I foolishly applied provisioning from this file: http://fw.obifirmware.com/EnableGVSIPWithoutOBiTALK.xml, which as some say applies to SP1 service only... In SP1 I had configuration for my local FreeSWITCH. Now connection to it shows "Connecting to 192.168.1.15;Token Error". There should be no token. I'm using a regular password only.
How to revert to the previous configuration and remove "Token Error"?
And second question. I applied sp4.xml from local TFTP server. It allows me to set SP4 as GV connector. And it was fine, until I started to restore recent backup to fix the first problem. Now on SP4 I'm getting "Connecting to 64.9.240.172;Token Error" :(
Is there any way to fix those problems?
↧
Problems with OBi200
↧
GVSIP configuration helper for FreePBX users
This is a very simple script to help if you are creating multiple GVSIP accounts in FreePBX. You must replace the three items at the top of the file with values you use for oAuth Client ID and Client Secret (you only need one working pair of those, no matter how many accounts you have), and also a one word name for your system (can be anything, when you run it once you'll see how it's used). It then walks you through getting an oAuth Refresh Token (instructions mostly adapted from RonR's previous post), asks for a bit of additional information, and then gives you a suggested configuration. There is no error or sanity checking; "Garbage In - Garbage Out" applies here. This sort of assumes you have created at least one account already and therefore the only configuration information you need is for the new accounts. If you have never created any GVSIP account before, you should probably follow naf's instructions, at least for the [global] context. And also you may need to follow RonR's instructions to get an oAuth Client ID and Client Secret, if you don't already have those.
I do things a little differently than others who have posted scripts and I think you will find that if you have several accounts, the way I do things makes it a little bit easier to keep track of your accounts. It works for me, but you are of course free to modify it as you see fit. No error checking, bells, whistles, or frills. Give it a dry run with dummy data to get a sense of what it does. If you're not satisfied, feel free to return it for a full refund. :D
One hint to others who write scripts: No, you really don't need any added dialplan in extensions_custom.conf, not to make GVSIP work anyway. If you are using lines of custom dialplan to strip the +1 from the start of the incoming Google Voice number then forget about it and run this script once and it will show you the right way to do that, using the from-pstn-e164-us context in your trunk settings.
EDIT: Added reminders to click Submit after creating pages in the FreePBX GUI. Also edited to make it more clear which token you are looking for on the page, since I found out the hard way that if you make a mistake and copy the access token rather than the refresh token, it actually will work for a short time (probably an hour) but then the access token expires and it stops working. You don't want to find out you've used the wrong token after you have done several accounts. Don't ask me how I know that, I don't want to talk about it! :(
EDIT 2: Edited to use transport=0.0.0.0-tls rather than transport=transport_tls from naf's original instructions. If this isn't changed, you will get errors when you set tls - 0.0.0.0 - All to YES on FreePBX's Chan PJSIP Settings page (thanks to uid://1917447 for pointing this out to me in the other thread). If you have previously used this script and need to make that change in your existing pjsip_custom_post.conf file, just run
sed -i -e 's/transport=transport_tls/transport=0.0.0.0-tls/g' /etc/asterisk/pjsip_custom_post.conf
That should change all occurrences in the file. Then remove naf's original [transport_tls] context (all four lines) from your pjsip_custom_post.conf file, and set tls - 0.0.0.0 - All to YES on FreePBX's Chan PJSIP Settings page.
#!/bin/bashoAuthClientID="PUT YOUR OAUTH CLIENT ID HERE"oAuthClientSecret="PUT YOUR OAUTH CLIENT SECRET HERE"SystemName="PUT A ONE WORD NAME FOR YOUR SYSTEM HERE (NO SPACES!!!)"echo ""echo ""echo "GVSIP Config Helper"echo ""echo ""echo "Instructions to get Refresh Token:"echo ""echo ""echo "1. Log into the Google Voice account for which you need a token"echo ""echo "2. Go to https://developers.google.com/oauthplayground"echo ""echo "3. Click the gear icon in the top right of the page"echo ""echo "4. Check Use your own oAuth credentials"echo ""echo "5. Enter oAuth Client ID: $oAuthClientID"echo " and oAuth Client secret: $oAuthClientSecret"echo ""echo "6. Click Close"echo ""echo "7. Enter https://www.googleapis.com/auth/googletalk at Input your own scopes"echo ""echo "8. Click Authorize API"echo ""echo "9. Sign into your Google Voice account again if you are prompted to do so"echo ""echo "10. Click Allow"echo ""echo "11. Click Exchange authorization code for tokens"echo ""echo "12. Find the Refresh Token - it is the shorter of the two tokens returned, and appears at a lower position on the page."echo -n " Record the Refresh Token (without the quotation marks) in a safe place and enter it here: "read oAuthRefreshTokenecho ""echo ""echo "Now please enter the following additional information:"echo ""echo -n "Enter the Google Voice account name (the part before @gmail.com): "read AccountNameecho ""echo -n "Enter the 10 digit Google Voice telephone number: "read GVPhoneNumberecho ""echo -n "Enter the account user's name with NO SPACES: "read UserFirstNameecho ""echo -n "Enter the city associated with the Google Voice telephone number, with NO SPACES: "read PrefixCityecho ""echo ""echo "Add the following to the file /etc/asterisk/pjsip_custom_post.conf:"echo ""echo "[gvsip-$AccountName]"echo "type=registration"echo "outbound_auth=gvsip-$AccountName"echo "server_uri=sip:obihai.sip.google.com"echo "outbound_proxy=sip:obihai.telephony.goog:5061\;transport=tls\;lr\;hide"echo "client_uri=sip:$AccountName$GVPhoneNumber@obihai.sip.google.com"echo "retry_interval=60"echo "support_path=yes"echo "support_outbound=yes"echo "line=yes"echo "endpoint=gvsip-$AccountName"echo "contact_additional_params=obn=$SystemName-$UserFirstName-$PrefixCity"echo "transport=0.0.0.0-tls"echo "transport_reuse=no"echo "contact_user=$GVPhoneNumber"echo ""echo "[gvsip-$AccountName]"echo "type=auth"echo "auth_type=oauth"echo "refresh_token=$oAuthRefreshToken"echo "oauth_clientid=$oAuthClientID"echo "oauth_secret=$oAuthClientSecret"echo "username=$AccountName$GVPhoneNumber"echo "realm=obihai.sip.google.com"echo ""echo "[gvsip-$AccountName]"echo "type=aor"echo "contact=sip:obihai.sip.google.com"echo ""echo "[gvsip-$AccountName]"echo "type=endpoint"echo "context=from-pstn-e164-us"echo "disallow=all"echo "allow=ulaw"echo "allow=opus"echo "outbound_auth=gvsip-$AccountName"echo "outbound_proxy=sip:obihai.telephony.goog:5061\;transport=tls\;lr\;hide"echo "aors=gvsip-$AccountName"echo "direct_media=no"echo "ice_support=yes"echo "rtcp_mux=yes"echo "media_use_received_transport=yes"echo "outbound_registration=gvsip-$AccountName"echo ""echo ""echo ""echo "In the FreePBX GUI:"echo ""echo "Add a custom trunk and make the trunk name: GVSIP-$AccountName"echo "Set the Outbound CallerID to: $GVPhoneNumber"echo "In the Dial Number Manipulation Rules tab add the pattern: 1|NXXNXXXXXX and also optionally aaa+NXXNXXXXXX (replace aaa with the assumed area code for 7 digit calls)"echo "In Custom Settings add the Custom Dial String: PJSIP/\$OUTNUM\$@gvsip-$AccountName"echo "Click Submit."echo ""echo "Add an Inbound Route with the Description: Google Voice - $UserFirstName - $PrefixCity"echo "Set the DID Number to: $GVPhoneNumber"echo "Set Destination to the desired extension, ring group, IVR, etc."echo "In the Advanced tab, set Force Answer to YES if you don't want to use Google's Voicemail."echo "In the Other tab, optionally enable a CID Lookup Source, or Enable Superfecta Lookup and select a Superfecta Scheme (probably Default)."echo "Click Submit."echo ""echo "Don't forget to add or modify Outbound Routes to use the GVSIP-$AccountName trunk and click Submit after each."echo ""echo "Click the Apply Config button in FreePBX and wait for the page to reload. If the new account doesn't register within several seconds,"echo "then go to the Asterisk CLI and enter core restart now (this will interrupt any calls in progress) or core restart when convenient"echo "(which will delay the restart until there are no active calls)."echo ""echo ""
↧
↧
PIAF/3CX/Tokens etc. (OT SPLIT from another topic)
need I say PIAF is now a product of 3CX and we know who else is tied in there too as the forums are shared. At the order of 3CX those tokens could be eliminated.
Look how 3CX dismantled the ELestix commuity , even deleting old posts and information
I am NOT COMFORTABLE with 3CX having ANYTHING to do with it , not even vaguely.
↧
How to dial international with Obi200 and Google voice?
Im in the US. I used to just dial country code followed by the number when I used SPA-2102 with Simonics. But now that I moved to Obi200 I get "We could not complete your call. Please try again". Do I need to dial an exit code or something?
Domestic (in USA) calls work just fine.
↧
[Voip.ms] For purpose of call blocking, would legit CID ever start with 0?
Looking at my voip.ms call log today I see this: "Anonymous" (0000000000). I've got 280 call-filter rules and I guess none of them would have caught this. I don't have NPAN filter turned on because we sometimes get international calls to this DID. I'm wondering if there would ever be a situation where you'd have a valid CID number starting with 0 (zero) even for international callers.?
↧
↧
[Equipment] Linksys SPA2102 vs SPA3102 for placing outgoing calls
I have both a SPA2102 and a SPA3102.
I am not interested in using the FXO port on the SPA3102 and I know that the SPA2102 has two FXS ports (two phone lines) vs only one on the SPA3102.
I wonder if either or both the SPA2102 and SPSA3102:
1. Can make outbound calls w/o needing registration and, on the same line, receive SIP calls from another provider*?
2. Can make outbound calls using various providers by pressing #1, #2 etc?
(Is this called "gateway calling"?)
*) Specifically, I would like to send outgoing calls to an Obi202 and receive incoming calls from CallCentric.
↧
Offering Replacement for Simonics/Google Voice
Bill at Simonics offered a great service, too bad it is going away, as he has announced already.
I am working to rescue some folks that need to keep using a similar service that were previously using simonics, as well as a handfull of others that were on my own XMPP Google Voice services.
I have limited space for more, but because maintaing SIP registrations is much more resource intensive than XMPP , I can not do it at prices previously offered by Simonics. It allows only a single registration which connects directly to your GV trunk.
you can choose either:
$33.00 USD one time fee
Or
$12.00 USD Per Year
You can sign up here:
http://googlevoicephone.com/byod.php
Once you sign up please send your GV Phone number to the paypal email address.
You will then be sent a ClientID.ClientSecret that you can use on the Google Playground to get your "refresh token" . This is currently the ONLY way I can activate any services.
↧
ObiHai 1022/1032/1062 and obifirmware
I'd like to flash an OBIHAI 1022 with firmware from the obifirmware site.
It states that "wont flash from stock GVSIP due to x509 signature check in stock upgrade. you were warned"
The release notes state that it has patched obiapp to bypass x509 signature check for new fw...
Are there instructions for flashing available on the web that may explain?
Any help is appreciated.
Thanks in advance.
Rich
↧
Question about call screening.
I get a fair amount of robo calls on my cell phone. Is there a way for someone to call my cell phone, which forwards to my GV voice and have it screened by Google voice, and then forward the call back to my cell phone?
I know it sounds complex, but I thought google voice screening was a neat idea, since the robo calls would have to say their name before being put through the call.
↧
↧
Almost free PBX in the Google Cloud
I've recently have been experimenting with Google Cloud Platform and am quite impressed. Their scheduling latency performance of their shared-core VPS offerings is far superior to the cheap stuff on LowEndBox and LowEndStock (IMO unsuitable for production PBXes) and at least equals that of RentPBX, Linode, DigitalOcean, Rackspace, RamNode, etc.
The platform includes hardware firewall protection, snapshots, ease of creating / moving / destroying servers on demand, as well as many other management features.
In addition to their one-year free trial with $300 credit applicable to most services, many services offer truly free usage (not a trial) within certain limits. For details, see https://cloud.google.com/free/ . These limits are sufficient to run a small PBX indefinitely, except for egress bandwidth priced at $0.12/GB after the first GB/mo. For example, if talking 2000 minutes monthly, once your free year is up you'll be billed ~$0.25/mo.
Here is an easy way to try FreePBX on GCP: I took the instructions at https://wiki.freepbx.org/display/FOP/Installing+FreePBX+14+on+CentOS+7 and assembled them into a script, adding modifications for GCP and automating all interactive steps.
Once you've signed up, go to https://console.cloud.google.com .
Select Compute Engine -> VM instances.
Click Create Instance.
Choose a name, select zone; free options are us-east1 (South Carolina), us-central1 (Iowa) or us-west1 (Oregon).
If you're not sure, check at http://www.gcping.com/ (test takes about 30 seconds to run, actual ping time is ~12 ms less than HTTP time reported).
(Leave default Machine type for faster build -- we will change to a free one later.)
Change Boot disk to CentOS 7, press Select.
Under Firewall, click Allow HTTP traffic. Click Create.
On the VM Instances page, under Connect, click SSH (need to click twice if blocking popups). A console window will open (takes ~30 seconds).
Paste the code below into the console window.
sudo -icurl http://x-sc.com/fpbx14.sh > fpbx14.shchmod +x fpbx14.sh./fpbx14.sh When the script is done (about 20 minutes), "You have successfully installed FreePBX" should appear in green. If a minute goes by with no activity, it has probably stopped on an error.
Open the external IP address in your browser.
Set up admin account as requested, log in, set timezone, etc. Apply Config.
With luck, there will be no errors shown in the System Overview.
Go to Settings -> Asterisk SIP Settings. Click Detect Network Settings. The External Address field should populate. Click Submit then Apply Config.
In the cloud Console tab (not the SSH console window) go to VPC network -> Firewall rules, click Create Firewall Rule.
Give it a name e.g. sip. Under Target tags, type a tag name, e.g. sip.
Under Specified protocols and ports enter "udp:5060; udp:5160; udp:10000-20000" (without the quotes and assuming default FreePBX settings). Under Source IP ranges, enter 0.0.0.0/0 (or a restricted range if desired), then click Save.
Go to VPC network -> External IP addresses. For your instance, change the Type from Ephemeral to Static. Give it a name, e.g. mypbx. Click RESERVE.
Go back to Compute Engine -> VM instances. Select your instance, click EDIT. Under Network tags add sip or whatever tag you chose above. Click Save.
Reboot (needed after network settings changes, also to confirm that automatic startups are working). You can reconnect to SSH server after one minute.
In the FreePBX UI, configure an extension. Set up a VoIP device or app to use it.
Call *43, confirm that the echo test works and latency and quality are ok.
In the FreePBX UI, configure a trunk and an outbound route.
Confirm that you can make a call and that latency and quality are ok.
Restrict access to the FreePBX GUI. For example, remove the http-server tag and add a rule that only grants access to your PC.
Close the SSH console window.
In the Console, stop your instance. Go to Snapshots. Click CREATE SNAPSHOT.
Give your snapshot a suitable name, e.g. as-built. Under Source disk, select your instance's boot disk. Click Create.
Edit your instance, change Machine type to micro (will become free if in qualifying zone), click Save.
Start your instance up again.
In the FreePBX UI, go to Module Admin. Click Check Online, select desired Repositories, install modules as desired.
Configure PBX for your application.
Take another snapshot.
Enjoy.
I'm aware of two problems with this build:
1. It can't send mail (such as voicemail notifications), because Google blocks outgoing ports 25, 465 and 587. My mail provider, Fastmail, has a proxy that avoids this issue. I'll try some of Google's suggested general purpose workarounds and will post an update if successful.
2. It crashes after configuring a Google Voice account. An apparently good motif.conf is created but Asterisk segfaults after reading it. I had the same problem with a FreePBX 13 / Asterisk 13 build. If you just install the FreePBX module without an account (creating dummy motif.conf and xmpp.conf files) and reboot, Asterisk will read those and start running the respective modules. I have no idea what may be wrong and would appreciate it if a GV expert could take a look. Create a snapshot just before attempting to add a GV account.
↧
Any method for testing Caller ID delivery?
Hello VOIP Forum brain trust,
Anyone have ideas about how to verify callerID delivery to global mobile phones?
Back story:
We've recently encountered some situations in the EU where callerIDs that originate in the EU do not deliver properly despite using reputable providers that have contractual obligations to deliver these. The providers deliver non-EU CallerIDs but not local ones and I believe are honestly unaware themselves of when this is failing.
We're looking for some way to test for this before it affects a customer.
Thanks,
Sam
↧
[Future9] Future 9 down?
I can call out but those who call me, including myself from my cell, get a busy signal. Going to contact F9 myself now.
↧
[Equipment] Advisability of turning SIP phones' and ATAs' power on and off?
Good day.
For over thirty years, I haven''t turned the power off on a computer without first shutting it off internally.
For about two years, I've been using a Yealink T22P SIP phone in my office. Its power adapter is plugged in to a switched power strip. I turn the power strip on in the morning and turn it off in the evening. To date, I've had no problems with the phone.
Just finished a similar set-up at home for an OBi1022 SIP phone and an OBi302 ATA. Because both devices have seldom had the power turned-off, I'm somewhat apprehensive about turning the power strip off when I go to sleep.
Having enough knowledge to be dangerous, I surmise that the firmware, being read only, should not be affected by a power interruption. I'm not so sure, though, about mounted writable RAM sectors.
Is this more or less correct or are their other things I should be aware of (and should not turn the power off)?
Thank you.
↧
↧
Obi200 - different providers for incoming/outgoing?
I suspect there will be a simple answer to this question, but I am not very VOIP-savvy, despite having used voip for several years. Hopefully someone here can help me out.
1. I have an Obi 200, currently configured with a voip.ms sip connection only. I would like to reconfigure it so that the connected phone will automatically use freephoneline for outgoing calls, and voip.ms for incoming calls. How do I go about doing this? Does the freephoneline connection need to be on SP1 to be dialled automatically, or is there some digit map/dialling plan I can use to accomplish this if FPL is on SP2?
2. Also, can I use the same FPL credentials on 2 different ATA's, or is this going to cause one ATA not to function properly? I do not use the FPL connection for any incoming calls, only outgoing.
3. Can I (and if so, how) change the outgoing caller ID on the FPL connection (to be used for outgoing calls) to match the number of my voip.ms connection that I use for incoming calls?
Thanks in advance.
↧
[General] Vonage grandstream ht801
Please can someone help.
Had anyone else experienced the receiver of a call hearing beeps.
Complained to Vonage for 2 weeks, with no improvement.
Also, does anyone know the password to get to advanced settings on ht 801 box.
↧
Unable To Make Outgoing Call w/Google Voice
Up until last Friday I was able to use my obihai 110 and a work around(my google voice number+2+number dialed+#) to make outbound calls.Now when I dial my google voice number I get this message: "the number you are trying to call is rejected by the service provider reason 500."
Anyone have this issue as well,if so were you able to resolve it and how?
Thanks
↧
Cloud Hosted GVSIP (Future)
We plan to document a hosted GVSIP setup next week that will run on any cloud platform and let users connect a SIP trunk from any VoIP platform to take advantage of GVSIP. In short, you can set up your own Simonics gateway clone for about $1 a month.
↧
↧
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).
↧
Obi202 naf firmware local config guide.
Hello all.
Old Obi110 stopped working. I decided to get an Obi202 so that I could get 2 GV numbers. on 2 phone lines.
I do not want to be beholden to the ObiHai portal.
Is there a step by step guide for installing naf's firmware to the device AND configuring it locally? (The guide I saw only covered installing the firmware and OAUTH credentials).
Does the firmware support 2 phone lines?
Thank you.
↧
[Unlock] New Grandstream ATA used by Vonage [Unlocked]
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.
↧
