[asterisk-users] chan_dahdi.conf for TDM404E
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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 911, location
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 > > > - -- > > -BEGIN PGP SIGNATURE- > Version: GnuPG v1.4.5 (GNU/Linux) > > iD8DBQFLYdtpCFu3bIiwtTARAhZyAJ941RBJz615PkJkOBLkWF8WalaMTgCfRsdR > UnWTQQ1anTXtDqfk54QVj/k= > =LtAE > -END PGP SIGNATURE- > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 911, location
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 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
> 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
>>
>>
>> - --
>>
>> -BEGIN PGP SIGNATURE-
>> Version: GnuPG v1.4.5 (GNU/Linux)
>>
>> iD8DBQFLYdtpCFu3bIiwtTARAhZyAJ941RBJz615PkJkOBLkWF8WalaMTgCfRsdR
>> UnWTQQ1anTXtDqfk54QVj/k=
>> =LtAE
>> -END PGP SIGNATURE-
>>
>> --
>> _
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>>
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>
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[asterisk-users] dropping line (s) for 911
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(${EMERGENCY_TRUNK}-1)
exten => s,n,Wait(12)
exten => s,n,Goto(checkavail)
exten => s,s+2(inprogress),Congestion
exten => s,checkavail+101(notavail),Goto(trunkbusy)
exten => h,1,GotoIf($[${SET_EMERG_FLAG}=1]?3)
exten => h,3,Set(EMERGENCY=0,g)
Regards
Shahnawaz Mir
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Re: [asterisk-users] dropping line (s) for 911
I got it working now. I was not including context ninioneone in default context.
On Fri, Feb 12, 2010 at 1:49 PM, mir shahnawaz wrote:
> Hi all,
>
> I am trying to implement call dropping funtionality in asterisk for
> 911. I mean if all lines are busy and someone wants to dial 911 at
> least one line should be dropped. Here is my extensions.conf which i
> copied from internet. Could somebody help me figure out what is wrong.
> Thanks in advance.
>
>
> [globals]
> CONSOLE=Console/dsp ; Console interface for demo
> ;CONSOLE=DAHDI/1
> ;CONSOLE=Phone/phone0
> TRUNK=DAHDI/g0 ; Trunk interface
> EMERGENCY=0
> EMERGENCY_TRUNK=DAHDI/g0
> EMERGENCY_NUM=12345678 (for testing)
>
>
> [default]
>
> exten => 911,1,Goto(nineoneone,s,1)
>
>
> [nineoneone]
> exten => s,1,Set(SET_EMERG_FLAG=0)
> exten => s,n(checkavail),ChanIsAvail(${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(${EMERGENCY_TRUNK}-1)
> exten => s,n,Wait(12)
> exten => s,n,Goto(checkavail)
> exten => s,s+2(inprogress),Congestion
> exten => s,checkavail+101(notavail),Goto(trunkbusy)
> exten => h,1,GotoIf($[${SET_EMERG_FLAG}=1]?3)
> exten => h,3,Set(EMERGENCY=0,g)
>
>
> Regards
>
> Shahnawaz Mir
>
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[asterisk-users] 911, channel full
Hi,
I am trying to implement 911 funtionality in my PBX. A call should
drop if all lines are busy. Here is my context nineoneone from
extensions.conf
[nineoneone]
exten => s,1,Set(SET_EMERG_FLAG=0)
exten => s,n(checkavail),ChanIsAvail(${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(${EMERGENCY_TRUNK}-1)
exten => s,n,Wait(12)
exten => s,n,Goto(checkavail)
exten => s,s+2(inprogress),Congestion
exten => s,checkavail+101(notavail),Goto(trunkbusy)
exten => h,1,GotoIf($["${SET_EMERG_FLAG}"="1"]?3)
exten => h,3,Set(EMERGENCY=0,g)
If all lines connecting to PSTN are busy. I get busy tone upon dialing
911 and following message is generated by CLI.
app_dial.c:1547 dial_exec_full: Unable to create channel of type
'DAHDI' (cause 34 - Circuit/channel congestion)
I would appreciate if somebody help me solve this issue.
Regards
Shahnawaz
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Re: [asterisk-users] 911, channel full
Thanks for your reply. This all I have, am I missing something? Please
help in this regard. Here is full output from CLI
-- Executing [...@default:1] Goto("SIP/501-0137",
"nineoneone,s,1") in new stack
-- Goto (nineoneone,s,1)
-- Executing [...@nineoneone:1] Set("SIP/501-0137",
"SET_EMERG_FLAG=0") in new stack
-- Executing [...@nineoneone:2] ChanIsAvail("SIP/501-0137",
"DAHDI/g0") in new stack
-- Executing [...@nineoneone:3] Set("SIP/501-0137",
"EMERGENCY=1,g") in new stack
-- Executing [...@nineoneone:4] Set("SIP/501-0137",
"SET_EMERG_FLAG=1") in new stack
-- Executing [...@nineoneone:5] Dial("SIP/501-0137",
"DAHDI/g0/91234567") in new stack
[Mar 3 11:26:06] WARNING[28572]: app_dial.c:1547 dial_exec_full:
Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel
congestion)
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/501-00000137' status is 'CONGESTION'
Regards
Shahnawaz
On Wed, Mar 3, 2010 at 10:54 AM, Steve Howes wrote:
>
> On 3 Mar 2010, at 17:21, mir shahnawaz wrote:
>> [nineoneone]
>> exten => s,1,Set(SET_EMERG_FLAG=0)
>> exten => s,n(checkavail),ChanIsAvail(${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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Re: [asterisk-users] Asterisk on MPLS VPN
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 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 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] softhangup
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] softhangup
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 24 FXS Port Voip Gateway and Asterisk
Xorcom XR005 is highly recommended. They work great. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=120550484883&ssPageName=STRK:MESELX:IT Smir On Tue, Mar 30, 2010 at 7:34 AM, Juan Miguel wrote: > Hello Joseph > > I recommend that you use The Mediatrix 4100 Series are very good. > > Juan M. > > 2010/3/28 Joseph Begumisa >> >> 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 >> >> -- >> _ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Priority based softhangup
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Priority based softhangup
Thanks Danny,
Can you please give me idea about Softhang.agi
Thanks
Smir
On Tue, Mar 30, 2010 at 1:31 PM, Danny Nicholas 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
Thanks Steve and Danny for your help.
S Mir
On Tue, Mar 30, 2010 at 3:16 PM, Steve Edwards
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] Michael Wegner
http://www.villasantilles.com/home.php -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Location with PRI / Analog lines
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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
