommon to have 3-4 users dialed into this number for multiple hour conferences. And since this began I've been getting these reports from those users of the drops. In fact 1 user will drop while the others are fine. Dropped user calls back in and is fine for the rest of the call.
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Beware FreePBX 13 upgrade if you use "firstnotonphone" strategy
Don't have a lot of time to go into detail right now, but just a quick heads-up to anyone thinking of upgrading from an earlier version of FreePBX to FreePBX 13 - it appears the "firstnotonphone" ring strategy is broken in follow-me's and ring groups. Apparently it works fine in FreePBX 11, but not in FreePBX 13. So if you depend on that strategy, you might want to hold off on any planned upgrades, or at least test it on a separate server before upgrading a system you depend on to handle calls.
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Issabel (elastix 2.5 fork) with thespout or flowroute from Canada
I am trying to set up Issabel with thespout but since thespout is not answering their tech support messages I am wondering if flowroute will work with issabel ? Issabel is the fork of elastix 2.5 (so presumably compatible?) with flowroute ?
Also anyone used flowroute from canada ?
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[PBX] Mitel and snom-Pbx and sip
Tech guys been here 3 days can't even make a call. Something not right..
Ok here we go...
I am getting involved because they asked me to find out what's wrong and fix.
My first question
Local host Pbx
ISP Comcast infinity business
Installing sip conference phone
Do we need to register our pbx with a sip company or something Simlair?
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Does Anveo's Login Page Have A Security Problem?
I normally use Firefox to log into Anveo. I tried it just now and landed on a Firefox Insecure Connection splash page. The reason shown is:quote:www.anveo.com uses an invalid security certificate. The certificate is not trusted because the issuer certificate is unknown. The server might not be sending the appropriate intermediate certificates. An additional root certificate may need to be imported. Error code: SEC_ERROR_UNKNOWN_ISSUER
I've never seen this problem before. I don't know if there is a new problem at Anveo or if Firefox tightened their security testing criteria. There are no security warnings when using Chrome.
Using SSL Labs' Server Test, Anveo has an F rating due to one particular vulnerability.quote:This server supports 512-bit export suites and might be vulnerable to the FREAK attack. Grade set to F.
https://www.ssllabs.com/ssltest/analyze.html?d=anveo.com
Edit: I just noticed that the security certificate validity date is from yesterday, so there must be an error introduced when updating the certificate:quote:Valid from Thu, 26 Jan 2017 00:00:00 UTC
Hope they can fix this soon.
Can anyone elaborate on these issues?
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