Topic: selintra-sail and fax leads to connection debits?

[tias]

I've tried to set up an Asterix server and let it configure through selintra, but I can't actually think that the system is very user friendly.

First of all I had to manually put in a register => under [general] in sip.conf for get the connection to my SIP provider to work. But now something strange has occured. Even if I don't answer my phone the caller is debited for the connection cost.

My suspicion is that somekind of fax control automatic lift the phone to check if there's a fax machine in the outher side... But in my trunkline there are no activation for any fax.

I would like to set up this so no unanswered phone calls are beeing debited. Any suggestions how to fix this?

[SARK devs]

Hello Tias

I've tried to set up an Asterix server and let it configure through selintra, but I can't actually think that the system is very user friendly

Sorry to hear that - please tell us more

I had to manually put in a register => under [general] in sip.conf

You don't mention who your carrrier is but there is a registration field available to you in the modify trunk panel - here is the relevant page from the /docs site... http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter096

Many carriers demand different registration strings so we allow you the opportunity to add your own string here and further, where we know the format, we supply it in the carrier definition for you.  We're really sorry if you find this unfriendly but in our defence, none of the competitive systems which we benchmark against has any concept of a carrier pre-definition library at all and you end up having to specify everything from scratch.

Even if I don't answer my phone the caller is debited for the connection cost. My suspicion is that somekind of fax control automatic lift the phone to check if there's a fax machine in the outher side...

Yes, for the most part this is true.  SAIL is not designed for domestic use.  It is targetted squarely at small to medium sized businesses.  Businesses ALWAYS want to terminate an incoming call.  Even if no-one is present they still want to run an IVR script, drop to voicemail or relay the call on someplace else.   However, having said that, in an ideal world we should take the call as late as possible anyway, simply as a courtesy to our callers.

Thanks for this input, we'll have a look at the termination issue again.  In the meantime, do let us know of any other issues/thoughts you may have.

Kind Regards

Selintra

[tias]

What I mean when I say that the system isn't rather user friendly I mostly referer to the lack of info in direct connection to page. (Making links och popups with descriptive info would make the thing a whole lot easier). To trace down the info in u'r documentation wasn't the easiest part.

I understand that most company always will have something that answer the phone, but I read that the fax function didn't was involved in SIP but if u ran on pci cards and analog phone. Why are there difference?

Mine provider isn't in u'r list therefore I had to do all the work. But it would be nice if some info would describe the form and give an example for what can be filled there.

I traced the answering thing to the selintra script that's running when an incoming connection on SIP comes. The debit comes when the script starts playing a sound for the dialer, just silence...

Could not this function be disabled?

For what I can count out the dialer is debited for the call from the very beginning, and if someone isn't answering the phone for let's say half a minute, then the dialer will still have to pay for that time. I am aware that there isn't any big costs but it just the courtesy to the caller, not force them to pay for something that they didn't get anything from. And hopely, someone will answer the phone.

[SARK devs]

Hello Tias

Making links och popups with descriptive info would make the thing a whole lot easier). To trace down the info in u'r documentation wasn't the easiest part.

SAIL is written in the same style as the rest of the SME Server Manager interface and none of the other panels give much interactive help either.  That's a poor excuse, but it's the reason why it is the way it is.  One thing we are looking at is context sensitive help which will throw up the requisite page from the manual when you click on the "help" key. However, in the end, SAIL admin is not targetted at end users and it will probably always demand a certain amount of skill/know-how from the administrator - just like all of the other admin panels.

I understand that most company always will have something that answer the phone, but I read that the fax function didn't was involved in SIP but if u ran on pci cards and analog phone. Why are there difference?

FAX doesn't work very well (if at all) over VOIP.  You can get away with it but only using lossless Codecs such as alaw/ulaw which are very bandwidth hungary.  Even then, sometimes it works, sometimes it doesn't.  In any event we don't think its good enough for commercial use so we don't check for FAX tones on SIP lines.

I traced the answering thing to the selintra script that's running when an incoming connection on SIP comes. The debit comes when the script starts playing a sound for the dialer, just silence...

Yes you are quite correct.  The silence is a hangover from 1.0.x of Asterisk.  There was a bug in asterisk which would not play IVR messages properly unless you issued a specific sound command first.  It's probably time to remove it now.   Let me check tomorrow and get back to you.

[tias]

Hi again, is there possible to remove this none silence sound?

Haven't got any response from you,
hope that it will be possible.

It will make the server solution a whole lot better.  

[SARK devs]

Hello Tias,

Hi again, is there possible to remove this none silence sound?

The AGI developer looked at this today and the answer is yes...  and no.  The silence has been moved rather than removed.  Here is what the he has done...

In the event that a call is made which is routed to an extension, the call will not be terminated until the receiver is picked up (which handles your non-pickup/non-termination requirement).   However if the endpoint involves any kind of IVR or messaging (e.g. voicemail) then a one-second silence will be issued to terminate the call before any voice response is attempted.  The reason for this is that if we allow the voice response to terminate the call, the caller will not hear the first second or so of the response.  This seems to do with the time it takes for the call to come up and stabilize.  

This will be issued as part of the 2.1.14 release which will be available from tomorrow evening.

Kind Regards and thanks for your interest.

[tias]

Looking forward to try the changed AGI.

Thanks for the help,
regards,
Tias

[tias]

Sorry, but I cant find the rpm for the new version

where can I find this new version,
best regards,
tias

[SARK devs]

It's not ready yet.    We are just packaging it now.  There will be an announce here when its ready.

Best

Selintra

[tias]

ah, evening and morning is kind of a difference Sat here waiting but maybe soon it will come...

Well gonna be fun to test it anyway.

[tias]

Has there been any delays?
I feel like a little boy  

Wonder if I will get something to play with tomorrow...

---
Tias

[SARK devs]

Hello Tias,

I'm afraid that Santa Clause is a little late.  The 2.1.14 release failed in test this afternoon.  We may have broken FAX detection when removing the silence code.  As soon as we fix it we'll release.  

Courage mon vieux!

Best

Selintra

[SARK devs]

Hello Tias,

I'm afraid that Santa Clause is a little late.  The 2.1.14 release failed in test this afternoon.  We may have broken FAX detection when removing the silence code.  As soon as we fix it we'll release.  

Courage mon vieux!

Best

Selintra

[SARK devs]

Hello Tias,

I'm afraid that Santa Clause is a little late.  The 2.1.14 release failed in test this afternoon.  We may have broken FAX detection when removing the silence code.  As soon as we fix it we'll release.  

Courage mon vieux!

Best

Selintra

[tias]

Buhu, wondering how that will be welcomed by a boy... sorry santa is late

Well, I wait and hope for the best, if it shall not work is there any possibility to release it anyway, for what I understand the fax thing didn't work any longer, and for my use it's no big deal

---
Tias waiting for santa...  

[SARK devs]

Hello Tias

Patience rewarded -

The 2.1.14-301 release is now available to test.  You can find it here

ftp://81.149.154.14/AlphaReleases-unstable/selintra-sail-2.1.14-301.noarch.rpm

Have fun and let us know if you find any errors.

Kind Regards

Selintra

[tias]

Santa has been good

Well, finaly got it up and running but when will something work from start? I've configurated a Vood box to connect to the asterisk. When the box registers against asterisk and asterisk sees the box as online, the phone makes a short ringing. This ringing will then come repeatly with something like 3 minutes between. The phone works in all ways, except that about every third minute it makes a little ring.... (just a short one) The box didn't record any incomming calls if I check in the log.

At my SME I've checked for something in CLI but nothing will come up... Don't know how to solve this problem. Does anyone know how to solve this problem?

thankful for help in this situation.

regards,
Tias

[SARK devs]

Hello Tias

Don't know the vood box but I'm assuming it is an ATA; yes?  Because conventional phones have no message waiting light, most ATA's "chirrup" to signify a message waiting (Sipura were first to do this and most of the others have simply followed suit).  They usually also give a couple of seconds of "stutter" dial tone when you first take the phone off-hook.  Both these things signify message waiting.  Usually you can turn the chirrup off in the ATA's browser settings.  

Kind Regards

Selintra

[tias]

Seems like u have right. Yes it's an ata... Tried to make the mailbox inavailible by erasing the text from the extension where my ata is connected at. And the ghost signal is no more. Activated it again and there it is again. Tried to listen to my mailbox but no new messages. So my ata-box signals also when no message has arrived. (it's configure as an ordinary sip) But I also got one problem. I can dial from my ata to another ext (where's my computer with x-lite is connected) but not the oposited. If I make a call from the computer to the ata, then the extension is not availible. Both extensions can make outline calls and both extensions will ring on incomming phone calls. But can't call from computer to ata.

So my questions are, is there the ata-box wrong that it make noises even if no messages is in my mailbox?

and what are the reason I can't dial my ata box from my computer.

I'm starting to understand how Selintra system is build and where to make the changes. But it would be nice if the fields get some info i direct connection.

Thankful for u'r help.
regards, Tias

[tias]

Why I couldn't make a call to the ata is solved. Mine own misstake

But IRW isn't solved Any suggestions? Can I disable the function and still have voicemail?

---
Tias

[SARK devs]

HI

But IRW isn't solved Any suggestions? Can I disable the function and still have voicemail?

You've lost me - IRW? I'm guessing you mean the chirruping sound from the ATA.  I don't know why your ATA is chirruping if you don't have mail.  Unfortunately, I don't know anything about the ATA you have, I've simply never heard of a Vood ATA,  so you might want to contact the manufacturer to see if they can help with its config.

Kind Regards

Selintra

[tias]

MWI did I mean... IRW don't know where that term came from. Ok, have to dive into the asterisk and read to see how to solve it. Took contact with my provider and so far nothing has came up

But thanx for all u'r help. I'm starting to like Selintra