how to setup freephone line in obi202
SteveInWA:
Forum member Mango is the Freephoneline/Fongo expert; maybe he's lurking and will respond.
I do know that OpenDNS can cause problems with OBis. Try using Google's high-performance DNS servers instead:
8.8.8.8 primary
8.8.4.4 secondary
Faisi:
Hello everyone, I made a little progress.
Help me out of this situation please, Now I am again blocked from another problem.
Outgoing calls works from SP1 = Google Voice = without adding 1 (I can reply using the Phone1 Handset)
Outgoing calls are working from SP2 - FPL with the following restrictions( I cannot reply using Phone 2 Handset)
If replied to the same number when a call is received, call ended with error with reason 484.
If added 1 in front(of the received call), call ended with reason 480
If not added 1 in front(of the received call), call ended with reason 484
If called with **21<then number> it works, I don't want to write down the received call and then press **21 then number every time.
Question is how can I hide pressing **21 ? and If I am able to reply directly from Phone2 Handset, same as Phone1 handset ?
I don't want to redirect through SP1 google voice number either, I want a home phone which is a custom number(FPL) to be able to make and receive calls same as used to be before porting ?
Can you help me out here, i believe digimaps or outgoing rule sets have to be updated, I am very very new to this and don't know how to fix this...
here are the further setting:
Inline image 1
Brisk:
Quote from: Faisi on February 23, 2016, 08:41:59 am
Question is how can I hide pressing **21 ?
It may have saved you some headaches had you provisioned Freephoneline on SP1 and GV on SP2 instead of the reverse.
You need to enter the Obitalk Expert Configuration menu.
Go to Physical Interfaces>>PHONE2 Port>>PHONE Port>>PrimaryLine
Then select SP2 Service for PrimaryLine. Save your settings.
It's doubtful that dialing 1 matters. If it does, you should post your ITSP Profile B's Digitmap, Phone2 Port's Digitmap, and Phone2 Port's Outboundcallroute for others to examine.
That's not Fongo, by the way. It's Freephoneline. Fongo is not a BYOD service. They're both divisions of Fibernetics, which is a Canadian CLEC.
Definitely do not port forward, unless you like to invite skiddies into your LAN.
For future reference, voip4.freephoneline.ca:6060 is intended for Rogers' integrated modem and router devices with SIP ALG settings hidden. To make matter worse, those Hitron devices have SIP ALG enabled with no way of disabling that setting without Rogers disabling it for the customer. I doubt having SIP ALG enabled and hidden from the user is a problem that is isolated to these Hitron products that Rogers distributes. If a user has his or her own router, it's advisable to put Rogers' modem in bridge mode. From there, should one way audio issues still arise after testing, the user should attempt to disable SIP ALG or SPI in his or her own router. If those options aren't available, switching ProxyServer to voip4.freephoneline.ca:6060 from voip.freephoneline.ca or from voip2.freephoneline.ca should be the next test. Additionally, changing X_UserAgentPort in an Obihai ATA to a value higher than 20000 may help circumvent a SIP ALG issue, depending on the router being used.
Perhaps even more perplexing, Freephoneline doesn't force proxy RTP at the switch. Without X_DiscoverPublicAddress and X_UsePublicAddressInVia enabled in an Obihai ATA, it's possible to get RTP packets sent pointlessly to LAN IPs, 192.168.x.x perhaps, instead of WAN IPs with Freephoneline.
Webslinger:
A lot of useful information can be found over in these threads:
http://forums.redflagdeals.com/newegg-obihai-obi200-ata-49-99-1-50-ehf-5-99-shipping-tax-2046832/
http://forums.redflagdeals.com/freephoneline-ca-free-local-soft-phone-line-lifetime-voip-821229/
Freephoneline's recommended settings can be found here: http://support.freephoneline.ca/hc/en-us/articles/212430746-VoIP-Unlock-Key-Credentials
#1 was written by Pianoguy on redflagdeals forums.
For anyone using Obihai ATAs, the .pdf guides and Obitalk.com do not configure the Obihai ATAs to conform to the recommended guidelines provided by Freephoneline.
These changes should be made:
1. Under Voice Services-->SP(FPL) Service
X_KeepAliveEnable: enabled
X_KeepAliveExpires: 20 (Obitalk.com uses 15, which differs from what Freephoneline recommends)
X_KeepAliveMsgType: notify
If notify, is not, as a X_KeepAliveMsgType available for you, try what Pianoguy wrote here:
Quote from: Pianoguy
X_KeepAliveEnable: Checked
X_KeepAliveExpires: 20
X_KeepAliveMsgType: custom
X_CustomKeepAliveMsg: mtd=NOTIFY
For OBi20x, with firmware 3.1.0 (Build: 5285), "X_KeepAliveMsgType: notify" is selectable: http://fw.obihai.com/OBi202-3-1-0-5285.fw
Again, refer to Freephoneline's published recommended settings: http://support.freephoneline.ca/hc/en-us/articles/212430746-VoIP-Unlock-Key-Credentials
2. Under Service Providers-->ITSP Profile (FPL)-->RTP
KeepAliveInterval: 20
3. Service Providers-->ITSP Profile (FPL)-->SIP
X_UsePublicAddressInVia: enabled
Freephoneline (FPL) configures its switches oddly. To help avoid one-way audio issues, enable this setting.
4. Service Providers-->ITSP Profile (FPL)-->SIP
Uncheck the box under the Value column to disable X_Use302ToCallForward
Freephoneline requires calls to be bridged if you want your ATA to forward calls.
If X_Use302ToCallForward is enabled, calls that are forwarded by the ATA (as opposed to using Freephoneline’s Follow Me feature) will be dropped to voicemail
5. Voice Services-->SP(FPL) Service
X_UserAgentPort would be better as a random port number between 30000 and 65535. Just pick a port number in that range.
By using a high random port you help to thwart SIP scanners and may also circumvent a faulty SIP ALG feature in your router.
6. Unless I'm not seeing something, the configuration .pdf guides miss
RegisterRetryInterval needing to be 120
That's found under Service Providers-->ITSP Profile (FPL)-->SIP
If you make more than 5 registration attempts within a 5 minute period, you will be temporarily IP banned by the FPL server you were attempting to register with.
7. Service Providers–>ITSP Profile (FPL)–>General
DigitMap: (1xxxxxxxxxx|011XX.S3|[2-9]xxxxxxxxx|*98|911)
Here’s an alternative example:
(1xxxxxxxxxx|011XX.S3|[2-9]xxxxxxxxx|<211:4163974636>|<311:4163922489>|<511:4162354686>|<611:4164772010>|<811:8667970000>|*98|911)
The digitmap is appropriate for Toronto. For 211, 311, 511, 611, 811 you will need to look up the corresponding phone numbers for your area and replace the phone number after the colon.
Mipd is for IP dialing
[^*#]@@. is for sip uri
Neither is needed with Freephoneline. They should be removed.
[6-7]x*xxxxxxxxxxx. in the .pdf guide is complete nonsense and should never be used since it can't logically apply to anything.
XX. is usually not needed (and actually, inadvisable, since it can apply to anything and, due to it being an indefinite variable creates a 10 second timeout in the ATA while it waits for the user to finish entering a phone number).
*98 is for voicemail, which works with Freephoneline, by default.
8. Navigate to Codecs–>Codec Profile (A or whatever the VoIP service you're using is assigned to. You can determine this under Voice services-->SP[freephoneline] Service-->X_CodecProfile)
i. Uncheck the default boxes for Enable (Fax Event) and T38ECM
ii. Check or enable the boxes under the Value column for Enable (Fax Event) and TC8ECM.
T.38 Fax protocol works with an OBi200 or OBi202 and Freephoneline.
9. Star Code Profiles (A & B)
Code28: *99, Blind Transfer, coll($Bxrn)
Blind Transfer can't be *98 with Freephoneline. *98 is meant for Voice mail. Change Blind transfer to *99 or something other than *98. Obitalk's preset configuration doesn't have this set properly for FPL.
10. If FPL is your primary service, then
Under Voice Services-->SP(FPL), the following should be enabled (and they're not in the .pdf guides for some unknown reason) if you want Voicemail notification to work:
MWIEnable
MWIEnable2 (OBi202)
X_VMWIEnable
X_VMWIEnable2 (OBi202)
11. If you were issued a modem/router combo by your ISP that has SIP ALG forced on with no way to disable it (Rogers Hitron router/modem combos, for example), try using voip4.freephoneline.ca:6060 for the ProxyServer, which is intended to help avoid SIP ALG issues.
No one should be port forwarding.
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