Topic: SAIL config on receiving calls from FXO

[Franco]

I installed 02 cards on my system but I only want to use 01 card to
receive calls, the other will be used only to make calls. There's a
regular phone attached to the line-out which will receive the calls.
How can I bypass asterisk on the receiving call?

Thanks,

[SARK devs]

Hi Stuntshell

Not entirely sure I understand this.  However, here's what we know.

Before 2.1.13, you could only force an outbound call onto a particular zap trunk by giving it a carrier select string (in trunks) and dialing the carrier select string to force the call onto that zap line.  Routes ALWAYS used the zap group to dial so you couldn't guarantee which line would be used.  In 2.1.13, in the route drop down, you can choose the group or a specific zap line.  In this way you can force the system to choose a particular zap trunk for a given route if you wish.  This takes care of your outbound requirement to only use one line.

For inbound, if you wish to bypass asterisk/sail, then why bother having a card at all?  Why not just attach the phone to the wall socket?   Or is it that there is only one phone?  If only one phone you can't bypass asterisk unless you use something like an spa3000 instead of a card.

Hope we've understood this.

Kind Regards

Selintra

[Franco]

Hi,
Sorry for the confusion.
Basically we'll have 02 lines, but only use one of them to dial out. When receiving calls we do not want asterisk to answer them.
In a regular asterisk I would modify the extensions to something like this

Code: [Select]

exten => s,1,Hangup

Thanks,

[SARK devs]

Ah,  

OK, just create a custom app (maybe called hangup_inbound) with the code you've included above (exten=>s,1,Hangup) and then point the two analogue trunks at the custom app in the inbound route drop down.  Any inbound calls will then be dropped.

However, this does mean that the call will be physically put on-hook - is that what you want?

Kind Regards

Selintra

[Franco]

Not really  
I wanted it to ignore the call.
The FXO has a line in and a line out for a phone to work in case electricity goes down, so I want this phone to answer all the calls. If I'm using a voip phone I can still use this same line to dial out.

Thanks,

[soprom]

I have to deal with a similar situation but the thread stops where I need extra info!

My zap channel is used for outbound only. How to deactivate any inbound call processing for this particular channel?

[SARK devs]

Here's something you can try...  No guarantees tho'

You'll need the latest sail release (-322) with the "late terminate" code in it and you will need to turn FAX detection OFF for the trunk.

Code up a custom app which simply hangs up and drive that from the trunk.

exten=>s,1,hangup

In theory, this should drop the call without ever terminating it, which would allow an upstream phone on a T-bar, or splitter, to take the call.  

Let me know  how you get on.

Kind Regards

Selintra

[soprom]

Thanks... I'm giving your method a test but do I have to manage the kernel with this new version?

Actually, I need to boot with vmlinuz-2.6.9-34.ELsmp
after a yum update bringing vmlinuz-2.6.9-34.0.2.ELsmp
to my server.

With version -312 this method did not work.
I'll keep you posted!!

Thanks

Sophie

[SARK devs]

do I have to manage the kernel with this new version?

SAIL is noarch.  It is pretty much asterisk release independent (it won't run with very old releases).  It really doesn't care what asterisk you have or how it was created.  You can use our rpms or the ones mentioned above or you can compile asterisk from source, it doesn't matter.  

The kernel problems are to do with the fact that zaptel, the asterisk TDM device driver, installs its modules into the kernel directory tree in lib/modules/kernel-name/extra.  This is fine until yum update installs a new kernel.  New kernels are installed alongside their predecessors into a new directory tree.  Look in /lib/modules and you'll see what I mean. When you fetch up the new kernel it can't find the zaptel modules because they're still in the old directory tree.   There doesn't appear to be any simple answer to this, (short of producing Zaptel rpms for every kernel point release)  at least from a maintainable, regressible code point of view.  We may be able to do something in the SME Server post upgrade templates.

However, it is important to note that this is an Asterisk Zaptel issue.  It isn't a SAIL issue.  Anyone running asterisk under Rhel, fedora or CentOS, in any guise, will be confonted with this issue when there is a kernel change.  Yum further exacerbates the issue because users probably didn't update their kernels as frequently prior to its widespread use.

Kind Regards

Selintra

[soprom]

Many thanks! It works. Calls to the zap channel are not ansered by the pbx. Exactly what was needed.

Now I'm trying to use the zap channel for outbound...

After a reboot changes were removed and the problem reappeared.

[soprom]

What happens if I edit extensions.conf manually to remove these lines:

exten => s,1,GotoIf($["${CHANNEL}" = "Zap/1-1"]?s-514388XXXX,1)

The line reappeared when I manage the PBX.

FAX and ACT are unchecked but I can see in extensions.conf the lines:
FAX=5XXX
exten => s,1,GotoIf($["${CHANNEL}" = "Zap/1-1"]?s-514388XXXX,1)

[SARK devs]

After a reboot changes were removed and the problem reappeared.

Hi

Did you put your changes in a custom-app or did you change extensions.conf directly?

Direct changes to extensions.conf will be overwritten at the next commit, unless you make your changes in a custom app or in extensions.conf headers.

As to the generated lines you can see in extensions....

FAX=5XXX
exten => s,1,GotoIf($["${CHANNEL}" = "Zap/1-1"]?s-514388XXXX,1)

You really shouldn't attempt to remove these lines otherwise the treatment of an incoming zap call will be unpredictable.  

The FAX stmt is irrelevent if FAX is turned off for your line, however it still gets generated because many users have multiple zap lines, some of which may be fax enabled and some of which may not.

Kind Regards

Selintra

[soprom]

I'm still having problem with this outbound-only channel...

This line keeps showing extensions.conf and I can't trace the context it in the conf files (ie: where is s-5141112945 ?):
exten => s,1,GotoIf($["${CHANNEL}" = "Zap/1-1"]?s-5141112945,1)

As I don't want the incoming call to be processed on this channel, I thought I would see something like:
exten => s,1,GotoIf($["${CHANNEL}" = "Zap/1-1"]?DoNotAnswer,1)
as defined with CustomApps  and TrunkLine.
Also, FAX and ACT are un-checked.

When a call is received and answered by a POTS, the pbx grabs the line after about 10 sec and says the extension is invalid. Then the pbx records a few seconds of conversation and send this as a vm to the operator. Then the pbx hangs up and the conversation on the POTS continues as if nothing happened!

Any advise or help is appreciated...