[asterisk-users] catch-all extension in context

[Vieri]

Hi,

Suppose I have the following in my AEL dialplan:

context incoming-1 {
  _. => {
Set(GROUP()=1);
goto incoming|${EXTEN}|1;
}
};

context incoming-2 {
  _. => {
Set(GROUP()=2);
goto incoming|${EXTEN}|1;
}
};

context incoming {
  fax => {
Do stuff for incoming fax...
}
  _. => {
Do stuff for incoming voice call...
}
};

faxdetection is activated.

I'm expecting 'incoming-1' and 'incoming-2' to goto 'incoming' EVEN if Asterisk 
detects the call as being a fax BEFORE going to 'incoming'.
Is that correct? (ie. _. also matches 'fax')

Thanks,

Vieri


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Re: [asterisk-users] Connecting Skype to Asterisk

[asterisk asterisk]

The only inexpensive way is to get siptosis but the developer has stopped
the support and upgrade unfortunately. I have been using it for two years
or more.

Excellent quality and works very well

On Sat, Oct 13, 2012 at 5:17 AM, Philip Bennefall wrote:

> From what I gather, it costs extra for each channel even for direct Skype
> to Asterisk calls. Since my plan was to use this for business purposes, I'd
> need at least something like 30 channels which would be way out of my
> monthly budget unfortunately.
>
> Kind regards,
>
> Philip Bennefall
> - Original Message - From: "Duncan Turnbull" <
> dun...@e-simple.co.nz>
> To: ; "Asterisk Users Mailing List - Non-Commercial
> Discussion" 
> 
> >
> Cc: 
> Sent: Friday, October 12, 2012 11:08 PM
> Subject: Re: [asterisk-users] Connecting Skype to Asterisk
>
>
>
>
> On 13/10/2012, at 7:54 AM, Christopher Harrington  wrote:
>
>  On Fri, Oct 12, 2012 at 1:44 PM, Philip Bennefall 
>> wrote:
>>
>>> Hi all,
>>>
>>> I have an Asterisk PBX under development, that I would like to link to a
>>> Skype account if possible. The idea is that people would call a
>>> particular
>>> Skype username, and be redirected to my SIP and through that to
>>> Asterisk. Is
>>> this doable? I have looked around and saw the Skype for Asterisk driver,
>>> but
>>> of course that has been discontinued. Are there any other options? I
>>> would
>>> prefer not to have to go through the regular PSTN telephone network but
>>> directly from Skype to Asterisk via SIP. If you have any tips on how to
>>> configure my sip.conf to get this working, this would also be highly
>>> appreciated.
>>>
>>>
>> It looks like this is what you want:
>> http://www.skype.com/intl/en/**business/skype-connect/
>>
>>  This is pretty straight forward to use for inbound skype business user
> names and outbound either to pstn, skype numbers are a little more to setup
>
> There is a monthly cost but its not much and if you have skype users out
> there its a good way for them to connect in
>
>
>> --
>> -Chris Harrington
>> ACSDi Office: 763.559.5800
>> Mobile Phone: 612.326.4248
>>
>> --
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Re: [asterisk-users] Connecting Skype to Asterisk

[Philip Bennefall]

From what I gather, it costs extra for each channel even for direct Skype to 
Asterisk calls. Since my plan was to use this for business purposes, I'd 
need at least something like 30 channels which would be way out of my 
monthly budget unfortunately.


Kind regards,

Philip Bennefall
- Original Message - 
From: "Duncan Turnbull" 
To: ; "Asterisk Users Mailing List - Non-Commercial 
Discussion" 

Cc: 
Sent: Friday, October 12, 2012 11:08 PM
Subject: Re: [asterisk-users] Connecting Skype to Asterisk



On 13/10/2012, at 7:54 AM, Christopher Harrington  wrote:

On Fri, Oct 12, 2012 at 1:44 PM, Philip Bennefall  
wrote:

Hi all,

I have an Asterisk PBX under development, that I would like to link to a
Skype account if possible. The idea is that people would call a 
particular
Skype username, and be redirected to my SIP and through that to Asterisk. 
Is
this doable? I have looked around and saw the Skype for Asterisk driver, 
but
of course that has been discontinued. Are there any other options? I 
would

prefer not to have to go through the regular PSTN telephone network but
directly from Skype to Asterisk via SIP. If you have any tips on how to
configure my sip.conf to get this working, this would also be highly
appreciated.



It looks like this is what you want:
http://www.skype.com/intl/en/business/skype-connect/

This is pretty straight forward to use for inbound skype business user names 
and outbound either to pstn, skype numbers are a little more to setup


There is a monthly cost but its not much and if you have skype users out 
there its a good way for them to connect in




--
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248

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Re: [asterisk-users] Connecting Skype to Asterisk

[Duncan Turnbull]

On 13/10/2012, at 7:54 AM, Christopher Harrington  wrote:

> On Fri, Oct 12, 2012 at 1:44 PM, Philip Bennefall  wrote:
>> Hi all,
>> 
>> I have an Asterisk PBX under development, that I would like to link to a
>> Skype account if possible. The idea is that people would call a particular
>> Skype username, and be redirected to my SIP and through that to Asterisk. Is
>> this doable? I have looked around and saw the Skype for Asterisk driver, but
>> of course that has been discontinued. Are there any other options? I would
>> prefer not to have to go through the regular PSTN telephone network but
>> directly from Skype to Asterisk via SIP. If you have any tips on how to
>> configure my sip.conf to get this working, this would also be highly
>> appreciated.
>> 
> 
> It looks like this is what you want:
> http://www.skype.com/intl/en/business/skype-connect/
> 
This is pretty straight forward to use for inbound skype business user names 
and outbound either to pstn, skype numbers are a little more to setup 

There is a monthly cost but its not much and if you have skype users out there 
its a good way for them to connect in

> 
> -- 
> -Chris Harrington
> ACSDi Office: 763.559.5800
> Mobile Phone: 612.326.4248
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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> 
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users


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[asterisk-users] Storing Custom greeting VM in DB

[Ahmed Munir]

Hi all,

I configured the voicemail using realtime and for record voice messages,
I'm storing it in to MySQL DB as this setup works perfectly without any
issues.  Later I tried to insert the custom greeting (busy) for VM in DB
for particular extension, it was unable to play the custom greeting but
play the default prompt.

Even though, I created the folders (busy and unavail) in the
/var/spool/asterisk/voicemail/default/'1234567 directory, converted the
.wav file to 8KHz 16 bit mono,converted to .gsm format and using default
context for voicemail.  Listing down the data and query;

+--+-+-++--+--+---+---++++-+--++--+--+-+--+---++--+---++-+-+---+---++
| uniqueid | customer_id | context | mailbox| password | fullname |
email | pager | tz | attach | saycid | dialout | callback | review |
operator | envelope | sayduration | saydurationm | sendvoicemail | delete |
nextaftercmd | forcename | forcegreetings | hidefromdir |
stamp   | profile   | forwardno | queue_extn |
+--+-+-++--+--+---+---++++-+--++--+--+-+--+---++--+---++-+-+---+---++
| 8475 | 0   | default | 1234567 |  |  |
|   | en | yes| yes| |  | no | no   |
no   | no  |1 | no| no |
yes  | no| no | yes | 2012-10-12
15:42:40 | voicemail | NULL  | NULL   |
+--+-+-++--+--+---+---++++-+--++--+--+-+--+---++--+---++-+-+---+---+--

INSERT INTO voicemessages
(msgnum,dir,mailboxuser,mailboxcontext,recording)
VALUES
(-1,'/var/spool/asterisk/voicemail/default/'1234567/busy','1234567','default',LOAD_FILE('/var/spool/asterisk/voicemail/default/'1234567/busy.wav')),
(-1,'/var/spool/asterisk/voicemail/default/'1234567/unavail',''1234567','default',LOAD_FILE('/var/spool/asterisk/voicemail/default/'1234567/unavail.wav'));


Do I need to modify any other configuration? Please advise to resolve this
issue.

-- 
Regards,

Ahmed Munir Chohan
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Re: [asterisk-users] Connecting Skype to Asterisk

[Christopher Harrington]

On Fri, Oct 12, 2012 at 1:44 PM, Philip Bennefall  wrote:
> Hi all,
>
> I have an Asterisk PBX under development, that I would like to link to a
> Skype account if possible. The idea is that people would call a particular
> Skype username, and be redirected to my SIP and through that to Asterisk. Is
> this doable? I have looked around and saw the Skype for Asterisk driver, but
> of course that has been discontinued. Are there any other options? I would
> prefer not to have to go through the regular PSTN telephone network but
> directly from Skype to Asterisk via SIP. If you have any tips on how to
> configure my sip.conf to get this working, this would also be highly
> appreciated.
>

It looks like this is what you want:
http://www.skype.com/intl/en/business/skype-connect/


-- 
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248

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[asterisk-users] Connecting Skype to Asterisk

[Philip Bennefall]

Hi all,

I have an Asterisk PBX under development, that I would like to link to a 
Skype account if possible. The idea is that people would call a particular 
Skype username, and be redirected to my SIP and through that to Asterisk. Is 
this doable? I have looked around and saw the Skype for Asterisk driver, but 
of course that has been discontinued. Are there any other options? I would 
prefer not to have to go through the regular PSTN telephone network but 
directly from Skype to Asterisk via SIP. If you have any tips on how to 
configure my sip.conf to get this working, this would also be highly 
appreciated.




Thanks in advance for your help!



Kind regards,



Philip Bennefall


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Re: [asterisk-users] Digium D40 phones and Caller ID

[Christopher Harrington]

On Fri, Oct 12, 2012 at 11:47 AM, Matthew Jordan  wrote:
> After loading, [peer01] shows up as a known SIP endpoint.  Calling
> peer01 displays the same caller ID as before, i.e., either 101/D40 01 or
> 101/foo.  Note that none of this using the DPMA either.
Here's something important that I think I was missing. The CallerID
construct seems to be working, but only after the call is answered.
(In my testing, I didn't think to pick up the line, just ending the
call from the source if I didn't see the CID.)

Turning on SIP debugging, I can see the PAI header being sent in the
invite packet being sent to the D40 upon initiating (prior to
answering). Does this shed any light on the issue?

>> I don't know if that is necessarily true; the phones were new in box
>> but were not purchased from Digium or an authorized reseller.
>>
>
> Did they fall off the back of a truck? :-)
>
> Call anyway.  If you purchased said phones in a legitimate manner, they
> should be able to help you.
They were acquired from a company that unexpectedly imploded,
actually. I'm trying to avoid calling as that kills my ability to
multitask on my other projects; naturally I'm the "do everything" guy
at my company.

> (And, as previously suggested, you may want to make sure you've upgraded
> to the latest firmware, just to rule out any solved problems).
They are absolutely all on 1.1.0.

-- 
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ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248

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Re: [asterisk-users] Recommendation for extension mapping on inbound T1 line

[Mitch Claborn]

The "s" extension did not catch the incoming call.  It was only when I 
added a specific 366 or the _. wildcard that I was able to capture the 
incoming call.



Mitch

On 10/12/2012 10:18 AM, A J Stiles wrote:

If  (and only if)  all the extensions you are using in all your contexts are
numeric, then "_." is fine.  (But you don't really need it anyway in your
example, since the "s" extension in your from-pstn context will already catch
the incoming call.)


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Re: [asterisk-users] SoftHangup for emergency calls

[Dave Platt]

> Setting up a group of analog lines to use for outbound emergency calls 
> (911).  My current dial plan and debug output shown below.  It appears 
> that when the SoftHangup() is executed that the line does not really 
> hang up.  In the case shown, I had reduced the group to a single DAHDI 
> (analog) channel and dialed in to that number from the outside. You can 
> see in the output that the SoftHangup() was executed, but the call was 
> not terminated - the outside caller stayed connected to something.  
> Caller no longer heard the sounds from the menu he was in, but the call 
> itself seemed to stay connected.

That may be due to a common characteristic of PSTN lines (at least,
it's common here in the U.S.)

By design, most U.S. PSTN lines have a very asymmetrical response
to a physical hangup:

-  If the calling party hangs up, the call is terminated
   immediately.

-  If the called party hangs up, and the calling party does not,
   the line remains "live" for some time (typically around 30
   seconds, I believe).  If the called party goes off-hook again
   during this period, they can resume the call.

If I recall correctly, things were designed this way so that
the called party could say "Oh, hang on, I answered this call
in the bedroom and the stuff I need is in the living room",
hang up the extension phone, go to another room, pick up the
other phone and carry on with the call.

If that's what you're running into here - if the line you
are trying to SoftHangup() was handing an inbound call - then
there may be no good solution.  As far as I know, there is no
way to force an incoming PSTN call to release the line, other
than "go on-hook, and wait for 30 seconds to pass".

Several possible workarounds, roughly in order of increasing
complexity and decreasing reliability:

(1) Keep one of your PSTN lines reserved for emergency calls
only;  remove it from your inbound hunt group and place
it in a Dahdi line group of its own (or don't group it at
all).

(2) Keep one of your PSTN lines reserved for *outbound* calls
only;  you should be able to SoftHangup() an outbound call
within a second or two.

(3) Figure out a way to check the PSTN lines that are in use
at the time of an emergency - if they're all in use,
somehow find one which was in use for an outbound call,
and select it as the one to SoftHangup() and dial upon.

(4) If you must keep all of your PSTN lines in bidirectional
use, you may have to *tell* the parties that the line is
needed for an emergency call, and ask them to release the
line.  Do a barge-in on the channel, play an alert sound,
play a message saying "Emergency call in progress, please hang
up this line immediately, play the alert sound again for
a few seconds, SoftHangup(), Wait(2), and then try dialing.



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Re: [asterisk-users] SoftHangup for emergency calls

[Richard Mudgett]

> Setting up a group of analog lines to use for outbound emergency
> calls
> (911).  My current dial plan and debug output shown below.  It
> appears
> that when the SoftHangup() is executed that the line does not really
> hang up.  In the case shown, I had reduced the group to a single
> DAHDI
> (analog) channel and dialed in to that number from the outside. You
> can
> see in the output that the SoftHangup() was executed, but the call
> was
> not terminated - the outside caller stayed connected to something.
> Caller no longer heard the sounds from the menu he was in, but the
> call
> itself seemed to stay connected.
> 
> Asterisk 1.8 on Ubuntu
> 
> Any ideas?

I think this behavior is country specific.  I know in the UK, the
caller controls the analog line.  If the called party hangs up, the
caller can stay online and keep the connection.  You may need to limit
these analog lines to outgoing only or reserve the emergency priority
line for outgoing only.

Richard

> [emergency-services]
> exten =>911,1,Goto(dialpsap,1)
> exten =>9911,1,Goto(dialpsap,1)
> exten =>999,1,Goto(dialpsap,1)
> exten =>112,1,Goto(dialpsap,1)
> 
> exten =>dialpsap,1,Verbose(1,Call initiated to PSAP!)
>same =>n(dialit),Dial(${LOCAL}/${EMERGENCY},30)
>same =>n,Verbose(2,DIALSTATUS=${DIALSTATUS})
>same =>n,GotoIf($["${DIALSTATUS}" = "ANSWER"]?good)
>same =>n(hu),SoftHangup(${EMERGENCY_CHANNEL},a)
>same =>n,Wait(5)
>same =>n,Goto(dialit)
>same =>n(good),NoOp(call good)
>same =>n,Hangup()
> 
> 
>== Using SIP RTP CoS mark 5
>  -- Executing [911@LocalSets:1] Goto("SIP/mlcm800-",
> "dialpsap,1") in new stack
>  -- Goto (LocalSets,dialpsap,1)
>  -- Executing [dialpsap@LocalSets:1]
>  Verbose("SIP/mlcm800-",
> "1,Call initiated to PSAP!") in new stack
>   Call initiated to PSAP!
>  -- Executing [dialpsap@LocalSets:2] Dial("SIP/mlcm800-",
> "DAHDI/g20/19725232703,30") in new stack
> [Oct 11 19:30:13] WARNING[3740]: app_dial.c:2218 dial_exec_full:
> Unable
> to create channel of type 'DAHDI' (cause 34 - Circuit/channel
> congestion)
>== Everyone is busy/congested at this time (1:0/1/0)
>  -- Executing [dialpsap@LocalSets:3]
>  Verbose("SIP/mlcm800-",
> "2,DIALSTATUS=CONGESTION") in new stack
>== DIALSTATUS=CONGESTION
>  -- Executing [dialpsap@LocalSets:4]
>  GotoIf("SIP/mlcm800-",
> "0?good") in new stack
>  -- Executing [dialpsap@LocalSets:5]
> SoftHangup("SIP/mlcm800-", "DAHDI/49,a") in new stack
> [Oct 11 19:30:13] WARNING[3740]: app_softhangup.c:122
> softhangup_exec:
> Soft hanging DAHDI/49-1 up.
>  -- Executing [dialpsap@LocalSets:6] Wait("SIP/mlcm800-",
> "5") in new stack
>== Spawn extension (MainMenu, s, 13) exited non-zero on
>'DAHDI/49-1'
>  -- Hanging up on 'DAHDI/49-1'
>  -- Hungup 'DAHDI/49-1'
>  -- Executing [dialpsap@LocalSets:7] Goto("SIP/mlcm800-",
> "dialit") in new stack
>  -- Goto (LocalSets,dialpsap,2)
>  -- Executing [dialpsap@LocalSets:2] Dial("SIP/mlcm800-",
> "DAHDI/g20/19725232703,30") in new stack
>  -- Called DAHDI/g20/19725232703
>  -- DAHDI/49-1 answered SIP/mlcm800-
>  -- Hanging up on 'DAHDI/49-1'
>  -- Hungup 'DAHDI/49-1'
>== Spawn extension (LocalSets, dialpsap, 2) exited non-zero on
> 'SIP/mlcm800-'

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Re: [asterisk-users] Digium D40 phones and Caller ID

[Matthew Jordan]

On 10/12/2012 10:35 AM, Christopher Harrington wrote:
> On Fri, Oct 12, 2012 at 9:59 AM, Matthew Jordan  wrote:
> Is "type=peer" strictly necessary? I don't know how they're currently
> being specified from users.conf, is that possible to specify in
> users.conf? I was under the impression that peers specified in
> users.conf would be type=friend.

Okay, using users.conf:

[peer01]
type = friend
secret = 
callerid = "D40 01" <101>
host = dynamic
nat = no
disallow = all
allow = ulaw
allow = g722
hassip = yes

After loading, [peer01] shows up as a known SIP endpoint.  Calling
peer01 displays the same caller ID as before, i.e., either 101/D40 01 or
101/foo.  Note that none of this using the DPMA either.

And as shown above, no, type = peer is not strictly necessary.  This
should not impact the display of caller ID.

>> secret = 
>> callerid = "D40 01" <101>
>> host = dynamic
> My hosts are manually specified (ie they do not register), that
> shouldn't matter, correct?

No, that shouldn't matter either.

>> sendrpid = pai
> I have this specified in the general section of sip.conf. Does this
> need to be specified per-peer?

No, that does not impact this situation.  Prior to testing this for this
e-mail I did not specify sendrpid at all; I added that to the
configuration as you called out that setting and I wanted to make sure
it didn't change the behaviour.  It did not.



> I don't know if that is necessarily true; the phones were new in box
> but were not purchased from Digium or an authorized reseller.
> 

Did they fall off the back of a truck? :-)

Call anyway.  If you purchased said phones in a legitimate manner, they
should be able to help you.

(And, as previously suggested, you may want to make sure you've upgraded
to the latest firmware, just to rule out any solved problems).

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org



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[asterisk-users] Scheduled Maintenance for Asterisk Project community services

[Asterisk Development Team]

On Monday, October 15th, 2012, the Asterisk community services
listed below will be undergoing maintenance (software upgrades and
updates). The services will be shut down at approximately 9:00 PM CDT
(2:00 AM October 16th UTC), and will return no later than 10:00 PM
CDT. We apologize in advance for any inconvenience this may cause.

The affected services are:

issues.asterisk.org/jira

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Re: [asterisk-users] Digium D40 phones and Caller ID

[A J Stiles]

On Thursday 11 October 2012, Christopher Harrington wrote:
> First post to this mailing list. I'll keep it brief: My D40 phones
> don't show the "name" component of CALLERID.
> It only displays the number.
> . [stuff deleted] .
> From what I can tell, this appears to be a Digium phone limitation. Or
> am I missing something crucial?

We have D40 phones here  (even without using that icky, non-free DPMA module)  
and Set(CALLERID(name)=...)  just works fine for us -- used to work fine with 
the Zultys phones we had, too.

Make sure you have the latest firmware on your phones.

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] Digium D40 phones and Caller ID

[Christopher Harrington]

On Fri, Oct 12, 2012 at 9:59 AM, Matthew Jordan  wrote:
> On 10/11/2012 05:39 PM, Christopher Harrington wrote:
>> First post to this mailing list. I'll keep it brief: My D40 phones
>> don't show the "name" component of CALLERID.
>> It only displays the number. This includes calls originating from PSTN
>> with their own CID already set, and calls
>> where we've specifically filled in this data. Changing the destination
>> of my test extension to a softphone (zoiper
>> in this case) correctly displays the information. sip.conf already
>> contains sendrpid=pai.
>>
>> From what I can tell, this appears to be a Digium phone limitation. Or
>> am I missing something crucial?
>>
>
> No, the D40s display the name.
>
> Using the following configuration in sip.conf:
I'm using users.conf, so my questions will mostly pertain to that. I
apologize for what I'm sure are some dumb questions up ahead here.

>
> [peer01]
> type = peer
Is "type=peer" strictly necessary? I don't know how they're currently
being specified from users.conf, is that possible to specify in
users.conf? I was under the impression that peers specified in
users.conf would be type=friend.

> secret = 
> callerid = "D40 01" <101>
> host = dynamic
My hosts are manually specified (ie they do not register), that
shouldn't matter, correct?

> sendrpid = pai
I have this specified in the general section of sip.conf. Does this
need to be specified per-peer?

> disallow = all
> allow = ulaw
> allow = g722
>
> And extensions.conf:
>
> exten => 101,1,NoOp()
> same => n,Set(CALLERID(name)=foo)
> same => n,Dial(SIP/101)
> same => n,Hangup()
This is effectively what I've done with my test extension. I've tried
both CALLERID(all)=... and CALLERID(name)=...

>
> Shows the following on the D40:
> 101
> foo
>
> If I remove the CALLERID function call, the D40 shows:
> 101
> D40 01
>
> Note that this is using the 1.1.0 firmware.  I imagine there is a
Yep, 1.1.0.0.

> configuration issue somewhere.  You may want to provide your entire
It occurs to me that you're probably using DPMA, and I am not. That's
probably where this issue is.

> configuration, or - since you have purchased phones from Digium - call
> technical support.  They should be able to help you resolve this issue.
I don't know if that is necessarily true; the phones were new in box
but were not purchased from Digium or an authorized reseller.

>
> --
> Matthew Jordan
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
>
>
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-- 
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248

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Re: [asterisk-users] Recommendation for extension mapping on inbound T1 line

[A J Stiles]

On Friday 12 October 2012, Mitch Claborn wrote:
> Converting this customer from a MiTel system to asterisk. Discovered
> that the inbound calls from the T1 are going to extension 366.  (This
> was mapped in the MiTel for some arcane purpose.)  The dial plan I am
> currently using is shown below.  When loading the dial plan, I get this
> warning:
> 
>   WARNING[5004]: pbx_config.c:1561 pbx_load_config: The use of '_.' for
> an extension is strongly discouraged and can have unexpected behavior.
> Please use '_X.' instead at line 331 of extensions.conf
> 
> Question: Do I need to worry about this warning?

You only need to worry about it if  (1)  you are using non-numeric extensions 
anywhere in your dialplan and  (2)  another context includes the "from-pstn" 
context or something might jump into it.

If  (and only if)  all the extensions you are using in all your contexts are 
numeric, then "_." is fine.  (But you don't really need it anyway in your 
example, since the "s" extension in your from-pstn context will already catch 
the incoming call.)

*But* some hardware specifically expects to work with non-numeric extensions -- 
I know from bitter experience that the OpenVox G400P/E cards do.  If you later 
install one of these cards, it will want to call extension "sms" when a text 
message comes in, and either "sms_send_ok" or "sms_send_failed" when a text 
has been sent.  If you specify the wrong context in your chan_extra.conf  (and 
Sod's Law says you *will* do that at first, while you're setting it up),  one 
of these could potentially match against "_." -- which probably is not what 
you want. 

> I'm a little leery of just using 366 in the dialplan, since the company
> we are dealing with is a little flaky.

That's quite sensible!  Some telcos do some really counter-intuitive things.  
If you have only one number for all incoming calls, a catch-all is fine.

> [from-pstn]
> exten =>s,1,NoOp(pstn call from ${CALLERID(all)} exten=${EXTEN})
>same =>n,Goto(MainMenu,s,1)
; you don't really need the following 2 lines:
> exten =>_.,1,NoOp(pstn call from ${CALLERID(all)} exten=${EXTEN})
>same =>n,Goto(MainMenu,s,1)

If you later decide to set up direct dial-in lines, and route calls depending 
on the dialled number, then you will have to do something a bit more 
sophisticated, obviously.  But cross that bridge when you come to it  :)

-- 
AJS

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Re: [asterisk-users] Digium D40 phones and Caller ID

[Matthew Jordan]

On 10/11/2012 05:39 PM, Christopher Harrington wrote:
> First post to this mailing list. I'll keep it brief: My D40 phones
> don't show the "name" component of CALLERID.
> It only displays the number. This includes calls originating from PSTN
> with their own CID already set, and calls
> where we've specifically filled in this data. Changing the destination
> of my test extension to a softphone (zoiper
> in this case) correctly displays the information. sip.conf already
> contains sendrpid=pai.
> 
> From what I can tell, this appears to be a Digium phone limitation. Or
> am I missing something crucial?
>

No, the D40s display the name.

Using the following configuration in sip.conf:

[peer01]
type = peer
secret = 
callerid = "D40 01" <101>
host = dynamic
sendrpid = pai
disallow = all
allow = ulaw
allow = g722

And extensions.conf:

exten => 101,1,NoOp()
same => n,Set(CALLERID(name)=foo)
same => n,Dial(SIP/101)
same => n,Hangup()

Shows the following on the D40:
101
foo

If I remove the CALLERID function call, the D40 shows:
101
D40 01

Note that this is using the 1.1.0 firmware.  I imagine there is a
configuration issue somewhere.  You may want to provide your entire
configuration, or - since you have purchased phones from Digium - call
technical support.  They should be able to help you resolve this issue.

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org



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Re: [asterisk-users] Recommendation for extension mapping on inbound T1 line

[Christopher Harrington]

On Fri, Oct 12, 2012 at 9:10 AM, Mitch Claborn  wrote:
> Converting this customer from a MiTel system to asterisk. Discovered that
> the inbound calls from the T1 are going to extension 366.  (This was mapped
> in the MiTel for some arcane purpose.)  The dial plan I am currently using
> is shown below.  When loading the dial plan, I get this warning:
>
>  WARNING[5004]: pbx_config.c:1561 pbx_load_config: The use of '_.' for an
> extension is strongly discouraged and can have unexpected behavior.  Please
> use '_X.' instead at line 331 of extensions.conf
>
> Question: Do I need to worry about this warning?
>From what I've seen, _X. will always match any extension starting with
a digit, whereas _. matches everything, including things other than
incoming calls (like hangups and timeouts for instance).

In any case, your question is already answered in the diagnostic
output: that pattern is strongly discouraged. So yes, you need to
worry about it. If you didn't, they wouldn't warn you about it.

>
> I'm a little leery of just using 366 in the dialplan, since the company we
> are dealing with is a little flaky.
>
>
> [from-pstn]
> exten =>s,1,NoOp(pstn call from ${CALLERID(all)} exten=${EXTEN})
>   same =>n,Goto(MainMenu,s,1)
> exten =>_.,1,NoOp(pstn call from ${CALLERID(all)} exten=${EXTEN})
>   same =>n,Goto(MainMenu,s,1)
>
>
>
> --
>
> Mitch
>
>
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-- 
-Chris Harrington
ACSDi Office: 763.559.5800
Mobile Phone: 612.326.4248

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[asterisk-users] Recommendation for extension mapping on inbound T1 line

[Mitch Claborn]

Converting this customer from a MiTel system to asterisk. Discovered 
that the inbound calls from the T1 are going to extension 366.  (This 
was mapped in the MiTel for some arcane purpose.)  The dial plan I am 
currently using is shown below.  When loading the dial plan, I get this 
warning:


 WARNING[5004]: pbx_config.c:1561 pbx_load_config: The use of '_.' for 
an extension is strongly discouraged and can have unexpected behavior.  
Please use '_X.' instead at line 331 of extensions.conf


Question: Do I need to worry about this warning?

I'm a little leery of just using 366 in the dialplan, since the company 
we are dealing with is a little flaky.



[from-pstn]
exten =>s,1,NoOp(pstn call from ${CALLERID(all)} exten=${EXTEN})
  same =>n,Goto(MainMenu,s,1)
exten =>_.,1,NoOp(pstn call from ${CALLERID(all)} exten=${EXTEN})
  same =>n,Goto(MainMenu,s,1)



--

Mitch


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[asterisk-users] SoftHangup for emergency calls

[Mitch Claborn]

Setting up a group of analog lines to use for outbound emergency calls 
(911).  My current dial plan and debug output shown below.  It appears 
that when the SoftHangup() is executed that the line does not really 
hang up.  In the case shown, I had reduced the group to a single DAHDI 
(analog) channel and dialed in to that number from the outside. You can 
see in the output that the SoftHangup() was executed, but the call was 
not terminated - the outside caller stayed connected to something.  
Caller no longer heard the sounds from the menu he was in, but the call 
itself seemed to stay connected.


Asterisk 1.8 on Ubuntu

Any ideas?

[emergency-services]
exten =>911,1,Goto(dialpsap,1)
exten =>9911,1,Goto(dialpsap,1)
exten =>999,1,Goto(dialpsap,1)
exten =>112,1,Goto(dialpsap,1)

exten =>dialpsap,1,Verbose(1,Call initiated to PSAP!)
  same =>n(dialit),Dial(${LOCAL}/${EMERGENCY},30)
  same =>n,Verbose(2,DIALSTATUS=${DIALSTATUS})
  same =>n,GotoIf($["${DIALSTATUS}" = "ANSWER"]?good)
  same =>n(hu),SoftHangup(${EMERGENCY_CHANNEL},a)
  same =>n,Wait(5)
  same =>n,Goto(dialit)
  same =>n(good),NoOp(call good)
  same =>n,Hangup()


  == Using SIP RTP CoS mark 5
-- Executing [911@LocalSets:1] Goto("SIP/mlcm800-", 
"dialpsap,1") in new stack

-- Goto (LocalSets,dialpsap,1)
-- Executing [dialpsap@LocalSets:1] Verbose("SIP/mlcm800-", 
"1,Call initiated to PSAP!") in new stack

 Call initiated to PSAP!
-- Executing [dialpsap@LocalSets:2] Dial("SIP/mlcm800-", 
"DAHDI/g20/19725232703,30") in new stack
[Oct 11 19:30:13] WARNING[3740]: app_dial.c:2218 dial_exec_full: Unable 
to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)

  == Everyone is busy/congested at this time (1:0/1/0)
-- Executing [dialpsap@LocalSets:3] Verbose("SIP/mlcm800-", 
"2,DIALSTATUS=CONGESTION") in new stack

  == DIALSTATUS=CONGESTION
-- Executing [dialpsap@LocalSets:4] GotoIf("SIP/mlcm800-", 
"0?good") in new stack
-- Executing [dialpsap@LocalSets:5] 
SoftHangup("SIP/mlcm800-", "DAHDI/49,a") in new stack
[Oct 11 19:30:13] WARNING[3740]: app_softhangup.c:122 softhangup_exec: 
Soft hanging DAHDI/49-1 up.
-- Executing [dialpsap@LocalSets:6] Wait("SIP/mlcm800-", 
"5") in new stack

  == Spawn extension (MainMenu, s, 13) exited non-zero on 'DAHDI/49-1'
-- Hanging up on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
-- Executing [dialpsap@LocalSets:7] Goto("SIP/mlcm800-", 
"dialit") in new stack

-- Goto (LocalSets,dialpsap,2)
-- Executing [dialpsap@LocalSets:2] Dial("SIP/mlcm800-", 
"DAHDI/g20/19725232703,30") in new stack

-- Called DAHDI/g20/19725232703
-- DAHDI/49-1 answered SIP/mlcm800-
-- Hanging up on 'DAHDI/49-1'
-- Hungup 'DAHDI/49-1'
  == Spawn extension (LocalSets, dialpsap, 2) exited non-zero on 
'SIP/mlcm800-'





--

Mitch


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Re: [asterisk-users] Tips for installing and configuring Digum cards

[Mitch Claborn]

Last night we did a trial run.  I am happy to report that both analog 
and T1 lines worked well with the config files generated by 
dahdi_genconf.  Had a couple of minor issues that I'll ask about in 
separate posts.


Of course when we got on-site, discovered that customer really has 6 
analog lines instead of just 4.  Hopefully the card I ordered last night 
will make it here by Saturday.



Mitch

On 10/11/2012 09:40 AM, Jeff LaCoursiere wrote:




Totally typical.  I don't think you will have any issues.

j


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Re: [asterisk-users] Asterisk 1.8 - ADDMEMBER event in queue_log not using member name [SOLVED]

[Matthew Jordan]

On 10/12/2012 02:16 AM, Olivier wrote:
> 
> 
> Yes I agree that new features should be committed to new versions but in
> this specific case, current 1.8 behaviour is all Queue events  but two
> (ADDMEMBER and REMOVEMEMBER) are using members name for logging into
> queue_log.
> 
> So, to me, queue_log content is currently inconsistent.
> 
> But as I also understand current installs might already have found a way
> around (what I rate as a bug), maybe a configuration option would be
> acceptable for all.
> 
> What do you think ?

Configuration options such as this are already rampant throughout
Asterisk.  They make the codebase harder to maintain; they make Asterisk
more difficult to configure; they increase the burden on those who
attempt to document these options; they make it more difficult to
provide correct behaviour in future versions.  Making this behaviour
configurable implies that future versions of Asterisk will want to
maintain the current Queue log behaviour, which is not the case.

The only time to have these kinds of configuration parameters is when
the behaviour is so onerous that we have to backport the behavioural
change, or when the two behaviours should be mutually supported for all
future versions.  I don't believe either is true in this situation.

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org



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Re: [asterisk-users] DTMF digits are coming through twice

[Vik Killa]

Sorry the attachment was too big. here is link:
http://www.2shared.com/file/Ola640Pn/doubledigit.html


On Fri, Oct 12, 2012 at 9:24 AM, SamyGo  wrote:
> Why am I feeling like I'm the only one here who is not able to see any
> pastebin link or attachments in this thread !
>
>

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Re: [asterisk-users] DTMF digits are coming through twice

[SamyGo]

Why am I feeling like I'm the only one here who is not able to see any
pastebin link or attachments in this thread !

On Fri, Oct 12, 2012 at 6:18 PM, Vik Killa  wrote:

> The trace is attached 3 emails back.
>
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Re: [asterisk-users] DTMF digits are coming through twice

[Vik Killa]

The trace is attached 3 emails back.

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Re: [asterisk-users] How to use 'Transfer' to send calls to another asterisk?

[Joshua Colp]

Deepesh D wrote:

This doesn't work reliably well with all all clients. I tested it
using a zoiper soft phone and it worked. But from an ATA device it
failed. On the S2 server it failed to authenticate

The console of S2 showed
[Oct 12 18:21:06] WARNING[30483]: chan_sip.c:13952 check_auth:
username mismatch, have, digest has<>
[Oct 12 18:21:06] NOTICE[30483]: chan_sip.c:22046
handle_request_invite: Failed to authenticate device
;tag=3047


Yes, it all depends on how the implementation in question handles it and 
how smart it is. You can't expect them all to behave as I mentioned. In 
the case of Asterisk it treats it as a SIP URI when promiscredir is 
enabled and doesn't use a peer entry, so it has no idea how to authenticate.


Cheers,

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at:  www.digium.com  & www.asterisk.org

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Re: [asterisk-users] How to use 'Transfer' to send calls to another asterisk?

[Deepesh D]

This doesn't work reliably well with all all clients. I tested it
using a zoiper soft phone and it worked. But from an ATA device it
failed. On the S2 server it failed to authenticate

The console of S2 showed
[Oct 12 18:21:06] WARNING[30483]: chan_sip.c:13952 check_auth:
username mismatch, have , digest has <>
[Oct 12 18:21:06] NOTICE[30483]: chan_sip.c:22046
handle_request_invite: Failed to authenticate device
;tag=3047

On Fri, Oct 12, 2012 at 5:00 PM, Joshua Colp  wrote:
> Deepesh D wrote:
>>
>> I made these changes in dialplan and it worked. Thanks a lot.
>>
>> In most of the cases S1, S2 and C1 are in my control. But in some
>> cases the dialplan of C1 is not in my control. Also in some cases C1
>> can be any SIP client like a softphone or SIP device, so it wont work
>> in those case. Is there some way I can get those also working.
>
>
> You can certainly execute Transfer() and most clients will then send the
> call to where you have specified. If you have the same users on all possible
> servers you would Transfer to with the same username/password a challenge
> should occur and authentication happen.
>
> I haven't tested that though.
>
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at:  www.digium.com  & www.asterisk.org
>
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Re: [asterisk-users] DTMF digits are coming through twice

[SamyGo]

Well like I said before and you increased my doubt as well by saying that
this happens from callers using E-link internet. Can you share the trace !

On Fri, Oct 12, 2012 at 5:24 PM, Vik Killa  wrote:

> Any ideas?
>
> On Thu, Oct 11, 2012 at 2:32 PM, Vik Killa  wrote:
> > Call was to 7167436110
> >
>
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Re: [asterisk-users] DTMF digits are coming through twice

[Vik Killa]

Any ideas?

On Thu, Oct 11, 2012 at 2:32 PM, Vik Killa  wrote:
> Call was to 7167436110
>

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[asterisk-users] dropping audio on avaya

[Jerry Geis]

I am using 1.4.43 connected on PRI to avaya PBX.

If I call one extension through the PRI and speak a message (recorded 
file) sounds fine.
If I call an extension through the PRI that brings together a group of 
phones on the avaya side

and play the same recorded file the audio drops out.

What might that be from?

THanks,

Jerry

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Re: [asterisk-users] How to use 'Transfer' to send calls to another asterisk?

[Joshua Colp]

Deepesh D wrote:

I made these changes in dialplan and it worked. Thanks a lot.

In most of the cases S1, S2 and C1 are in my control. But in some
cases the dialplan of C1 is not in my control. Also in some cases C1
can be any SIP client like a softphone or SIP device, so it wont work
in those case. Is there some way I can get those also working.


You can certainly execute Transfer() and most clients will then send the 
call to where you have specified. If you have the same users on all 
possible servers you would Transfer to with the same username/password a 
challenge should occur and authentication happen.


I haven't tested that though.

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Re: [asterisk-users] How to use 'Transfer' to send calls to another asterisk?

[Deepesh D]

I made these changes in dialplan and it worked. Thanks a lot.

In most of the cases S1, S2 and C1 are in my control. But in some
cases the dialplan of C1 is not in my control. Also in some cases C1
can be any SIP client like a softphone or SIP device, so it wont work
in those case. Is there some way I can get those also working.

On Fri, Oct 12, 2012 at 1:12 AM, Joshua Colp  wrote:
> Deepesh D wrote:
>>
>> If C1 dials S1 and then S1 dials S2 to transfer the call then S1 still
>> remains in the loop till the call is finished. What I wanted to do is
>> to reduce the number of calls on S1, so as soon as S1 receives a call
>> from C1 it redirects the call to S2 using 'Transfer' application and
>> exits from the loop, the call should now be handled by S2
>
>
> With some crafty configuration you can achieve this using Transfer. With
> promiscredir disabled Asterisk will not follow the SIP URI in the 302
> response sent back as a result of calling Transfer. The call reenters the
> dialplan at the user portion of the URI passed back and executes dialplan as
> normal. By prefixing the user portion with a unique identifier you can write
> dialplan that strips the prefix and then dials out to S2.
>
> Flow being:
>
> C1 executes Dial(SIP/${EXTEN}@S1) (matched using _1NXXNXX)
> S1 executes Transfer(002${EXTEN}) (matched using _1NXXNXX)
> C1 executes Dial(SIP/${EXTEN:3}@S2) (matched using _0021NXXNXX)
>
> So simply:
> C1 calls S1
> S1 decides to send it to S2, but wants to tell C1 to do it directly
> S1 sends back a SIP message saying hey call 00218005551212 instead
> C1 matches the number to the dialplan which explicitly calls out through S2
> S2 receives the call and life is good
>
> * Note: This requires control over both C1 and S1, you can't just do it to
> every random system calling.
>
> If you don't get the finer details of this just experiment with the general
> idea and configuration option I mentioned. It won't hurt.
>
> Cheers,
>
> --
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at:  www.digium.com  & www.asterisk.org
>
>
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[asterisk-users] asterisk 1.8 app_monitor problem

[cfh]

Hi

I have just update my old asterisk 1.4 to 1.8 version and all works good 
but I have a problem with the monitor features.


With the old version this part of dialplan worked without problem :

exten => _90.,1,NoOp(call out interoute REC)
exten => _90.,n,Monitor(wav,${CALLERID(num)}-${EXTEN:2}})
exten => _90.,n,Dial(SIP/provider/${EXTEN},90)


now with 1.8 the same dialplan with the same hw sometimes causes lost 
audio call problem



server: debian 6.0.6

asterisk 1:1.8.11.1-1digium1~squeeze



what can I do ?


best

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Re: [asterisk-users] Asterisk 1.8 - ADDMEMBER event in queue_log not using member name [SOLVED]

[Olivier]

2012/10/11 Kinsey Moore 

> On 10/11/2012 10:31 AM, Olivier wrote:
>
>> 2012/10/11 Kinsey Moore mailto:kmo...@digium.com>>
>>
>>
>> Hi Olivier,
>> >My questions are:
>> >1. Is it possible to configure Asterisk 1.8 (I'm using 1.8.12 but
>> I can
>> >upgrade to 1.8.17 if, and only if, necessary) so that ADDMEMBER
>> entries in
>> >queue_log refers to member name instead of member location ?
>> >If positive how should this be configured ?
>>
>> This feature is not present in Asterisk 1.8.17.  It was only
>> committed to trunk since it is a change in behavior.  It is in the
>> Asterisk 11 branch, betas, and release candidate.
>>
>>
>> Now I'm seeing the related commit in Asterisk 11 changelogs (I closely
>> looked at 1.8 changelog but I didn't looked at Asterisk 11 branch).
>> I don't know if the corresponding patch is invasive or not but  having
>> the option to change Asterisk 1.8 behaviour would have been much
>> appreciated.
>>
>> Is there any change to have this option considered ?
>>
> It is highly unlikely that this feature will go into an existing branch.
>  In general, new features are not committed to released branches.
>

Yes I agree that new features should be committed to new versions but in
this specific case, current 1.8 behaviour is all Queue events  but two
(ADDMEMBER and REMOVEMEMBER) are using members name for logging into
queue_log.

So, to me, queue_log content is currently inconsistent.

But as I also understand current installs might already have found a way
around (what I rate as a bug), maybe a configuration option would be
acceptable for all.

What do you think ?



>
> Kinsey Moore
>
>
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