eir F2000 and VOIP

Johnboy1951

I have recently had FTTH installed at home.
I have been using VOIP over the ADSL broadband for many years.

My POTS landline is still enabled and working (have not checked the broadband side of it as I moved directly to FTTH broadband).

So the VOIP set up in the F2000 is not activated for eir VOBB I guess?
I am unsure what the situation is there.

While I have no objection presently with having my landline remain on the copper line, I would like to transfer my working VOIP from my old Draytek Vigor to the F2000.

I suspect I cannot, but would like to hear from someone who has better info than my guesses.

At present I have the Draytek WAN side getting a LAN IP from the F2000 and the VOIP phones are working in that set up, so I am not at a loss of functionality presently.

I am concerned that if eir changes my landline to VOIP (I think this is their intention) I might lose that functionality, and still be unable to set up my existing VOIP accounts in the F2000.

Can anyone enlighten me?

What are the limitations of the eir F2000 in this regard?

Thanks.

Johnboy1951

Got registered with a free Linphone account!

The usual thing .... ask for help and soon after have success!

So I am hopeful that I can do all I require using this F2000, as it allows multiple VOIP accounts and providers.

I will do some tests next weekend hopefully.

tsue921i8wljb3

Johnboy1951 wrote: »

Got registered with a free Linphone account!

The usual thing .... ask for help and soon after have success!

So I am hopeful that I can do all I require using this F2000, as it allows multiple VOIP accounts and providers.

I will do some tests next weekend hopefully.

I have the eir VoIP service as part of my bundle and I could not get line 2 on the F2000 to register with my Blueface account. I suspect if eir move you to VoIP you'll have this limitation too in that the F2000 can only be used for eir VoIP.

Johnboy1951

Duke Tall Cow wrote: »

I have the eir VoIP service as part of my bundle and I could not get line 2 on the F2000 to register with my Blueface account. I suspect if eir move you to VoIP you'll have this limitation too in that the F2000 can only be used for eir VoIP.

That is what concerns me.
I did have some trouble getting the Linphone account to register but got there in the end with some helpful indications from this site
https://www.geekzone.co.nz/forums.asp?forumid=43&topicid=156054
Maybe you could check the entries for your Blueface account against the above and let me know if you have any success?
I have all accounts linked to both lines.

Multiple accounts are possible for now, but I don't know what the limit is. I entered 6 and stopped.

I failed to get outgoing calls with my choice of account when I have multiple accounts registering.
It appears to always use the same one, regardless of the setting for the number dialled and its associated account.
It does use the Speed Dial correctly, just not the correct account - yet!

I have kept the landline separated from the F2000, although I think it can be plugged into it (not sure).

tsue921i8wljb3

Johnboy1951 wrote: »

That is what concerns me.
I did have some trouble getting the Linphone account to register but got there in the end with some helpful indications from this site
https://www.geekzone.co.nz/forums.asp?forumid=43&topicid=156054
Maybe you could check the entries for your Blueface account against the above and let me know if you have any success?
I have all accounts linked to both lines.

Multiple accounts are possible for now, but I don't know what the limit is. I entered 6 and stopped.

I failed to get outgoing calls with my choice of account when I have multiple accounts registering.
It appears to always use the same one, regardless of the setting for the number dialled and its associated account.
It does use the Speed Dial correctly, just not the correct account - yet!

I have kept the landline separated from the F2000, although I think it can be plugged into it (not sure).

Yeah I'll give it a read and try it tomorrow.

tsue921i8wljb3

No go I'm afraid. I can't get it to connect to Blueface. I tried all manner of combinations of settings but it remained offline. There are no messages in the logs relating to the connection attempts.

Johnboy1951

Duke Tall Cow wrote: »

No go I'm afraid. I can't get it to connect to Blueface. I tried all manner of combinations of settings but it remained offline. There are no messages in the logs relating to the connection attempts.

Thanks Navi.

That really is a horrible thing to do to the router!

Supply a good router ... but cripple it! :mad:


I wonder if there is some setting being missed that would allow more than one account to be used.

I expect there must be because some users/households would have more than one phone number.

/

Johnboy1951

Navi ..... will Blueface allow you to set different ports for VOIP?
If so it might be possible to get it working?

tsue921i8wljb3

Johnboy1951 wrote: »

Navi ..... will Blueface allow you to set different ports for VOIP?
If so it might be possible to get it working?

The eir service uses port 5080 by default. I tried Blueface on 5060 and I think 5061. I don't know if Blueface support any other ports for SIP. Their site used to be a treasure trove of technical information but that all changed several years ago. I'm surprised that they have not gotten rid of all residential customers such as myself.

I suspect eir have the firmware locked down. When you add a new VoIP provider the primary proxy is eircom and the SIP domain is sip.eircom.net. These have to be deleted to enter the correct details.

This is all moot for me anyway as I'll be cancelling the eir VoIP in November when they plan to jack up my bill to €93 per month.

Johnboy1951

The eir service uses port 5080 by default.

As you say when configuring a new provider the eir details are filled in, and it is Local 6050; registrar & proxy 5060 that are filled in here, not 5080.

Maybe flashing the router would get things better.
I might look into that.

tsue921i8wljb3

Johnboy1951 wrote: »

As you say when configuring a new provider the eir details are filled in, and it is Local 6050; registrar & proxy 5060 that are filled in here, not 5080.

Maybe flashing the router would get things better.
I might look into that.

5080 for the main eir service. When I go to add the second provider I get the same 6050 & 5060 defaults as you.

Johnboy1951

Although I can enter a SIP account in the Speed Dial list, it seems I am unable to call out to that.
I am able to call out to phone numbers.

This does not seem to be a limitation as the SIP account is accepted as a valid 'number', but I have not figured out what I might be doing wrong.

tsue921i8wljb3

Johnboy1951 wrote: »

Although I can enter a SIP account in the Speed Dial list, it seems I am unable to call out to that.
I am able to call out to phone numbers.

This does not seem to be a limitation as the SIP account is accepted as a valid 'number', but I have not figured out what I might be doing wrong.

IP dialling? Either the dial plan or the device doesn't support it I'd guess. On my Linksys ATA you have to explicitly enable IP dialling although I don't use it.

Johnboy1951

Duke Tall Cow wrote: »

IP dialling? Either the dial plan or the device doesn't support it I'd guess. On my Linksys ATA you have to explicitly enable IP dialling although I don't use it.

I mean calling a SIP account by 'dialling' the account in the form of

someone@sip.server.com

This format is accepted as a valid 'number' in the Dial Plan but is not making any connection when 'dialled'.

I can receive calls into the F2000 from that account and it shows the caller ID correctly.
Just have not figured out how to call out using it.

Overall I am rather disappointed with the telephony functions in the F2000. For a much more modern device than my old Draytek Vigor (more than 10 years in use now) it is sadly lacking in telephony options.

tsue921i8wljb3

Johnboy1951 wrote: »

I mean calling a SIP account by 'dialling' the account in the form of

someone@sip.server.com

This format is accepted as a valid 'number' in the Dial Plan but is not making any connection when 'dialled'.

I can receive calls into the F2000 from that account and it shows the caller ID correctly.
Just have not figured out how to call out using it.

Overall I am rather disappointed with the telephony functions in the F2000. For a much more modern device than my old Draytek Vigor (more than 10 years in use now) it is sadly lacking in telephony options.

I would say the device is not capable of it. I imagine the telephony part of the device was designed with ISPs transitioning to VoIP in mind more than a fully functional ATA.

Johnboy1951

Duke Tall Cow wrote: »

I would say the device is not capable of it. I imagine the telephony part of the device was designed with ISPs transitioning to VoIP in mind more than a fully functional ATA.

That might be so ..... but the fact that such a 'number' is acceptable made me believe otherwise.

championc

Johnboy1951 wrote: »

Got registered with a free Linphone account!

The usual thing .... ask for help and soon after have success!

So I am hopeful that I can do all I require using this F2000, as it allows multiple VOIP accounts and providers.

I will do some tests next weekend hopefully.

Hi there,

I have a separate SIP account and was trying to set it up on my F2000 last night. The router has my landline configured as a VOBB. I added a new SIP account, but where do you enter your username and password for this secondaru SIP provider ?

Johnboy1951

championc wrote: »

Hi there,

I have a separate SIP account and was trying to set it up on my F2000 last night. The router has my landline configured as a VOBB. I added a new SIP account, but where do you enter your username and password for this secondaru SIP provider ?

On the Telephony - VOIP page you first set up the provider .... example Linphone. Blueface etc.

Then add your telephone number (voip account) with that provider. The link to add your 'number' is on the same page.

So, in essence you enter the VOIP provider once and can add multiple accounts (numbers) separately, linked to that provider.


I have failed to make good use of the VOIP functions on this device - due, I believe, to eir having them locked down for their own VOBB service.

I do have one VOIP account set up for dialling out, and have the most frequent calls in my speed dial list.

I did not find an easy to use digitmap so I only use this one as a supplementary to my old Draytek which continues to function.

championc

Johnboy1951 wrote: »

I do have one VOIP account set up for dialling out, and have the most frequent calls in my speed dial list.

I got my secondary account setup, and it's showing Online and Idle. However, I cannot for the life of me get any calls to use this secondary account. I don't mind even using the secondary one permanently for all outbound dialing.

So while I can setup Speed Dials, I cannot see how to route the calls out over a specific account - how did you manage this please ?

Johnboy1951

championc wrote: »

I got my secondary account setup, and it's showing Online and Idle. However, I cannot for the life of me get any calls to use this secondary account. I don't mind even using the secondary one permanently for all outbound dialing.

So while I can setup Speed Dials, I cannot see how to route the calls out over a specific account - how did you manage this please ?

Under Telephone ...... select the account to be associate with each phone, do a test call and view log to see which account it used.
That should be the easiest method of confirming things.

From there on I am unsure because I was unable to get connections to VOIP accounts in the form of
sip: user@sip.provider.org

championc

Johnboy1951 wrote: »

Under Telephone ...... select the account to be associate with each phone, do a test call and view log to see which account it used.
That should be the easiest method of confirming things.

From there on I am unsure because I was unable to get connections to VOIP accounts in the form of
sip: user@sip.provider.org

Thanks,

I'm only interested in using a physical phone for calls. It's connected to the "Phone 1" socket. No matter what, the Telephony > Call Logs shows all calls going out over Eir VOBB. So I cannot find any way for steering a call out over the secondary account.

In fact, within the VoIP Providers, I have the option for allowing Outbound Calls ticked on my secondary connection while it's not even ticked on the VoBB one.

I'm obviously missing something simple.

What does the ABCD in the Dial Plan > Digitmap do ? It's set to [X*#ABCD].T by Eir. Are there parameters which could be added in here to direct a call at Account 1 or Account 2 ?

Johnboy1951

championc wrote: »

Thanks,

I'm only interested in using a physical phone for calls. It's connected to the "Phone 1" socket. No matter what, the Telephony > Call Logs shows all calls going out over Eir VOBB. So I cannot find any way for steering a call out over the secondary account.

In fact, within the VoIP Providers, I have the option for allowing Outbound Calls ticked on my secondary connection while it's not even ticked on the VoBB one.

I'm obviously missing something simple.

I suspect you have hit a limitation applied by Eir to the device, which does not allow any SIP account, other than their own, to function, once their own VOBB is set up.

Only Eir can confirm if that is the case or not. You will need to contact them either via their 'help' line or maybe through their 'Talk to Eir' boards section.

What does the ABCD in the Dial Plan > Digitmap do ? It's set to [X*#ABCD].T by Eir. Are there parameters which could be added in here to direct a call at Account 1 or Account 2 ?

The Digimap determines how the device interprets the numbers dialled.
I have never used a Digimap similar to this so cannot provide more info.

.

championc

I have plugged a physical phone into Port 2 on the router and the call has routed out over my secondary provider. Therefore, I assume that I just need something in the Digitmap to tell the router to use the secondary provider rather than the primary one when maybe a particular series of digits is matched.

tsue921i8wljb3

championc wrote: »

I have plugged a physical phone into Port 2 on the router and the call has routed out over my secondary provider. Therefore, I assume that I just need something in the Digitmap to tell the router to use the secondary provider rather than the primary one when maybe a particular series of digits is matched.

Under Telephony > Telephone there should be Telephone 1 and 2 sections.

Have you tried setting Telephone 2 to the secondary provider and 1 to eir? I don't think the digitmap has anything to do with it.

championc

While I will have to test incoming calls, that would work alright for outgoing.

I'm beginning to think that maybe the ABCD is maybe telling a call that it can use Channels A, B, C and D - whatever they might be.

I tried "x.B" but the primary route was still used. I then checked the entire alphabet and the only letters which can be used within the digitmap are A to D, X and T. X and T are standard allowed characters but A to D certainly are non-standard.

tsue921i8wljb3

championc wrote: »

While I will have to test incoming calls, that would work alright for outgoing.

I'm beginning to think that maybe the ABCD is maybe telling a call that it can use Channels A, B, C and D - whatever they might be.

I tried "x.B" but the primary route was still used. I then checked the entire alphabet and the only letters which can be used within the digitmap are A to D, X and T. X and T are standard allowed characters but A to D certainly are non-standard.

X is any number from 0 to 9.
T is the delay which I guess is the inter-digit timeout under Advanced (3 seconds)
The dot I believe means repetition so the dialplan is looking for more than one of X, # (used for routing prefix), * (used for speed dial)

I don't know what A B C or D refer to but I don't think it is anything to do with selecting lines.

tsue921i8wljb3

I think I've found out what A B C D refer to. It is to do with DTMF tones.

https://en.wikipedia.org/wiki/Dual-tone_multi-frequency_signaling##,_*,_A,_B,_C,_and_D

ED E

The digitmap is basically "whats a valid phone number" so when you type:

016011234 <- At the 4 it says oh, its a number, go ahead and start dialing. Saves you manually sending it.

The Call Agent can ask the gateway to collect digits dialed by the
user.  This facility is intended to be used with residential gateways
to collect the numbers that a user dials; it can also be used with
trunking gateways and access gateways alike, to collect access codes,
credit card numbers and other numbers requested by call control
services.
 
One procedure is for the gateway to notify the Call Agent of each
individual dialed digit, as soon as they are dialed.  However, such a
procedure generates a large number of interactions.  It is preferable
to accumulate the dialed numbers in a buffer, and to transmit them in
a single message.
 
The problem with this accumulation approach, however, is that it is
hard for the gateway to predict how many numbers it needs to
accumulate before transmission.  For example, using the phone on our
desk, we can dial the following numbers:
 
     ------------------------------------------------------
    |  0                     |  Local operator             |
    |  00                    |  Long distance operator     |
    |  xxxx                  |  Local extension number     |
    |  8xxxxxxx              |  Local number               |
    |  #xxxxxxx              |  Shortcut to local number at|
    |                        |  other corporate sites      |
    |  *xx                   |  Star services              |
    |  91xxxxxxxxxx          |  Long distance number       |
    |  9011 + up to 15 digits|  International number       |
     ------------------------------------------------------
 
The solution to this problem is to have the Call Agent load the
gateway with a digit map that may correspond to the dial plan.  This
digit map is expressed using a syntax derived from the Unix system
command, egrep.  For example, the dial plan described above results
in the following digit map:
 
   (0T|00T|[1-7]xxx|8xxxxxxx|#xxxxxxx|*xx|91xxxxxxxxxx|9011x.T)
 
The formal syntax of the digit map is described by the DigitMap rule
in the formal syntax description of the protocol (see Appendix A) -
support for basic digit map letters is REQUIRED while support for
extension digit map letters is OPTIONAL.  A gateway receiving a digit
map with an extension digit map letter not supported SHOULD return
error code 537 (unknown digit map extension).
 
A digit map, according to this syntax, is defined either by a (case
insensitive) "string" or by a list of strings.  Each string in the
list is an alternative numbering scheme, specified either as a set of
digits or timers, or as an expression over which the gateway will
attempt to find a shortest possible match.  The following constructs
can be used in each numbering scheme:
 
* Digit:    A digit from "0" to "9".
* Timer:    The symbol "T" matching a timer expiry.
* DTMF:     A digit, a timer, or one of the symbols "A", "B", "C",
            "D", "#", or "*".  Extensions may be defined.
* Wildcard: The symbol "x" which matches any digit ("0" to "9").
* Range:    One or more DTMF symbols enclosed between square brackets
            ("\[" and "]").
* Subrange: Two digits separated by hyphen ("-") which matches any
            digit between and including the two.  The subrange
            construct can only be used inside a range construct,
            i.e., between "\[" and "]".
* Position: A period (".") which matches an arbitrary number,
            including zero, of occurrences of the preceding
            construct.
 
A gateway that detects events to be matched against a digit map MUST
do the following:
 
1) Add the event code as a token to the end of an internal state
   variable for the endpoint called the "current dial string".
 
2) Apply the current dial string to the digit map table, attempting a
   match to each expression in the digit map.
 
3) If the result is under-qualified (partially matches at least one
   entry in the digit map and doesn't completely match another
   entry), do nothing further.
 
If the result matches an entry, or is over-qualified (i.e., no
further digits could possibly produce a match), send the list of
accumulated events to the Call Agent.  A match, in this
specification, can be either a "perfect match," exactly matching one
of the specified alternatives, or an impossible match, which occurs
when the dial string does not match any of the alternatives.
Unexpected timers, for example, can cause "impossible matches".  Both
perfect matches and impossible matches trigger notification of the
accumulated digits (which may include other events - see Section
2.3.3).
 
The following example illustrates the above.  Assume we have the
digit map:
 
   (xxxxxxx|x11)
 
and a current dial string of "41".  Given the input "1" the current
dial string becomes "411".  We have a partial match with "xxxxxxx",
but a complete match with "x11", and hence we send "411" to the Call
Agent.
 
The following digit map example is more subtle:
 
  (0[12].|00|1[12].1|2x.#)
 
Given the input "0", a match will occur immediately since position
(".") allows for zero occurrences of the preceding construct.  The
input "00" can thus never be produced in this digit map.
 
Given the input "1", only a partial match exists.  The input "12" is
also only a partial match, however both "11" and "121" are a match.
 
Given the input "2", a partial match exists.  A partial match also
exists for the input "23", "234", "2345", etc.  A full match does not
occur here until a "#" is generated, e.g., "2345#".  The input "2#"
would also have been a match.
 
Note that digit maps simply define a way of matching sequences of
event codes against a grammar.  Although digit maps as defined here
are for DTMF input, extension packages can also be defined so that
digit maps can be used for other types of input represented by event
codes that adhere to the digit map syntax already defined for these
event codes (e.g., "1" or "T").  Where such usage is envisioned, the
definition of the particular event(s) SHOULD explicitly state that in
the package definition.
 
Since digit maps are not bounded in size, it is RECOMMENDED that
gateways support digit maps up to at least 2048 bytes per endpoint.

ED E

AFAIK your routing prefixes are the only bit you should need to fiddle with.

Nice for reference.
https://helpdesk.voyager.co.nz/index.php?/Knowledgebase/Article/View/221/0/setting-up-voice-voip-on-your-voyager-huawei-router

hawthorne

https://www.boards.ie/discussion/comment/108919301#Comment_108919301

Just to revive this thread:

I have a F2000 from Eir. It was working ok when it comes to VOIP. I use "https://www.freevoipdeal.com/". No problems until last Friday. I had contacted Eir about a speed problem with our internet connection. Your man fiddled around a bit on his side- but achieved nothing. An engineer will be sent out in due course to check the line for a fault. But since then our VOIP service is gone. I did a factory reset and reprogrammed the device as shown in the above link. Unfortunately it did not help. The VOIP service is shown as being "offline" and the phone symbol on the screen of the router keeps blinking. No calls are possible. Internet works- but is slow.

Any ideas?