[asterisk-users] Digium TDM400P in Soekris net5501-70?
Hello - I've been running Asterisk (quite happily!) for several years now using a Digium TDM400P card in an old Linux box (P4 1.6 w/ 256MB RAM). I'm also running another old PC running m0n0wall as a firewall. Between these two boxes, that run 24x7, I'm drawing a lot more power than needed and hoping to make a dent in my monthly electric bill by consolidating the two into a single box with efficient power supply, low power processor, and no spinning HD platters. Main question is whether anyone knows if the Digium TDM400P should be compatible with the 3.3V PCI slot in the Soekris Net5501-70 box? Soekris' description for the net5501-70 says, in part, it has support for one or two low-power standard PCI board I see on my Digium card that it requires a molex connector supplying voltage. The Net5501 has a small 4-pin molex header on the board, I wonder if a small to regular sized molex power cable would do the job to supply this card. If the Soekris isn't expected to work well, are there any mainstream small form factor/low-power solutions for a SoHo asterisk server? -- Brian McEntire Photographer Owner B Scott Photography (240) 358-6655 studio www.bscottphoto.com ___ -- 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] Digium TDM400P in Soekris net5501-70?
Thanks Ira - I may yet still go with a standard Intel solution, but I think there could be major power savings to be had going with a smaller box like a Soekris if it can work. A good rule of thumb for 24x7 devices is $1 per watt per year, so 45 watts, while good, will still be $45 per year. I don't know what a Soekris would draw, but without a power supply fan, and using a CF card rather than a conventional HD, I'm hoping the power use would be much reduced. I will look into AstLinux. I'm actually hoping to run a VM (like VMWare) on this solution and run the firewall (m0n0wall) inside the VM. M0n0wall is a tiny distro that runs from a CD (or can run from a CF card), so I think it would still run fine inside a VM. On Mon, Jul 20, 2009 at 2:20 PM, Irai...@extrasensory.com wrote: At 10:09 AM 7/20/2009, you wrote: If the Soekris isn't expected to work well, are there any mainstream small form factor/low-power solutions for a SoHo asterisk server? I just built a box for my Asterisk system using an Intel Motherboard with an Atom 330, 5400 RPM HD, TDM 400 with 4 red cards and the cheap PS hat came with the case. Draws 43 watts according to my Kill-A-Watt and except for the TDM 400 which I already had it cost under 250 as parts from NewEgg. The only annoyance might be it has only one ethernet port with the only easy place to put another being a USB port. Ira ___ -- 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 -- Brian McEntire Photographer Owner B Scott Photography (240) 358-6655 studio www.bscottphoto.com ___ -- 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] Digium TDM400P in Soekris net5501-70?
Thanks for the reply Alex. I'm not too scared of the soldering iron (I own one, but my work with it isn't pretty ;-) But can you confirm, are you just using the small power header on the board to supply power to the pci card? I was wondering if I was going to have to snake an another wall wort into the box to power the card, would be good if I don't have to do that! Not 100% sure I could run a VM on it, but the new net5501 board comes with 512MB ram and I think a 500-ish MHz processor, way more than what I'm currently using to run m0n0wall, so even if the VM takes a bite out of it, it should be fine, hardest part might be configuring the VM to boot monowall from CF. Can you partition a CF card? (ie, one partition for the monowall firmware and the other for the stripped down linux install to run Asterisk?) On Mon, Jul 20, 2009 at 4:44 PM, Alex Samada...@samad.com.au wrote: On Mon, Jul 20, 2009 at 01:09:30PM -0400, Brian McEntire wrote: Hello - I've been running Asterisk (quite happily!) for several years now using a Digium TDM400P card in an old Linux box (P4 1.6 w/ 256MB RAM). I'm also running another old PC running m0n0wall as a firewall. Between these two boxes, that run 24x7, I'm drawing a lot more power than needed and hoping to make a dent in my monthly electric bill by consolidating the two into a single box with efficient power supply, low power processor, and no spinning HD platters. Main question is whether anyone knows if the Digium TDM400P should be compatible with the 3.3V PCI slot in the Soekris Net5501-70 box? Hi I have a the same setup you mention here, except I have a tdm410 card. I have a cf boot and a SSD card as well. Running Debian for firewall and asterisk server. Works well I have 3 vpn tunnels and a 6to4 tunnel ending on this machine, 2 fxs + 1 fxo. from my collectd graphs it looks like it really only spike when I am taking readins :) One catch the case that comes from soekris is too tight to put the molex on, I had to solder it to the connectors underneath. all fine though I am not sure about running a vm on this box though - I have some thing similiar at another site, but a bigger box. Alex Soekris' description for the net5501-70 says, in part, it has support for one or two low-power standard PCI board I see on my Digium card that it requires a molex connector supplying voltage. The Net5501 has a small 4-pin molex header on the board, I wonder if a small to regular sized molex power cable would do the job to supply this card. If the Soekris isn't expected to work well, are there any mainstream small form factor/low-power solutions for a SoHo asterisk server? -- Expense Accounts, n.: Corporate food stamps. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkpk1yMACgkQkZz88chpJ2NvmgCg3+4zJhQBcnQzxMPeQ1N+KXn1 XBMAnjtAOUjpC/++2acwVuHcYOpPQG21 =eRW6 -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 -- Brian McEntire Photographer Owner B Scott Photography (240) 358-6655 studio www.bscottphoto.com ___ -- 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] Digium TDM400P in Soekris net5501-70?
Darrick - You seem adamant, and I will look deeper into the firewall in Astlinux! :-) The one thing running monowall in a VM would do for me is (in theory) make it very simple to move my existing, working m0n0wall configuration. I've been running it for a while, it serves a bunch of DHCP clients, does a little NAT, and has 20 or so specific rules for what can talk to what across the LAN, WAN, and DMZ segments of the firewall. If Astlinux can do all that, and I can grok it easily, it might be easier than running m0n0wall inside a VM. I suppose the other thing running m0n0wall inside a VM might do is a little extra security. If the firewall is in a VM and the asterisk part is running on the hardware without access to the LAN ports (which are all owned by the VM) then it *might* make the asterisk install a little more secure or less exposed to automated attacks. Not saying this is a high payoff for me, but another potential pro for a VM setup. On Mon, Jul 20, 2009 at 7:55 PM, Darrick Hartmandhart...@djhsolutions.com wrote: I still don't see what you gain by using m0n0wall and a separate Asterisk install. I can't think of one thing that you would need a separate m0n0wall instance to do that AstLinux can't do on it's own. The web interface has become quite completely in the last few releases. Traffic shaping, firewall, vpn support etc. I don't understand how a VM does anything more than complicate an otherwise simple set up. Darrick Brian McEntire wrote: Thanks for the reply Alex. I'm not too scared of the soldering iron (I own one, but my work with it isn't pretty ;-) But can you confirm, are you just using the small power header on the board to supply power to the pci card? I was wondering if I was going to have to snake an another wall wort into the box to power the card, would be good if I don't have to do that! Not 100% sure I could run a VM on it, but the new net5501 board comes with 512MB ram and I think a 500-ish MHz processor, way more than what I'm currently using to run m0n0wall, so even if the VM takes a bite out of it, it should be fine, hardest part might be configuring the VM to boot monowall from CF. Can you partition a CF card? (ie, one partition for the monowall firmware and the other for the stripped down linux install to run Asterisk?) On Mon, Jul 20, 2009 at 4:44 PM, Alex Samada...@samad.com.au wrote: On Mon, Jul 20, 2009 at 01:09:30PM -0400, Brian McEntire wrote: Hello - I've been running Asterisk (quite happily!) for several years now using a Digium TDM400P card in an old Linux box (P4 1.6 w/ 256MB RAM). I'm also running another old PC running m0n0wall as a firewall. Between these two boxes, that run 24x7, I'm drawing a lot more power than needed and hoping to make a dent in my monthly electric bill by consolidating the two into a single box with efficient power supply, low power processor, and no spinning HD platters. Main question is whether anyone knows if the Digium TDM400P should be compatible with the 3.3V PCI slot in the Soekris Net5501-70 box? Hi I have a the same setup you mention here, except I have a tdm410 card. I have a cf boot and a SSD card as well. Running Debian for firewall and asterisk server. Works well I have 3 vpn tunnels and a 6to4 tunnel ending on this machine, 2 fxs + 1 fxo. from my collectd graphs it looks like it really only spike when I am taking readins :) One catch the case that comes from soekris is too tight to put the molex on, I had to solder it to the connectors underneath. all fine though I am not sure about running a vm on this box though - I have some thing similiar at another site, but a bigger box. Alex Soekris' description for the net5501-70 says, in part, it has support for one or two low-power standard PCI board I see on my Digium card that it requires a molex connector supplying voltage. The Net5501 has a small 4-pin molex header on the board, I wonder if a small to regular sized molex power cable would do the job to supply this card. If the Soekris isn't expected to work well, are there any mainstream small form factor/low-power solutions for a SoHo asterisk server? -- Expense Accounts, n.: Corporate food stamps. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkpk1yMACgkQkZz88chpJ2NvmgCg3+4zJhQBcnQzxMPeQ1N+KXn1 XBMAnjtAOUjpC/++2acwVuHcYOpPQG21 =eRW6 -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 -- Brian McEntire Photographer Owner B
Re: [asterisk-users] Re: Can I add distinctive ring with asterisk and TDM400?
Maybe not, but I got it working good enough and time is scarce these days so I didn't mess with it after that. On 4/19/07, Stephen Bosch [EMAIL PROTECTED] wrote: Brian McEntire wrote: A follow-up with the solution in case anyone else is looking for this answer: I created two contexts in my zapata.conf file, since each VOIP line is terminated by a VOIP adapter and then just comes in hardwired to the TDM400 via RJ11 line, I know which VOIP number is connected to which Wildcard port. In Zapata.conf: usedistinctiveringdetection=yes Is this line even necessary? You're sending distinctive ring, not receiving it. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Can I add distinctive ring with asterisk and TDM400?
A follow-up with the solution in case anyone else is looking for this answer:
I created two contexts in my zapata.conf file, since each VOIP line is
terminated by a VOIP adapter and then just comes in hardwired to the
TDM400 via RJ11 line, I know which VOIP number is connected to which
Wildcard port.
In Zapata.conf:
usedistinctiveringdetection=yes
; ...
signalling=fxs_ks
context=incoming-voipsunr
rxgain=0
txgain=2.5
channel = 3
signalling=fxs_ks
context=incoming-voipvp
rxgain=0
txgain=2.5
channel = 4
- - -
Now in extensions.conf, I can ring the phones with a normal ring (DIAL
with no ring parameter) when a call comes in the voip-sunr context. Or
I can ring with a distinctive ring when a call comes in the voip-vp
context:
[incoming-voipsunr]
exten = s,1,NoOp
exten = s,n,Dial(${ALLPHONES})
exten = s,n,Hangup
[incoming-voipvp]
exten = s,1,NoOp
exten = s,n,Dial(${ALLPHONES}r2)
exten = s,n,Hangup
The r2 in the second context causes a distinctive ring. ALLPHONES is
defined above in my extensions.conf as ALLPHONES=ZAP/1ZAP/2
Hope that's useful to someone else. It works for me.
On 4/17/07, Brian McEntire [EMAIL PROTECTED] wrote:
Hello -
I have a TDM400P with 2 FXO and 2 FXS modules.
Feeding the FXS modules are two VOIP lines which are terminated by
VOIP adapters and have regular RJ11 wires connecting to the FXS ports.
Since the two different VOIP lines have different phone numbers, and I
know and can tell asterisk which VOIP line is connected to which FXS
port, can I cause a distinctive ring on the extensions if a call comes
in the 2nd VOIP line?
ie, I'd like to ring all extensions when a call comes in either VOIP
line, but I'd like to give a distinctive ring in the case the call
came in the 2nd VOIP line.
Here is what I have in extensions.conf right now, which works well,
except doesn't include any support for distinctive ring:
[globals]
ALLPHONES=Zap/1Zap/2
;
;other stuff
[default]
;
; other stuff
exten = s,1,NoOp
exten = s,n,Dial(${ALLPHONES})
exten = s,n,Hangup
I'm not sure if this is something that could/should be implemented in
extensions.conf or zapata.conf
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[asterisk-users] Can I add distinctive ring with asterisk and TDM400?
Hello -
I have a TDM400P with 2 FXO and 2 FXS modules.
Feeding the FXS modules are two VOIP lines which are terminated by
VOIP adapters and have regular RJ11 wires connecting to the FXS ports.
Since the two different VOIP lines have different phone numbers, and I
know and can tell asterisk which VOIP line is connected to which FXS
port, can I cause a distinctive ring on the extensions if a call comes
in the 2nd VOIP line?
ie, I'd like to ring all extensions when a call comes in either VOIP
line, but I'd like to give a distinctive ring in the case the call
came in the 2nd VOIP line.
Here is what I have in extensions.conf right now, which works well,
except doesn't include any support for distinctive ring:
[globals]
ALLPHONES=Zap/1Zap/2
;
;other stuff
[default]
;
; other stuff
exten = s,1,NoOp
exten = s,n,Dial(${ALLPHONES})
exten = s,n,Hangup
I'm not sure if this is something that could/should be implemented in
extensions.conf or zapata.conf
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[asterisk-users] Re: Can I add distinctive ring with asterisk and TDM400?
Thinking I might be able to do this with a context in zapata.conf and
possibly extensions.conf but I'd need some tips.
But just realized I can turn on distinctive ringing for one of the
lines through my VOIP provider. As long as asterisk will pass this
through, it solves my question.
On 4/17/07, Brian McEntire [EMAIL PROTECTED] wrote:
Hello -
I have a TDM400P with 2 FXO and 2 FXS modules.
Feeding the FXS modules are two VOIP lines which are terminated by
VOIP adapters and have regular RJ11 wires connecting to the FXS ports.
Since the two different VOIP lines have different phone numbers, and I
know and can tell asterisk which VOIP line is connected to which FXS
port, can I cause a distinctive ring on the extensions if a call comes
in the 2nd VOIP line?
ie, I'd like to ring all extensions when a call comes in either VOIP
line, but I'd like to give a distinctive ring in the case the call
came in the 2nd VOIP line.
Here is what I have in extensions.conf right now, which works well,
except doesn't include any support for distinctive ring:
[globals]
ALLPHONES=Zap/1Zap/2
;
;other stuff
[default]
;
; other stuff
exten = s,1,NoOp
exten = s,n,Dial(${ALLPHONES})
exten = s,n,Hangup
I'm not sure if this is something that could/should be implemented in
extensions.conf or zapata.conf
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Re: [asterisk-users] Adding DND to dialplan
Don't think that was it unless I still have a typo. Here's my line
from extensions.conf:
exten = _#78,n,Set(DB(DND/${CALLERID(num)})=1)
in the CLI issued 'reload' after saving the updated extensions.conf
and then picked up the phone and dialed #78. Still getting this error:
[Apr 4 08:56:20] WARNING[6866]: func_db.c:94 function_db_write: DB
requires an argument, DB(family/key)=value
Should DB support be built in by default?
Is there a DB schema I need to consult to make sure I have the right
family key pair?
I'm using a Zaptel driver so maybe the CALLERID(num) var is not set or
is confusing the Set line. I think the zap comes up Zap/2.
On 4/4/07, Bruce Reeves [EMAIL PROTECTED] wrote:
You have a syntax error.
exten = _#78,n,Set(DB(${DND/CALLERID
(num)})=1)
should read
exten = _#78,n,Set(DB(DND/ ${CALLERID
(num)})=1)
On 4/3/07, Brian McEntire [EMAIL PROTECTED] wrote:
Hmmm...
Had hoped this would be easy, maybe still is, but running into a problem:
When I dial #78, I get a fast busy and these errors on the CLI
[Apr 4 00:39:29] ERROR[4046]: pbx.c:1523 ast_func_read: Function
DND/CALLERID not registered
[Apr 4 00:39:29] WARNING[4046]: func_db.c:87 function_db_write: DB
requires an argument, DB(family/key)=value
- - -
Here is the extensions.conf entry:
[dnd-on]
exten = _#78,1,Answer()
exten = _#78,n,Wait(1)
exten = _#78,n,Set(DB(${DND/CALLERID(num)})=1)
exten = _#78,n,Playback(do-not-disturb)
exten = _#78,n,Playback(enabled)
exten = _#78,n,Hangup()
- - -
It appears to me that Set(DB ... as a function isn't working, isn't
built in, or needs more information.
I saw something about GLOBAL variables, perhaps I can use those instead?
On 4/3/07, Doug Lytle [EMAIL PROTECTED] wrote:
Brian McEntire wrote:
Hello -
I've read Asterisk should be able to activate a do not disturb feature
Instead of using 2 extensions, you can get away with just one. Check
the database entry at the start, if it's already set, remove it. If
it's not there, add it.
[dnd]
; **
; Do not disturb can be set via Asterisk
; instead of the phones by dialing this
; number.
; **
exten = 79*,1,Set(CALLBACK=${DB(DND/${CALLERIDNUM})})
exten = 79*,2,GotoIf($[${CALLBACK} = YES]?79*,3:79*,101)
exten = 79*,3,Set(DB(DND/${CALLERIDNUM})=NO)
exten = 79*,4,Playback(local/stutter)
exten = 79*,5,Playback(de-activated)
exten = 79*,6,Hangup()
exten = 79*,101,Set(DB(DND/${CALLERIDNUM})=YES)
exten = 79*,102,Playback(local/stutter)
exten = 79*,103,Playback(activated)
exten = 79*,104,Hangup()
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
Temporary Safety, deserve neither Liberty nor Safety.
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--
Bruce Reeves
Nortex Networks
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Re: [asterisk-users] Adding DND to dialplan
Ah!
Got it. Hard coding CallerID is a good idea and thank you for the example.
I decided to try the Noop(DB(...)) to see what was getting passed and
the empty CALLERID was the issue.
I decided to skip that and implement a global DND since that's what I
wanted anyway so I just set DND/ALL=1 in the DB line.
I'll post a full example here when I put on the finishing touches but
it is working now. Thanks all for the help.
One question... are there any places to get extra sound files like
activated or deactivated or do not disturb is... ?? I didn't
find them in the sounds directory after a vanilla install of the
latest stable asterisk 1.4.
On 4/4/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Wed, 4 Apr 2007, Brian McEntire wrote:
Don't think that was it unless I still have a typo. Here's my line
from extensions.conf:
exten = _#78,n,Set(DB(DND/${CALLERID(num)})=1)
in the CLI issued 'reload' after saving the updated extensions.conf
and then picked up the phone and dialed #78. Still getting this error:
[Apr 4 08:56:20] WARNING[6866]: func_db.c:94 function_db_write: DB
requires an argument, DB(family/key)=value
Should DB support be built in by default?
It seems that it's already there (else you'd not get an error from
func_db)
Is there a DB schema I need to consult to make sure I have the right
family key pair?
I do the same with:
exten = *49,n,Set(DB(${CALLERID(num)}/doNotDisturb)=1)
So I keep them the other way round.
Why not put in something like:
exten = _#78,n,Noop(DB(DND/${CALLERID(num)})=1)
and just see what it says? Maybe your callerId is somehow not being set
correctly?
I'm using a Zaptel driver so maybe the CALLERID(num) var is not set or
is confusing the Set line. I think the zap comes up Zap/2.
I'd definately check the caller-id - if it's a local phone on an FXS port,
then you might even want to hard-wire the caller-id in the
/etc/asterisk/zapata.conf file. eg.
; Channel 1: Local analogue line
context=internal
group=0
signalling=fxo_ks
sendcalleridafter=2
rxgain=3
txgain=3
mailbox=100
callerid=Shared DECT 100
channel = 1
Gordon
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[asterisk-users] Adding DND to dialplan
Hello -
I've read Asterisk should be able to activate a do not disturb feature
to turn off the ringers on extensions. I checked the wiki and can't
find documentation for how to do it.
Here's my attempt, added to extensions.conf:
[dnd-on]
exten = _#78,1,Answer
exten = _#78,n,Wait(1)
exten = _#78,n,Macro(user-callerid,)
exten = _#78,n,Set(DB(DND/${CALLERID(number)})=YES)
exten = _#78,n,Playback(do-not-disturbactivated)
exten = _#78,n,Macro(hangupcall,)
[dnd-off]
exten = _#79,1,Answer
exten = _#79,n,Wait(1)
exten = _#79,n,Macro(user-callerid,)
exten = _#79,n,dbDel(DND/${CALLERID(number)})
exten = _#79,n,Playback(do-not-disturbde-activated)
exten = _#79,n,Macro(hangupcall,)
;further down
include = dnd-on
include = dnd-off
- - -
Monitoring asterisk from the CLI, when I dial #78 on an extension, I
just get a fast busy signal and this information is reported on the
CLI:
Apr 3 10:41:33 WARNING[30702]: app_macro.c:144 macro_exec: No such
context 'macro-user-callerid' for macro 'user-callerid'
Apr 3 10:41:33 WARNING[30702]: func_db.c:97 function_db_write: DB
requires an argument, DB(family/key)=value
Apr 3 10:41:33 WARNING[30702]: file.c:504 ast_openstream_full: File
do-not-disturb does not exist in any format
Apr 3 10:41:33 WARNING[30702]: file.c:816 ast_streamfile: Unable to
open do-not-disturb (format unknown): No such file or directory
Apr 3 10:41:33 WARNING[30702]: app_playback.c:106 playback_exec:
ast_streamfile failed on Zap/2-1 for do-not-disturbactivated
Apr 3 10:41:33 WARNING[30702]: file.c:504 ast_openstream_full: File
activated does not exist in any format
Apr 3 10:41:33 WARNING[30702]: file.c:816 ast_streamfile: Unable to
open activated (format unknown): No such file or directory
Apr 3 10:41:33 WARNING[30702]: app_playback.c:106 playback_exec:
ast_streamfile failed on Zap/2-1 for do-not-disturbactivated
Apr 3 10:41:33 WARNING[30702]: app_macro.c:144 macro_exec: No such
context 'macro-hangupcall' for macro 'hangupcall'
- - -
Any tips?
All I really want to do is turn off the ringers / do not ring
extenstions when I've activated DND. Right now I'm just using a hack
which is to shutdown asterisk altogether when I don't want the phones
to ring, which of course also prevents dialing out, it's a
sledgehammer approach and I'm looking for something more typical.
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Re: [asterisk-users] Adding DND to dialplan
Ahh... got it now. Thanks for all the replies.
I was thinking that it was a function that was already built in, but
I see by setting a value and then testing it before ringing
extensions, it's easily added to the dialplan.
On 4/3/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
Bruce Reeves wrote:
exten = *73,1,Answer()
exten = *73,n,Wait(0.5)
exten = *73,n,Set(DB(${CALLERID(number)}/DND)=1)
Would prefer Set(DB(${DND/CALLERID(num)})=1)
exten = *73,n,Playback(do-not-disturb)
exten = *73,n,Playback(enabled)
exten = *73,n,Hangup()
and then
When someone calls say extension 1000 I would have a macro check for :
exten = s,n,Set(DNDStatus=$[${DB(1000/DND)} = 1]) = returns a 1 if
enabled or a 0
exten = s,n,GoToIf($[${DNDStatus} = 1]?DND)
exten = s,n(DND),Voicemail([EMAIL PROTECTED],u)
More complete:
[macro-check-dnd]
exten = s,n,Answer()
exten = s,n,Wait(1)
exten = s,n,Set(DNDStatus=$[${DB(DND/${ARG1})} = 1])
exten = s,n,GotoIf($[${DNDStatus} = 1]?DND)
exten = s,n,Dial(SIP/${ARG1})
exten = s,n,Hangup()
exten = s,n(DND),Voicemail([EMAIL PROTECTED],u)
exten = s,n,Hangup()
[default]
exten = _,1,Macro(check-dnd,${EXTEN})
Regards,
Philipp
--
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Re: [asterisk-users] Adding DND to dialplan
Hmmm...
Had hoped this would be easy, maybe still is, but running into a problem:
When I dial #78, I get a fast busy and these errors on the CLI
[Apr 4 00:39:29] ERROR[4046]: pbx.c:1523 ast_func_read: Function
DND/CALLERID not registered
[Apr 4 00:39:29] WARNING[4046]: func_db.c:87 function_db_write: DB
requires an argument, DB(family/key)=value
- - -
Here is the extensions.conf entry:
[dnd-on]
exten = _#78,1,Answer()
exten = _#78,n,Wait(1)
exten = _#78,n,Set(DB(${DND/CALLERID(num)})=1)
exten = _#78,n,Playback(do-not-disturb)
exten = _#78,n,Playback(enabled)
exten = _#78,n,Hangup()
- - -
It appears to me that Set(DB ... as a function isn't working, isn't
built in, or needs more information.
I saw something about GLOBAL variables, perhaps I can use those instead?
On 4/3/07, Doug Lytle [EMAIL PROTECTED] wrote:
Brian McEntire wrote:
Hello -
I've read Asterisk should be able to activate a do not disturb feature
Instead of using 2 extensions, you can get away with just one. Check
the database entry at the start, if it's already set, remove it. If
it's not there, add it.
[dnd]
; **
; Do not disturb can be set via Asterisk
; instead of the phones by dialing this
; number.
; **
exten = 79*,1,Set(CALLBACK=${DB(DND/${CALLERIDNUM})})
exten = 79*,2,GotoIf($[${CALLBACK} = YES]?79*,3:79*,101)
exten = 79*,3,Set(DB(DND/${CALLERIDNUM})=NO)
exten = 79*,4,Playback(local/stutter)
exten = 79*,5,Playback(de-activated)
exten = 79*,6,Hangup()
exten = 79*,101,Set(DB(DND/${CALLERIDNUM})=YES)
exten = 79*,102,Playback(local/stutter)
exten = 79*,103,Playback(activated)
exten = 79*,104,Hangup()
Doug
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[asterisk-users] Asterisk with Dialplan or TrixBox for this case?
Hi all - Been using Asterisk installed on Debian and love it. But it's time to rearrange some lines and looking for a few features I didn't enable or have in the dial plan the first time around and wondering if you would recommend doing it through configs again or if one of the prepackaged solutions would more easily support these needs. One that caught my eye was TrixBox but I'd be open to other suggestions. I have a Wildcat TDM400 (IIRC) with 2 FXS and 2 FXO ports. Currently I'm terminating a POTS line and a VoicePulse VOIP line (via the supplied adapter) into the FXS ports (forgive me if I confused the FXO/FXS it gets me every time.) I have the dialplan set up to ring all extensions when either incoming line rings. Ring available extensions if one is in use. For dial out, it only dials out the VOIP line unless I override by dialing 9 first (because we pay per call on the POTS line so I want to know I'm doing it rather than have asterisk do it for me if the VOIP line is already in use.) - - - What I'm looking to do is keep the functionality above but drop the POTS line and add a SunRocket line also terminated with a VOIP adapter just like the VoicePulse line. Although the net connection will be a single point of failure, at least I'll have two different VOIP providers for some redundancy. I'd like to: - ring all extensions when a call comes in either VOIP line. - distinctive ring for calls coming in the SunRocket line (which Asterisk will know by the port that the line comes in on.) - do not disturb functionality to disable all extensions from ringing by dialing a *XX number from any phone in the house. Ability to toggle ringing back on easily. - dial out any available line (now that both are VOIP) Easy to do with TrixBox or better off installing the latest Asterisk and doing it through the command line and configuration file interface? Thanks! PS - Oddly, the SunRocket VOIP adapter doesn't seem to give a dialtone but a regular old phone works fine when connected to it. Will this cause problems for Asterisk? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Virtual VOIP numbers going to separate Asterisk mailboxes?
I use Voicepulse as a VIOP provider, the line comes in via a Sipura 3002 box. That's connected to the Asterisk box via a TDM422B POTS card. I'd like to add a virtual phone number to my VOIP service so that I can direct calls to a home business to a different voice mail than calls to the home phone number. Is there a way for Asterisk to know which virtual number was dialed under this configuration? Thanks for any ideas or pointers! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Do Not Disturb?
I looked on the voip-info wiki and found sparse and conflicting information on how to do this with Asterisk... My incoming lines are all on Zaptel. Is there a simple why to implement a '*363 (do not disturb) toggle via the dialplan? It would be nice to be able to pick up an extension, dial *363, and have all calls sent to voicemail without ringing the extensions. Doing it again would disable the feature. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Answer call waiting / flash with Zaptel POTS and VOIP
Hello, hoping someone out there has some ideas - I have a VOIP line that has call waiting. It is terminated at a Sipura 3000 and the POTS side of that device connects to an FXO port in my * box. I also have a POTS/PSTN line that terminates in another FXO port on my * box. There are two FXS ports which feed cordless phones. I'm using the Zaptel TDM400 card. This gives 2 extensions + 2 lines in/out and the VOIP line has call waiting. This is the problem: Asterisk (or Zaptel) only interprets the flash from a handset to mean switch between FXO cards. Or at least I think that's what's happening. If I'm on an extension and using the VOIP line, and a call comes in on the POTS line, I get the audible beep and I can answer it by pressing flash on the extension. However, if I'm on the VOIP line and another call is placed to the VOIP number, I hear the beep, but pressing flash doesn't answer it. Instead it gives me a dial tone. Pressing flash again gets me back to the original VOIP call but there is no way to answer the call waiting on the VOIP line. What I'm looking for is a way to tell * not to interpret the flash and instead pass it out the line. I can always answer non-call-waiting incomming calls on the other extension. I'd like to be able to use flash for signaling the VOIP (upstream?) to switch to the callwaiting call and then back as needed. I tried setting callwaiting=no in zapata.conf and restarting * but that didn't have the desired effect. I think it prevented me from hearing an audible beep when one line and extension were in use and a call came in the other line... that's okay. But it didn't help answer the callwaiting call on the voip line. I need a way to tell * not to interpret the flash, and instead, pass it out the line connected to this extension. Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Cell phone dock/switch as Asterisk FXO source
No Bluetooth in the Samsung T309. I couldn't think of why I'd want BT... then of course I started looking at cell sockets, etc. after I got it and found several do not have a cable for the T309 yet. In hindsight, bluetooth would have made this easier. Live and learn! On 1/6/06, Jonathan Attwood [EMAIL PROTECTED] wrote: On 1/5/06, Brian McEntire [EMAIL PROTECTED] wrote: Wow! Thanks for all the responses! Very informative. Erik: I'm just looking for simple dial-out and pass-along incoming cell calls to *. Looks like the doc-n-talk should do it, except I checked with them and, silly me, the new Samsung t309 phone I just got is not supported yet. Hopefully it will be in a few months. Is it not supported, even with the Bluetooth module? (That's assumingthe phone's BTth)___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Cell phone dock/switch as Asterisk FXO source
Wow! Thanks for all the responses! Very informative. Erik: I'm just looking for simple dial-out and pass-along incoming cell calls to *. Looks like the doc-n-talk should do it, except I checked with them and, silly me, the new Samsung t309 phone I just got is not supported yet. Hopefully it will be in a few months. I'll check the rest of the links you and others provided in this thread. Thanks!!On 1/3/06, GeekSpeed [EMAIL PROTECTED] wrote: Has anyone checked out the UNIDEN ELBT-595(http://www.uniden.com/elbt/index.html) It supposedly is a handset that can provide the same services. I have not seen any info about * compatibility though. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Message Waiting Indicator (MWI) for remote voice mail?
I haven't received any responses. Just wanted to follow up and see if anyone has ideas? It seems like there ought to be a way to do this, especially since the TDM400 FXS card is able to send the proper signal to the connected phone. It seems like there just needs to be a way to configure the FXO card to pass though or bridge that signal/information to the FXS card when it is received at the FXO card. VOIP VM - Sipura - Phone worked. Asterisk VM - FXS - Phone works. Just need a way to do: VOIP VM - Sipura - FXO - ??? - FXS - Phone The above works great for everything else I've tried so far except for passing through the message waiting indicator. Thanks for any ideas! On 9/24/05, Brian McEntire [EMAIL PROTECTED] wrote: I have Asterisk voice mail setup locally. It works great, I'm impressed! Some details about my system: I'm using a TDM22B card to interface with both the PSTN and a VOIP provider. I'm running 1.2-beta from CVS. I have a regular VTech phone plugged into one of the FXS ports. Asterisk is able to indicate when a local voicemail message is waiting via the LCD display of my analog phone. It also gives a broken dial tone. This is achieved by specifying mailbox=mb# in zapata.conf and possibly also by specifing adsi=yes in the same file. The question I have is this: I also have voicemail with my VOIP provider. Before jumping into Asterisk, the VOIP provider could send the message waiting indicator to my phone when I had new messages. After putting Asterisk between my analog handset and the VOIP adapter, the message waiting indicator from the VOIP provider seems to no longer get through to the phone. The connection to the VOIP provider is Cable Modem - Sipura 3002 - TDM FXO interface - TDM FXS interface - phone. Is there a way for Asterisk to get notified and pass the message waiting indicator on to my handset when there is a voice mail waiting at the VOIP provider? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Voip provider
VoicePulse! (www.voicepulse.com) I've been extremely pleased with the quality of the VOIP calls, uptime, and on the rare occasions I've needed support, they responded within 1 day and followed through great on the tickets. Rates are exceptional too, national calling 200 minutes is $15/mo and unlimited local calls which don't count against the minutes is a huge calling area including your area code and all adjacent area codes usually. They have a nice web interface to manage features such as filters and scheduled do not disturb. When I was comparing other providers, VoicePulse seemed to have a super-set of all the features provided by the others and generally the best price. Oh, and I think they may use Asterisk for some services... for certain they've contributed sounds/voice prompts to the community. I'm happy to subscribe to them. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VM low volume - testers needed
Hmmm...
I checked out CVS-HEAD, built and installed it this morning. Most
testing was going well, but then I found out the behavior of
ChanIsAvail has changed (is broken?)
In my Dial Plan, if a call comes in on the PSTN line, and is not
answered by the extension (or if the extension is busy), ChanIsAvail
checks to see of the outgoing VOIP line is available. If so, it
forwards the call to the VOIP voice mail. If not, it forwards the call
to the Asterisk Voicemail.
With 1.2-beta, ChanIsAvail works for me. With CVS-HEAD, it hangs up on the caller.
Here is the relevant portion of my extensions.conf:
exten = s,7,Dial(${PHONE1},15)
exten = s,8,Goto(108)
exten = s,108,ChanIsAvail(${VOIP1})
exten = s,109,Dial(${VOIP1}/${VOIPNUM})
exten = s,209,VoiceMail(123|sbg(6))
In the globals section, VOIP1 is set equal to Zap/4
With 1.2-beta, -vvv logs show this, which is successful:
-- Executing ChanIsAvail(Zap/3-1, Zap/4) in new stack
-- Executing VoiceMail(Zap/3-1, 123|sbg(6)) in new stack
-- Playing '/var/spool/asterisk/voicemail/default/123/busy' (language 'en')
With CVS-HEAD -vvv logs show this, which is unsuccessful:
-- Executing ChanIsAvail(Zap/3-1, Zap/4) in new stack
== Spawn extension (incoming-pstn, s, 208) exited non-zero on 'Zap/3-1'
-- Hungup 'Zap/3-1'
Is there another list or someone I should mention this to? Asterisk should not hangup Zap/3-1 at this point.
On 9/24/05, Rich Adamson [EMAIL PROTECTED] wrote:
The patch is in cvs-head, which has been very stable for me. :) Hi Richard, I am experiencing the same problem. I'd like to test your patch. Thing, is, I don't know which
CVS it's in:) ... I checked out 1.2-beta on Tuesday (9/21) and compiled it. When I type 'show applicationvoicemail', it does not describe the g(#) option, so I think my version must not have it.
I am using a TDM22B card and voicemails seem very quiet if they are left from in incoming POTSconnection. When I enter voicemail by direct dialing a local extension and leave a message from the advanced options
menu, the recorded message is much louder. I should qualify, not only are my VMs coming in over POTS, I am actually calling out firstthrough the TDM22B, to Sipura, to VOIP provider, back in via PSTN, to TDM22B, to VM. I'm amazed it works at all:)... I'm very
impressed by Asterisk and especially it's voicemail. I would like to resolve the low volume issue though. If you can tell me which CVS to check out, I can try it. I'd like to stick to the 1.2-beta
branch though because I don't want to rework all my config files. On 9/21/05, Rich Adamson [EMAIL PROTECTED] wrote: On Monday 19 September 2005 12:38, Rich Adamson wrote:
The g(6) adds a 6 db gain for zap calls that end up recording a Voicemail message. ...* 'g(#)' the specified amount of gain will be requested during message
recording
(units are whole-number decibels (dB)) How in the hell does that make any sense?are your normal incoming calls quiet too or just voicemail? Yes, see bug 2022 and 2023 for details, as well as
http://www.routers.com/asteriskprob/asterisk-config.htm for a very detailed analysis of the problem. I believe one of the more serious issues amounts to: if asterisk is
located a fair distance from the central office (-7db in my case), setting the rxgain and/or txgain to any level that would be considered reasonable for that loss (eg, rxgain=5, txgain=5), hugh amounts of echo result that
cannot be addressed through zapata.conf echo entris, and changing compile options to agressive, etc, does not help. Its my believe (from working with several TDM users), the further one is from the CO,
the bigger the problem. (Or, short pstn cable lengths less then about 4 or 5db can almost always be addressed via parameters.) The above workaround is very usable (assuming it works) when someone
calls in via the pstn and leaves a voicemail (which is already at least 7db down plus their own pstn loss), and then I call in via the pstn to retrive the voicemail (now 14db down PLUS the original callers
pstn loss), the audio is so faint its difficult to impossible to listen to. In my case, the asterisk box is located about 7db from the central office. As noted in bug 2023 (and 2022), calls from an outside pstn
line coming into asterisk incure a 7db pstn loss (which can't be adjusted for with rxgain and txgain as changing those values to something reasonable generates echo).Retrieving that VM message from an outside
location creates another 7db loss (now -14db down in total), making it very difficult (if not impossible) to hear the message. (And, yes I've gone through all the recommendations with wav vs gsm files, etc.)
I am not sure I understand why the txgain/rxgain isn't fixing it without adding unacceptable echo...this all seems very odd...I mean for a test you should be able to dial an echo() application and have extremely quiet
echoed audio... is this the case? As an ex-telco transmission engineer,
Re: [Asterisk-Users] VM low volume - testers needed
Hmm. Thanks for the heads up, but I'm not sure that's it.
It's jumping to 208 rather than 209, so it looks more like an off-by-one error.
I tried changing to priorityjumping=yes in
/etc/asterisk/extensions.conf and reinstalled the CVS-HEAD version, but
it still jumps to 208 whereas it used to jump to 209.
On 9/24/05, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
Under 1.2 the +101 jumping is not enabled by default. There is avariable returned showing the status of the application. You need to adda j flag or put priorityjumping=yes in extensions.confJulian.
Brian McEntire wrote: Hmmm... I checked out CVS-HEAD, built and installed it this morning. Most testing was going well, but then I found out the behavior of ChanIsAvail has changed
(is broken?) In my Dial Plan, if a call comes in on the PSTN line, and is not answered by the extension (or if the extension is busy), ChanIsAvail checks to see of the outgoing VOIP line is available. If so, it forwards the call to the VOIP
voice mail. If not, it forwards the call to the Asterisk Voicemail. With 1.2-beta, ChanIsAvail works for me. With CVS-HEAD, it hangs up on the caller. Here is the relevant portion of my
extensions.conf: exten = s,7,Dial(${PHONE1},15) exten = s,8,Goto(108) exten = s,108,ChanIsAvail(${VOIP1}) exten = s,109,Dial(${VOIP1}/${VOIPNUM}) exten = s,209,VoiceMail(123|sbg(6))
In the globals section, VOIP1 is set equal to Zap/4 With 1.2-beta, -vvv logs show this, which is successful: -- Executing ChanIsAvail(Zap/3-1, Zap/4) in new stack
-- Executing VoiceMail(Zap/3-1, 123|sbg(6)) in new stack -- Playing '/var/spool/asterisk/voicemail/default/123/busy' (language 'en') With CVS-HEAD -vvv logs show this, which is unsuccessful:
-- Executing ChanIsAvail(Zap/3-1, Zap/4) in new stack == Spawn extension (incoming-pstn, s, 208) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1'
Is there another list or someone I should mention this to? Asterisk should not hangup Zap/3-1 at this point. On 9/24/05, Rich Adamson
[EMAIL PROTECTED] wrote:The patch is in cvs-head, which has been very stable for me. :)Hi Richard,
I am experiencing the same problem. I'd like to test your patch. Thing,is, I don't know whichCVS it's in :)... I checked out 1.2-beta on Tuesday (9/21) and compiled it. When I
type 'show applicationvoicemail', it does not describe theg(#) option, so I think my version must not have it.I am using a TDM22B card and voicemails seem very quiet if they are left
from in incoming POTSconnection. When I entervoicemail by direct dialing a local extension and leave a message fromthe advanced options
menu, the recorded message is muchlouder.I should qualify, not only are my VMs coming in over POTS, I am actuallycalling out first
through the TDM22B, to Sipura, toVOIP provider, back in via PSTN, to TDM22B, to VM. I'm amazed it worksat all :) ... I'm veryimpressed by Asterisk and
especially it's voicemail. I would like to resolve the low volume issuethough.If you can tell me which CVS to check out, I can try it. I'd like to
stick to the 1.2-betabranch though because I don't want torework all my config files.On 9/21/05, Rich Adamson
[EMAIL PROTECTED] wrote:On Monday 19 September 2005 12:38, Rich Adamson wrote:The g(6) adds a 6 db gain for zap calls that end up recording a
Voicemailmessage* 'g(#)' the specified amount of gain will be requested during
messagerecording (units are whole-number decibels (dB))How in the hell does that make any sense? are your normal incoming
callsquiet too or just voicemail?Yes, see bug 2022 and 2023 for details, as well as
http://www.routers.com/asteriskprob/asterisk-config.htmfor a very detailed analysis of the problem.I believe one of the more serious issues amounts to: if asterisk is
located a fair distance from the central office (-7db in my case),settingthe rxgain and/or txgain to any level that would be consideredreasonable
for that loss (eg, rxgain=5, txgain=5), hugh amounts of echo result thatcannot be addressed through zapata.conf echo entris, and changingcompile options to agressive, etc, does not help. Its my believe
(from working with several TDM users), the further one is from the CO,the bigger the problem. (Or, short pstn cable lengths less then about4 or 5db can almost always be addressed via parameters.)
The above workaround is very usable (assuming it works) when someonecalls in via the pstn and leaves a voicemail (which is already atleast 7db down plus their own pstn loss), and then I call in via the
pstn to retrive the voicemail (now 14db down PLUS the original callerspstn loss), the audio is so faint its difficult to impossible tolisten to.
In my case, the asterisk box is located about 7db from the centraloffice. As noted in bug 2023 (and 2022), calls from an outside pstnline coming into asterisk incure a 7db pstn loss (which can't be
adjustedfor with rxgain and txgain as changing those values to somethingreasonable generates echo). Retrieving that VM message from an
outsidelocation
Re: [Asterisk-Users] VM low volume - testers needed
Oops, I didn't cc the list. Julian suggested I should try the older version of app_chanisavail.c and that worked out well. I can now use the g(#) switch and that works very well. On 9/24/05, Brian McEntire [EMAIL PROTECTED] wrote: That fixes it! Thanks. So I can run CVS HEAD but I need to check out -r 1.17 asterisk/apps/app_chanisavail.c to revert just that file to the old version. I guess it could still be a prob with the new app_chanisavail.c but it also looks like whatever provides ast_goto_if_exists could be at fault. - - - To Rich: The new gain g(#) switch works great! I have to bump mine up to g(12) which seems rediculously high, but then again I'm going out voip and back in PSTN and perhaps the VOIP is quieting the signal too. Anway, with g(12), voicemail messages are recorded at a very acceptable volume and sound good too. Thanks! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Double cpu
Someone else may be able to get more specific. The term you might be looking for is processor affinity. I know with older versions of Informix, you could bind the informix process to a specific processor, so in your case you would just bind it to Processor 1 and let asterisk run on processor 0 by default. I don't know if MySQL supports this though.On 9/23/05, Anders Svensson [EMAIL PROTECTED] wrote: Hi! Probably another newbie question. Is it possible to run * on one processor and MySql on the other in a double cpu server? Anders ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Continue dialtone after pressing 9
Hello, Sorry, I know I read this somewhere but now I can't find it when I need it. I'd like to force a call to go out one line if we dial '9' first and then the number. Same for '8' only I will force it out a different line. There is a parameter or a method to allow the dialtone to come back after pressing the first 9... but I can't remember how to do it. Anyone know? Thanks, Brian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Continue dialtone after pressing 9
Thank you! Added that to the [default] section of my extentions.conf and it works as desired.On 9/23/05, Jesse Keating [EMAIL PROTECTED] wrote:On Fri, 2005-09-23 at 14:28 -0400, Brian McEntire wrote: I'd like to force a call to go out one line if we dial '9' first and then the number. Same for '8' only I will force it out a different line. There is a parameter or a method to allow the dialtone to come back after pressing the first 9... but I can't remember how to do it.ignorepat = 9ignorepat = 8Also, your phone digit map may need to be tweaked to allow for this aswell. --Jesse KeatingGameHouse -- Systems Engineer___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VM low volume - testers needed
Hi Richard, I am experiencing the same problem. I'd like to test your patch. Thing, is, I don't know which CVS it's in :) ... I checked out 1.2-beta on Tuesday (9/21) and compiled it. When I type 'show application voicemail', it does not describe the g(#) option, so I think my version must not have it. I am using a TDM22B card and voicemails seem very quiet if they are left from in incoming POTS connection. When I enter voicemail by direct dialing a local extension and leave a message from the advanced options menu, the recorded message is much louder. I should qualify, not only are my VMs coming in over POTS, I am actually calling out first through the TDM22B, to Sipura, to VOIP provider, back in via PSTN, to TDM22B, to VM. I'm amazed it works at all :) ... I'm very impressed by Asterisk and especially it's voicemail. I would like to resolve the low volume issue though. If you can tell me which CVS to check out, I can try it. I'd like to stick to the 1.2-beta branch though because I don't want to rework all my config files. On 9/21/05, Rich Adamson [EMAIL PROTECTED] wrote: On Monday 19 September 2005 12:38, Rich Adamson wrote: The g(6) adds a 6 db gain for zap calls that end up recording a Voicemail message. ... * 'g(#)' the specified amount of gain will be requested during message recording (units are whole-number decibels (dB)) How in the hell does that make any sense?are your normal incoming calls quiet too or just voicemail?Yes, see bug 2022 and 2023 for details, as well as http://www.routers.com/asteriskprob/asterisk-config.htmfor a very detailed analysis of the problem.I believe one of the more serious issues amounts to: if asterisk is located a fair distance from the central office (-7db in my case), settingthe rxgain and/or txgain to any level that would be considered reasonablefor that loss (eg, rxgain=5, txgain=5), hugh amounts of echo result that cannot be addressed through zapata.conf echo entris, and changingcompile options to agressive, etc, does not help. Its my believe(from working with several TDM users), the further one is from the CO,the bigger the problem. (Or, short pstn cable lengths less then about 4 or 5db can almost always be addressed via parameters.)The above workaround is very usable (assuming it works) when someonecalls in via the pstn and leaves a voicemail (which is already atleast 7db down plus their own pstn loss), and then I call in via the pstn to retrive the voicemail (now 14db down PLUS the original callerspstn loss), the audio is so faint its difficult to impossible tolisten to. In my case, the asterisk box is located about 7db from the central office. As noted in bug 2023 (and 2022), calls from an outside pstn line coming into asterisk incure a 7db pstn loss (which can't be adjusted for with rxgain and txgain as changing those values to something reasonable generates echo).Retrieving that VM message from an outside location creates another 7db loss (now -14db down in total), making it very difficult (if not impossible) to hear the message. (And, yes I've gone through all the recommendations with wav vs gsm files, etc.) I am not sure I understand why the txgain/rxgain isn't fixing it without adding unacceptable echo...this all seems very odd...I mean for a test you should be able to dial an echo() application and have extremely quiet echoed audio... is this the case?As an ex-telco transmission engineer, believe me I've done my homeworkand some very solid testing with expensive well-calibrated test equipment. As I've mentioned to Kevin, its almost like the TigerJet pci controlleron the TDM card is reversing bits six and seven (or something very oddlike that). Digium apparently now has a pci engineering type looking at the issues, which I'm told is using a pci logic analyzer, etc. The work around only kicks in if the call comes from a zap channel and ends up in voicemail, adding a 6db gain to that recorded message. No other channel types are impacted by this new parameter. This is a HELL of a band-aid.If you actually follow the logic that was originally stated in 2023,this gain setting is highly useful for those systems that are further away from the CO (as mentioned above). For those closer tothe CO, it has zero value.Rich___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Message Waiting Indicator (MWI) for remote voice mail?
I have Asterisk voice mail setup locally. It works great, I'm impressed! Some details about my system: I'm using a TDM22B card to interface with both the PSTN and a VOIP provider. I'm running 1.2-beta from CVS. I have a regular VTech phone plugged into one of the FXS ports. Asterisk is able to indicate when a local voicemail message is waiting via the LCD display of my analog phone. It also gives a broken dial tone. This is achieved by specifying mailbox=mb# in zapata.conf and possibly also by specifing adsi=yes in the same file. The question I have is this: I also have voicemail with my VOIP provider. Before jumping into Asterisk, the VOIP provider could send the message waiting indicator to my phone when I had new messages. After putting Asterisk between my analog handset and the VOIP adapter, the message waiting indicator from the VOIP provider seems to no longer get through to the phone. The connection to the VOIP provider is Cable Modem - Sipura 3002 - TDM FXO interface - TDM FXS interface - phone. Is there a way for Asterisk to get notified and pass the message waiting indicator on to my handset when there is a voice mail waiting at the VOIP provider? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SOHO Survey / Creative Asterisk Solutions
I hope the subject isn't too buzzword compliant :) I'm just curious: What have people done with Asterisk? I'm particularly interested in DIY projects and things that can be done on a small/home office (or even hobbiest's) budget. If you have clever hacks or creative functionality you've implemented, I'd love to hear what a few people have come up with. Thanks! -Brian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SOHO Survey / Creative Asterisk Solutions
Yes, and it is a fantastic resource! I don't think I could have gotten up and running without it. I assume you pointed me there because there are some articles about what people are doing with *. I'll dig for them. I previously read in an issue of ;Login: several columns discussing things you could do with Asterisk. Articles are good, I was just hoping to get a couple of quick ideas about what people on the list have accomplished. Things like I set up time and temperature for whenever I dial *22 ... or something like that. Figured it might be a change for people to show off a bit and for me to get some ideas/get inspired ;-) Thanks again for the voip-info ref, I'll work my way through that wiki.On 9/22/05, Damon Estep [EMAIL PROTECTED] wrote: Have you discovered www.voip-info.org yet? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Brian McEntire Sent: Thursday, September 22, 2005 6:18 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SOHO Survey / Creative Asterisk Solutions I hope the subject isn't too buzzword compliant :) I'm just curious: What have people done with Asterisk? I'm particularly interested in DIY projects and things that can be done on a small/home office (or even hobbiest's) budget. If you have clever hacks or creative functionality you've implemented, I'd love to hear what a few people have come up with. Thanks! -Brian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SayUnixTime in CVS?
Can anyone tell me what I missed? I'm trying to setup a simple extension (400) that reports the time when it is dialed. I searched the threads and it seems like this should work... Here's what's in my extensions.conf: exten = 400,1,Answer() exten = 400,n,Wait,1 exten = 400,n,SayUnixTime(,EST5EDT,) exten = 400,n,Playback(tt-weasels) [BTW, tt-weasels is hillarious! ;-) Props to whoever made that. ] When I call the extension, it answers and immediately falls through to tt-weasels, which I hear fine. It's like SayUnixTime gets jumped over or returns nothing (very quickly.) I checked and the at sound which is used by the default SayUnixTime format string is in place. Also, SayDigits and SayNumber seem to work okay. When asterisk starts up in verbose mode, it looks like SayUnixTime gets loaded okay: [app_sayunixtime.so]Sep 23 03:24:04 VERBOSE[3854] logger.c: [app_sayunixtime.so] = (Say time) == Registered application 'SayUnixTime' == Registered application 'DateTime' This is what is output from asterisk -vvv when I dial 400: -- Starting simple switch on 'Zap/1-1' -- Executing Answer(Zap/1-1, ) in new stack -- Executing Wait(Zap/1-1, 1) in new stack -- Executing SayUnixTime(Zap/1-1, |EST5EDT|) in new stack -- Executing Playback(Zap/1-1, tt-weasels) in new stack -- Playing 'tt-weasels' (language 'en') -- Executing Wait(Zap/1-1, 5) in new stack Any ideas? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SOHO Survey / Creative Asterisk Solutions
Hehe... that's awesome :) I laughed out loud when I read it. Someone else replied that they are going to use * to control their entry gate system by cell phone. Nice. Thanks for the examples! While reading over at voip-info.org, I found the auto-dial feature that can be combined with .call files. That should be perfect for an idea I had -- use cron and POP3 to check my e-mail account for any new messages from the transit authority... if there are any, dial/ring home phones at 6am and playback a message to check e-mail for possible morning commute problems. Not quite as good as drunkdial though :) On 9/22/05, Tom Hayden [EMAIL PROTECTED] wrote: Well, there are about a billion DIY * projects out there.Myslightly-insane use of asterisk on a tiny budget is running thesite:www.drunkdial.orgPeople call in - leave drunk messages then comment on them later. Cheers,--Tom HaydenOn 9/22/05, Brian McEntire [EMAIL PROTECTED] wrote: I hope the subject isn't too buzzword compliant:) I'm just curious: What have people done with Asterisk? I'm particularly interested in DIY projects and things that can be done on a small/home office (or even hobbiest's) budget. If you have clever hacks or creative functionality you've implemented, I'd love to hear what a few people have come up with.Thanks!-Brian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --Tom ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SayUnixTime in CVS?
I've partially figured it out: The voip-info.org wiki for SayUnixTime suggests that if you don't specify the 3rd parameter, it will default to: ABdY'digits/at'IMp For some reason, for my setup, the default wasn't working when I left the parameter blank. When I specified the format in the extensions line, it works fine: exten = 400,n,SayUnixTime(,EST5EDT,ABdY \'digits/at\' IMp ) I looked at the code and it seems to attempt to set the default format string based on chan-language ... maybe my language isn't set up correctly. Anyway, hope that may be useful to someone. On 9/22/05, Brian McEntire [EMAIL PROTECTED] wrote: Can anyone tell me what I missed? I'm trying to setup a simple extension (400) that reports the time when it is dialed. I searched the threads and it seems like this should work... Here's what's in my extensions.conf: exten = 400,1,Answer() exten = 400,n,Wait,1 exten = 400,n,SayUnixTime(,EST5EDT,) exten = 400,n,Playback(tt-weasels) [BTW, tt-weasels is hillarious! ;-) Props to whoever made that. ] When I call the extension, it answers and immediately falls through to tt-weasels, which I hear fine. It's like SayUnixTime gets jumped over or returns nothing (very quickly.) I checked and the at sound which is used by the default SayUnixTime format string is in place. Also, SayDigits and SayNumber seem to work okay. When asterisk starts up in verbose mode, it looks like SayUnixTime gets loaded okay: [app_sayunixtime.so]Sep 23 03:24:04 VERBOSE[3854] logger.c: [app_sayunixtime.so] = (Say time) == Registered application 'SayUnixTime' == Registered application 'DateTime' This is what is output from asterisk -vvv when I dial 400: -- Starting simple switch on 'Zap/1-1' -- Executing Answer(Zap/1-1, ) in new stack -- Executing Wait(Zap/1-1, 1) in new stack -- Executing SayUnixTime(Zap/1-1, |EST5EDT|) in new stack -- Executing Playback(Zap/1-1, tt-weasels) in new stack -- Playing 'tt-weasels' (language 'en') -- Executing Wait(Zap/1-1, 5) in new stack Any ideas? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Caller ID and Call Parking on an analog PSTN line?
Hello everyone. I'm new to Asterisk but got some basic functionality going last night and I'm just giddy to have my own PBX ;-) Sorry if these are silly questions: My Asterisk server has the TDM22B (2 FXO, 2 FXS) interface. I have a very basic PSTN line coming in from the phone company, I tried to get the most no-frills line possible (didn't pay for caller ID, voice mail, etc.). I know I can set up voicemail on * on this line. Can I also get caller ID by virtue of running Asterisk or is that information stripped out by the phone co before it gets to my box? Thanks for any advice. I hope to get up to speed on Asterisk and be able to contribute back to the list in time. :) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID and Call Parking on an analog PSTN line?
Okay, thanks all for the feedback! I accidentally sent this response to one person who replied rather than to the list so I'm going post it here too: I didn't go into full details in my message. Part of the reason of setting up Asterisk is because I recently signed up with VOIP service and I'm extremely happy with it. I'm only keeping the PSTN line for emergencies and because my local number currently isn't portable. Perhaps I'll be able to make * play a message to incoming PSTN callers announcing my new VOIP number before ringing the inside extensions? This could help speed the transition to almost purely VOIP, and at some later date I could dump incoming PSTN calls to VM or just reject them altogether. I do have lots of services on the VOIP line, callerID included, and so I should be able to use call filtering on that line. - - - Thanks for the tip about dial tone sensing. I made a couple test calls last night and things went smoothly, but if it's an intermittent problem, I'll watch for it. I'm running version 1.2 beta from CVS. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
