[asterisk-users] Location with PRI / Analog lines

2010-05-18 Thread mir shahnawaz 
Hi there,

I am stuck with the location issues. It would be easy if you have DID
for each extension so that outgoing caller id would be DID of the
respective extension and also physical address. Now if you are not
able to get DID's for some reason. I am thinking of some situations
and appreciate your thoughts.

1. If I have a PRI and map physical number (original numbers in hunt
group not the group number) to some extensions. If somebody calls
emergency from a specific location in a building either it will have
outgoing caller ID of that specific number in PRI group (if possible)
or always dial that physical line which has address of that location
in Telco database. Is it possible? Can we have different caller ID's
for a PRI? I mean my PRi has one number 12345667 known to outside
world. But it has 23 physical number in original.Please comment

2.If I have analog lines,  incoming call can use any channel and out
going calls from any extension can use any channel. If somebody dials
emergency then that specific extension dials specific channel which
has its physical location in Telco database.

I would highly appreciate your thoughts.

Shahnawaz Mir

http://www.aaanetworkx.com

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[asterisk-users] Michael Wegner

2010-04-25 Thread mir shahnawaz 
http://www.villasantilles.com/home.php

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Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk

2010-03-30 Thread mir shahnawaz 
Xorcom XR005 is highly recommended. They work great.

http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=120550484883ssPageName=STRK:MESELX:IT

Smir

On Tue, Mar 30, 2010 at 7:34 AM, Juan Miguel jmigu...@gmail.com wrote:
 Hello Joseph

 I recommend that you use The Mediatrix 4100 Series are very good.

 Juan M.

 2010/3/28 Joseph Begumisa j.begum...@gmail.com

 Hi,
 Can anyone recommend a 24 fxs port voip gateway that has worked well with
 asterisk?  I have a couple of analog handsets that I want to hookup to my
 asterisk server?  Any tested and tried product recommendations are welcome.
  Thanks.
 Best Regards,

 Joseph

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[asterisk-users] Priority based softhangup

2010-03-30 Thread mir shahnawaz 
Hi,

Is it possible to softhangup a channel based on priority. I mean I
want to put some calls in higher priority lets say 100. If all
channels are busy and somebody wants to dial an extension with
priority higher than 100. How can softhangup drop a line which has
priority less than 100? I will appreciate your valuable help.

Thanks

Smir

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Re: [asterisk-users] Priority based softhangup

2010-03-30 Thread mir shahnawaz 
Thanks Danny,

Can you please give me idea about Softhang.agi

Thanks

Smir

On Tue, Mar 30, 2010 at 1:31 PM, Danny Nicholas da...@debsinc.com wrote:
 AIUI, softhangup is strictly an address-type function.  Using Steve's
 suggestion, you could set a key with the priority at dial-time and when a
 congested condition occurred, match the database for the lowest priority to
 hangup. Something like this:
 Exten = 100,1,noop(prioritized dialing)
 Exten = 100,n,Set(callpri=${DB(Callpri/${EXTEN})})
 Exten = 100,n,Dial..
 Exten = 100-CONGESTION(softhang.agi,${callpri})

 Softhang.agi is a program you write to see if the callpri is high enough to
 cancel another call and dial or send back a message.


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
 Sent: Tuesday, March 30, 2010 2:09 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Priority based softhangup

 On Tue, 30 Mar 2010, mir shahnawaz wrote:

 Is it possible to softhangup a channel based on priority. I mean I want
 to put some calls in higher priority lets say 100. If all channels are
 busy and somebody wants to dial an extension with priority higher than
 100. How can softhangup drop a line which has priority less than 100? I
 will appreciate your valuable help.

 In Asterisk, a priority is the step number in a dialplan. For example:

        exten = *,5,    verbose(1,foo)

 5 is the priority.

 It sounds like you want to have an executive class that can step on an
 underling's call. Maybe setting a global channel variable or tracking
 resource usage in a database would work.

 --
 Thanks in advance,
 -
 Steve Edwards       sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
 Newline                                              Fax: +1-760-731-3000

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Re: [asterisk-users] Priority based softhangup

2010-03-30 Thread mir shahnawaz 
Thanks Steve and Danny for your help.

S Mir

On Tue, Mar 30, 2010 at 3:16 PM, Steve Edwards
asterisk@sedwards.com wrote:
 Un-top-posting...

 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mir
 shahnawaz

 Can you please give me idea about Softhang.agi

 On Tue, 30 Mar 2010, Danny Nicholas wrote:

 Simple code
 In PERL -
 -- agi header stuff --
 My ($pri, $callno) = @ARGV;
 -- read database to get priority of lowest active call, return $lowpri,
 $lowid --
 if ($pri  $lowpri) {
   print STDOUT exec softhangup $lowid\n;
   -- AMI dial $callno --
   }
 Else {
   Print STDOUT exec BACKGROUND congested message\n;
   }

 You can find real examples on voip-info.org.

 mir, please keep in mind the above is meant to give you an idea, not
 intended as literal code.

 You will save yourself a lot of frustration if you use an existing AGI
 library for your chosen language*. While conceptually simple, nobody
 gets the AGI protocol right the first time.

 *) I write most of my AGIs in C because it is the language I know best and
 you can execute hundreds of AGIs written in C in the time it takes to load
 the Perl or PHP interpreter and parse your script.

 --
 Thanks in advance,
 -
 Steve Edwards       sedwa...@sedwards.com      Voice: +1-760-468-3867 PST
 Newline                                              Fax: +1-760-731-3000

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[asterisk-users] softhangup

2010-03-16 Thread mir shahnawaz 
Hi all,

I am trying to drop a random  channel in asterisk 1.6. The following
line in extensions.conf works fine for the first channel

exten = 911,4,SoftHangup(DAHDI/1-1)

But I need to drop random channel for emergency not any specific one.
Can someone show correct syntax for this

Thanks

smir

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[asterisk-users] softhangup

2010-03-16 Thread mir shahnawaz 
Hi all,

I am trying to drop a random  channel in asterisk 1.6. The following
line in extensions.conf works fine for the first channel

exten = 911,4,SoftHangup(DAHDI/1-1)

But I need to drop random channel for emergency not any specific one.
Can someone show correct syntax for this

Thanks

smir

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Re: [asterisk-users] Asterisk on MPLS VPN

2010-03-13 Thread mir shahnawaz 
Are you able to ping 172.16.0.1 from your router (which is connected
to MPLS Network). If not then you have to tell your MPLS router where
is 172.16.0.0 connected. Do you have access on this router? If you
copy paste config here that would be helpful. I suspect that your ping
request is reaching MPLS router but router does not have a route for
this network. If this router is on the other side of MPLS network then
it is possible that ping is blocked by your service provider to hide
intermediate nodes.
 I don't think that you need any tunneling protocol for this as it is
already taken care by your provider.

Shahnawaz

AAA Networkx

On Sat, Mar 13, 2010 at 11:11 AM, Tilghman Lesher tles...@digium.com wrote:
 On Saturday 13 March 2010 09:56:23 Sriram wrote:
 Sorry, 172.16.0.1 is the local IP address and it is connected to the MPLS
 Router whose WAN interface is 10.224.6.121.this is for first NIC. Second
 NIC has a public IP. Now if I ping yahoo.com I get a reply and if I ping
 the 10.224.6.121 it just hangs on (no response). If I do ifup eth0 and fire
 ping yahoo.com I don't get any response but I get reply on 10.224.6.121

 Generally with MPLS, the packets themselves cannot have any IP addresses
 other than the addresses for the endpoints.  So what you need to do is to use
 a tunneling protocol for any packets which must be relayed beyond the MPLS
 infrastructure.  My personal favorite is openvpn, since knowing that tool can
 assist in creating cross-platform VPNs with outside computers.  However,
 openvpn can additionally merely encapsulate local connections through your
 MPLS infrastructure.  Remember that you need to set the source IP for any
 packets traveling through MPLS (using the 'ip route' command and the 'src'
 argument).

 --
 Tilghman Lesher
 Digium, Inc. | Senior Software Developer
 twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
 Check us out at: www.digium.com  www.asterisk.org

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[asterisk-users] 911, channel full

2010-03-03 Thread mir shahnawaz 
Hi,

I am trying to implement 911 funtionality in my PBX. A call should
drop if all lines are busy. Here is my context nineoneone from
extensions.conf

[nineoneone]
exten = s,1,Set(SET_EMERG_FLAG=0)
exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten = s,n,Set(EMERGENCY=1,g)
exten = s,n,Set(SET_EMERG_FLAG=1)
exten = s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
exten = s,s+2(trunkbusy),GotoIf($[${EMERGENCY}=1]?inprogress)
exten = s,n,SoftHangup(${EMERGENCY_TRUNK}-1)
exten = s,n,Wait(12)
exten = s,n,Goto(checkavail)
exten = s,s+2(inprogress),Congestion
exten = s,checkavail+101(notavail),Goto(trunkbusy)
exten = h,1,GotoIf($[${SET_EMERG_FLAG}=1]?3)
exten = h,3,Set(EMERGENCY=0,g)

If all lines connecting to PSTN are busy. I get busy tone upon dialing
911 and following message is generated by CLI.

app_dial.c:1547 dial_exec_full: Unable to create channel of type
'DAHDI' (cause 34 - Circuit/channel congestion)

I would appreciate if somebody help me solve this issue.

Regards

Shahnawaz

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Re: [asterisk-users] 911, channel full

2010-03-03 Thread mir shahnawaz 
Thanks for your reply. This all I have, am I missing something? Please
help in this regard. Here is full output from CLI

  -- Executing [...@default:1] Goto(SIP/501-0137,
nineoneone,s,1) in new stack
-- Goto (nineoneone,s,1)
-- Executing [...@nineoneone:1] Set(SIP/501-0137,
SET_EMERG_FLAG=0) in new stack
-- Executing [...@nineoneone:2] ChanIsAvail(SIP/501-0137,
DAHDI/g0) in new stack
-- Executing [...@nineoneone:3] Set(SIP/501-0137,
EMERGENCY=1,g) in new stack
-- Executing [...@nineoneone:4] Set(SIP/501-0137,
SET_EMERG_FLAG=1) in new stack
-- Executing [...@nineoneone:5] Dial(SIP/501-0137,
DAHDI/g0/91234567) in new stack
[Mar  3 11:26:06] WARNING[28572]: app_dial.c:1547 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)
-- Auto fallthrough, channel 'SIP/501-0137' status is 'CONGESTION'

Regards

Shahnawaz
On Wed, Mar 3, 2010 at 10:54 AM, Steve Howes steve-li...@geekinter.net wrote:

 On 3 Mar 2010, at 17:21, mir shahnawaz wrote:
 [nineoneone]
 exten = s,1,Set(SET_EMERG_FLAG=0)
 exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
 exten = s,n,Set(EMERGENCY=1,g)
 exten = s,n,Set(SET_EMERG_FLAG=1)
 exten = s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
 exten = s,s+2(trunkbusy),GotoIf($[${EMERGENCY}=1]?inprogress)
 exten = s,n,SoftHangup(${EMERGENCY_TRUNK}-1)
 exten = s,n,Wait(12)
 exten = s,n,Goto(checkavail)
 exten = s,s+2(inprogress),Congestion
 exten = s,checkavail+101(notavail),Goto(trunkbusy)
 exten = h,1,GotoIf($[${SET_EMERG_FLAG}=1]?3)
 exten = h,3,Set(EMERGENCY=0,g)

 If all lines connecting to PSTN are busy. I get busy tone upon dialing
 911 and following message is generated by CLI.

 app_dial.c:1547 dial_exec_full: Unable to create channel of type
 'DAHDI' (cause 34 - Circuit/channel congestion)

 Can you tell us the other lines too? i.e. the bit where it attempts to
 actually do the hangup..

 S

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[asterisk-users] dropping line (s) for 911

2010-02-12 Thread mir shahnawaz 
Hi all,

I am trying to implement call dropping funtionality in asterisk for
911. I mean if all lines are busy and someone wants to dial 911 at
least one line should be dropped. Here is my extensions.conf which i
copied from internet. Could somebody help me figure out what is wrong.
Thanks in advance.


[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=DAHDI/1
;CONSOLE=Phone/phone0
TRUNK=DAHDI/g0  ; Trunk interface
EMERGENCY=0
EMERGENCY_TRUNK=DAHDI/g0
EMERGENCY_NUM=12345678 (for testing)


[default]

exten = 911,1,Goto(nineoneone,s,1)


[nineoneone]
exten = s,1,Set(SET_EMERG_FLAG=0)
exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten = s,n,Set(EMERGENCY=1,g)
exten = s,n,Set(SET_EMERG_FLAG=1)
exten = s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
exten = s,s+2(trunkbusy),GotoIf($[${EMERGENCY}=1]?inprogress)
exten = s,n,SoftHangup(${EMERGENCY_TRUNK}-1)
exten = s,n,Wait(12)
exten = s,n,Goto(checkavail)
exten = s,s+2(inprogress),Congestion
exten = s,checkavail+101(notavail),Goto(trunkbusy)
exten = h,1,GotoIf($[${SET_EMERG_FLAG}=1]?3)
exten = h,3,Set(EMERGENCY=0,g)


Regards

Shahnawaz Mir

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Re: [asterisk-users] dropping line (s) for 911

2010-02-12 Thread mir shahnawaz 
I got it working now. I was not including context ninioneone in default context.


On Fri, Feb 12, 2010 at 1:49 PM, mir shahnawaz shahnawaz...@gmail.com wrote:
 Hi all,

 I am trying to implement call dropping funtionality in asterisk for
 911. I mean if all lines are busy and someone wants to dial 911 at
 least one line should be dropped. Here is my extensions.conf which i
 copied from internet. Could somebody help me figure out what is wrong.
 Thanks in advance.


 [globals]
 CONSOLE=Console/dsp                             ; Console interface for demo
 ;CONSOLE=DAHDI/1
 ;CONSOLE=Phone/phone0
 TRUNK=DAHDI/g0                                  ; Trunk interface
 EMERGENCY=0
 EMERGENCY_TRUNK=DAHDI/g0
 EMERGENCY_NUM=12345678 (for testing)


 [default]

 exten = 911,1,Goto(nineoneone,s,1)


 [nineoneone]
 exten = s,1,Set(SET_EMERG_FLAG=0)
 exten = s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
 exten = s,n,Set(EMERGENCY=1,g)
 exten = s,n,Set(SET_EMERG_FLAG=1)
 exten = s,n(dial),Dial(${EMERGENCY_TRUNK}/${EMERGENCY_NUM})
 exten = s,s+2(trunkbusy),GotoIf($[${EMERGENCY}=1]?inprogress)
 exten = s,n,SoftHangup(${EMERGENCY_TRUNK}-1)
 exten = s,n,Wait(12)
 exten = s,n,Goto(checkavail)
 exten = s,s+2(inprogress),Congestion
 exten = s,checkavail+101(notavail),Goto(trunkbusy)
 exten = h,1,GotoIf($[${SET_EMERG_FLAG}=1]?3)
 exten = h,3,Set(EMERGENCY=0,g)


 Regards

 Shahnawaz Mir


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[asterisk-users] 911, location

2010-01-28 Thread mir shahnawaz 
Hi there,

I am running a PBX under asterisk 1.6. I have few FXO analogue lines
connecting to PSTN. These lines are in a hunt group. I trying to make
my extensions to dial 91, but this is a bit scary, I mean if somebody
make an emergency call after hours and without completing call is not
able to tell his/her location. How can I make 911 call center to know
the exact location of my extension. I think its possible by having
DID's but I am looking for other options too. I would appreciate your
valuable ideas and suggestions.

Thanks in advance

Shahnawaz

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Re: [asterisk-users] 911, location

2010-01-28 Thread mir shahnawaz 
Thanks for your reply. Yes POTS lines are coming into the building but
I have multiple rooms. Suppose a person is working late hours and have
a heart attack. How could 911 locate the room when no or few people
around.

Thanks

 Smir

On Thu, Jan 28, 2010 at 11:46 AM, Barry L. Kline blkl...@attglobal.net wrote:
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1

 mir shahnawaz wrote:
 Hi there,

 I am running a PBX under asterisk 1.6. I have few FXO analogue lines
 connecting to PSTN. These lines are in a hunt group. I trying to make
 my extensions to dial 91, but this is a bit scary, I mean if somebody
 make an emergency call after hours and without completing call is not
 able to tell his/her location. How can I make 911 call center to know
 the exact location of my extension. I think its possible by having
 DID's but I am looking for other options too. I would appreciate your
 valuable ideas and suggestions.

 If you're using POTS lines to make the call to 911 they'll know the
 location, if the POTS lines come into the building that you're calling
 from.  Are you saying that these lines are located in a different location?

 Barry


 - --

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 Version: GnuPG v1.4.5 (GNU/Linux)

 iD8DBQFLYdtpCFu3bIiwtTARAhZyAJ941RBJz615PkJkOBLkWF8WalaMTgCfRsdR
 UnWTQQ1anTXtDqfk54QVj/k=
 =LtAE
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Re: [asterisk-users] 911, location

2010-01-28 Thread mir shahnawaz 
Thanks, Could you please explain this little bit more. I am not
getting IMAT=EXTEN.



On Thu, Jan 28, 2010 at 12:15 PM, Danny Nicholas da...@debsinc.com wrote:
 Here's one solution:
 - exten = _911,1,Set(IMAT=EXTEN)
 - exten = _911,2,Set(IMAT=CUT(IMAT|\/|2)
 - exten = _911,3,Dial(DAHDI/1,w911)
 - exten = _911,4,Background(emergencyin${IMAT})

 Where you would record /var/lib/asterisk/sound/emergencyin100 for extension
 100, etc.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mir shahnawaz
 Sent: Thursday, January 28, 2010 12:59 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] 911, location

 Thanks for your reply. Yes POTS lines are coming into the building but
 I have multiple rooms. Suppose a person is working late hours and have
 a heart attack. How could 911 locate the room when no or few people
 around.

 Thanks

  Smir

 On Thu, Jan 28, 2010 at 11:46 AM, Barry L. Kline blkl...@attglobal.net
 wrote:
 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1

 mir shahnawaz wrote:
 Hi there,

 I am running a PBX under asterisk 1.6. I have few FXO analogue lines
 connecting to PSTN. These lines are in a hunt group. I trying to make
 my extensions to dial 91, but this is a bit scary, I mean if somebody
 make an emergency call after hours and without completing call is not
 able to tell his/her location. How can I make 911 call center to know
 the exact location of my extension. I think its possible by having
 DID's but I am looking for other options too. I would appreciate your
 valuable ideas and suggestions.

 If you're using POTS lines to make the call to 911 they'll know the
 location, if the POTS lines come into the building that you're calling
 from.  Are you saying that these lines are located in a different
 location?

 Barry


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[asterisk-users] chan_dahdi.conf for TDM404E

2009-12-11 Thread mir shahnawaz 
Hi there,

I am trying to configure chan_dahdi.conf for TDM404E. Should I
separate channels for dialing out and recieveing calls on this card or
should I leave it random so that outgoing and incoming call get first
available channels.

;FXO Modules
group = 2
echocancel = yes
signalling = fxs_ks
context = Incoming

Is it possible to define more than one context here as above mentioned
config is serving only context incoming. Please help in this regard.

Thanks

Shahnawaz

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