Re: [asterisk-users] FXS ports on TDM410P card...
[Oct 15 01:48:02] NOTICE[3747] channel.c: Dropping incompatible voice
frame on SIP/2006- of format ulaw since our native format has
changed to 0x8 (alaw)
[Oct 15 01:48:49] WARNING[3750] pbx.c: Channel 'DAHDI/1-1' sent into
invalid extension 's' in context 'default', but no invalid handler
--
I have the code to set up an extension for toggling Telco pass through
working I think. What isn't working is the pass through. I get the
above error messages when I try to call the POTS line connected to
DAHDI/1 from my Comcast line.
I'm noticing other warning messages cropping up about this file or that
file not existing and modules not loading, but mostly the system seems
to be working so I'm wondering if these warnings are relevant. I'm
using Asterisk 1.8.
I think that [from-pstn] isn't working...
For those who don't know what I'm after, I'm trying when a phone company
call comes in to ring SIP phones and local FXS lines on my TDM410P. The
purpose of the toggle is to be able to disable this feature. Sometimes,
I really want to use this system as a private intercom system where at
other times, ringing remote SIP phones for an incoming telephone company
call might be needed. Say you are at extension 2000 or 2002, SIP phones
in other buildings, and you want or need to be able to receive calls
from the PSTN.
I'm in the U.S., under the [external] section am I blocking long
distance outgoing phone calls? In the U.S., you dial 1 and then
the number for long distance. Essentially, what I need to do is
block dialing 1 and then a number with the exception of 1-800 or
1-866.
Thank you for taking the time to look at my questions and
information ;-)
My current extensions.conf file in it's entirety follows:
-
[globals]
CENTURYLINK=DAHDI/1
COMCAST=DAHDI/2
ANDREWROOM=DAHDI/3
SERVERROOM=DAHDI/4
WIDE_PBX=SIP/2000SIP/2002SIP/2006SIP/2007SIP/2008SIP/2009
${SERVERROOM}${ANDREWROOM}
INSIDE_PBX=SIP/2006SIP/2007SIP/2008SIP/2009${SERVERROOM}
${ANDREWROOM}
OUTSIDE_PBX=SIP/2000SIP/2002
TELCO_ON=0
PSTN_THROUGH=${SERVERROOM}${ANDREWROOM}SIP/2000SIP/2002
[external]
exten = _9NXXNXX,1,Dial(${CENTURYLINK}/${EXTEN:1})
exten = _8NXXNXX,1,Dial(${COMCAST}/${EXTEN:1})
[my-phones]
exten = i,1,Playback(/var/lib/asterisk/sounds/custom/extns-list)
exten = i,n,Hangup()
exten = 2000,1,Dial(SIP/2000,40)
same = n,VoiceMail(2000,u)
exten = 2002,1,Dial(SIP/2002,40)
same = n,VoiceMail(2002,u)
exten = 2004,1,Dial(SIP/2004,40)
same = n,VoiceMail(2004,u)
exten = 2006,1,Dial(SIP/2006,40)
same = n,VoiceMail(2006,u)
exten = 2007,1,Dial(SIP/2007,40)
same = n,VoiceMail(2007,u)
exten = 2008,1,Dial(SIP/2008,40)
same = n,VoiceMail(2008,u)
exten = 2009,1,Dial(SIP/2009,40)
same = n,VoiceMail(2009,u)
exten = 2010,1,Dial(${SERVERROOM},40)
same = n,VoiceMail(2009,u)
exten = 2010,1,Dial(${SERVERROOM},40)
same = n,VoiceMail(2010,u)
exten = 2011,1,Dial(${ANDREWROOM},40)
same = n,VoiceMail(2011,u)
exten = 2012,1,Dial(${WIDE_PBX},40)
exten = 2013,1,Dial(${INSIDE_PBX},40)
exten = 2014,1,Dial(${OUTSIDE_PBX},40)
exten = 2015,1,GoToIf($[${TELCO_ON}=1]?2:5)
; 2 turns off telco_on
exten = 2015,2,Set(GLOBAL(TELCO_ON)=0)
exten = 2015,3,Playback(/var/lib/asterisk/sounds/custom/telco-off)
exten = 2015,4,hangup()
; 5 turns on telco_on
exten = 2015,5,Set(GLOBAL(TELCO_ON)=1)
exten = 2015,6,Playback(/var/lib/asterisk/sounds/custom/telco-on)
exten = 2015,7,hangup()
exten = 2999,1,VoiceMailMain(${CALLERID(num)},s)
[from-pstn]
exten = s,1,GoToIf($[${TELCO_ON}=1]?2:3)
; 2 rings all phones
exten = s,2,Dial(${PSTN_THROUGH},40)
exten = s,3,Hangup()
include = external
include = from-pstn
--
_
-- 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] FXS ports on TDM410P card...
On 10/15/2011 05:31 AM, Michael C. Robinson wrote:
[Oct 15 01:48:02] NOTICE[3747] channel.c: Dropping incompatible voice
frame on SIP/2006- of format ulaw since our native format has
changed to 0x8 (alaw)
[Oct 15 01:48:49] WARNING[3750] pbx.c: Channel 'DAHDI/1-1' sent into
invalid extension 's' in context 'default', but no invalid handler
--
I have the code to set up an extension for toggling Telco pass through
working I think. What isn't working is the pass through. I get the
above error messages when I try to call the POTS line connected to
DAHDI/1 from my Comcast line.
I'm noticing other warning messages cropping up about this file or that
file not existing and modules not loading, but mostly the system seems
to be working so I'm wondering if these warnings are relevant. I'm
using Asterisk 1.8.
I think that [from-pstn] isn't working...
You're not landing in [from-pstn]. Incoming calls are landing in
[default]. That's not a problem in extensions.conf, that's a problem in
dahdi.conf for those channels. They're not in the right context.
For those who don't know what I'm after, I'm trying when a phone company
call comes in to ring SIP phones and local FXS lines on my TDM410P. The
purpose of the toggle is to be able to disable this feature. Sometimes,
I really want to use this system as a private intercom system where at
other times, ringing remote SIP phones for an incoming telephone company
call might be needed. Say you are at extension 2000 or 2002, SIP phones
in other buildings, and you want or need to be able to receive calls
from the PSTN.
I'm in the U.S., under the [external] section am I blocking long
distance outgoing phone calls? In the U.S., you dial 1 and then
the number for long distance. Essentially, what I need to do is
block dialing 1 and then a number with the exception of 1-800 or
1-866.
[external]
exten = _91800NXX,1,Dial(${CENTURYLINK}/${EXTEN:1})
exten = _91888NXX,1,Dial(${CENTURYLINK}/${EXTEN:1})
exten = _91877NXX,1,Dial(${CENTURYLINK}/${EXTEN:1})
exten = _91866NXX,1,Dial(${CENTURYLINK}/${EXTEN:1})
exten = _91NXXNXX,1,Playback(im-sorry-Dave-you-cant-call-that)
exten = _91NXXNXX,n,Congestion
exten = _81800NXX,1,Dial(${COMCAST}/${EXTEN:1})
exten = _81888NXX,1,Dial(${COMCAST}/${EXTEN:1})
exten = _81877NXX,1,Dial(${COMCAST}/${EXTEN:1})
exten = _81866NXX,1,Dial(${COMCAST}/${EXTEN:1})
exten = _81NXXNXX,1,Playback(im-sorry-Dave-you-cant-call-that)
exten = _81NXXNXX,n,Congestion
That'll let you dial US toll free numbers out the channel specified by
dialing 9 or 8. It will playback a message and then generate a
congestion tone if some other number is dialed.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS ports on TDM410P card...
On Wednesday 12 October 2011, Michael C. Robinson wrote:
[stuff deleted]
Seems to be working now, good eyes.
I have the O'Reilly book on Asterisk 1.8, though I'm wondering where to
look in it to learn how to allow incoming calls from the phone company
to ring SIP box and FXS connected handsets? This would be a neat
feature, especially if there was a way from the handset to turn it off.
You already set a default context for incoming calls from the phone company.
You just need to specify multiple extensions in your Dial() statement,
delimited by and signs. For instance:
exten = s,1,Dial(DAHDI/3DAHDI/4SIP/301,60)
Or you can even create a global variable with the group of phones you want to
ring. For instance
[globals]
ANALOGUE=DAHDI/3DAHDI/4
BATPHONE=SIP/301
and later you can use something like
Dial(${ANALOGUE}${BATPHONE},60)
Just to make sure I'm clear, imagine you are hooked to a Linksys PAP2
and not only want to call out via the phone company but you want to
receive calls there from the phone company as well. Imagine there is
an extension to call that toggles this behavior on/off. So say 2025 is
the special extension which you call and a voice says relaying phone
company on. You hang up then, the phone rings, and you pick up a call
from somewhere remote via the phone company. You hang up when you're
done and decide the behavior should be turned off, so you dial 2025
again and the voice says relaying phone company off. Now if a call is
incoming from the phone company, your phone doesn't ring. You can call
all local extensions and even remote numbers, but you can't receive
remote calls.
You have to set a global variable when your special extension is dialled.
Then you can use GoToIf() to make decisions based on the value.
[globals]
TELCO_ON=1
ALL_PHONES=DAHDI/3DAHDI/4SIP/301
; (assuming 301 is the SIP extension you want to ring)
; .
[internal]
; .
exten = 2025,1,GoToIf($[${TELCO_ON}=1]?2:5)
; 2 turns off telco_on
exten = 2025,2,Set(GLOBAL(TELCO_ON)=0)
exten = 2025,3,Playback(telco-off)
exten = 2025,4,hangup()
; 5 turns on telco_on
exten = 2025,5,Set(GLOBAL(TELCO_ON)=1)
exten = 2025,6,Playback(telco-on)
exten = 2025,7,hangup()
; .
[from-pstn]
exten = s,1,GoToIf($[${TELCO_ON}=1]?2:3)
; 2 rings all phones
exten = s,2,Dial(${ALL_PHONES},60)
; 3 goes to voicemail (assuming you've configured VM on ext 301 .)
exten = s,3,VoiceMail(301,u)
exten = s,4,Hangup()
[stuff deleted] Another neat trick would be to list
what the extensions are when someone enters an invalid extension. Say
someone dials 1011, not one of my extensions and not a remote phone
number prefixed by 8 or 9.
Then you need to look at the i extension, which is called when someone dials
an invalid extension number. Record yourself a suitable message (cheating
way is to leave a voicemail message, which will already be in the format you
want, and cp the file across) and put something in your context like
exten = i,1,Playback(extns-list)
exten = i,n,Hangup()
The last trick I want to pull, I want an extension that will ring
inclusively 2000 to 2011, say 2012. How do I set this up by hand?
Again, use the sign notation for ringing multiple phones:
exten =
2012,1,Dial(SIP/2000SIP/2001SIP/2002SIP/2003SIP/2004SIP/2005SIP/2006SIP/2007SIP/2008SIP/2009SIP/2010SIP/2011,60)
exten = 2012,2,Hangup()
My personal preference is to split departments on the hundreds, and use x00 as
a ring all phones in department number. For instance if numbers like 2xx
are sales, 3xx are purchasing, 4xx are accounts, 5xx are IT, 6xx are factory
floor, then I would make the number to call everybody in accounts 400.
--
AJS
Answers come *after* questions.
--
_
-- 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] FXS ports on TDM410P card...
Quoting Michael C. Robinson plu...@robinson-west.com: My analog card, uses a PCI slot and a 12V power connector, is configured with 2 FXO and 2 FXS modules. I can ring handsets connected to the FXS ports but I can't dial out from them. Is extensions.conf where I need to make changes? You appear to have an inconsistency with context names... [root@robin asterisk]# cat chan_dahdi.conf ... [phone](!) ... context = myphones extensions.conf: ... [my-phones] Put that right and it should work, as you've designed it so far. -- Phil Reynolds mail: phil-aster...@tinsleyviaduct.com This message was sent using IMP, the Internet Messaging Program. -- _ -- 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] FXS ports on TDM410P card...
On Wednesday 12 October 2011, Michael C. Robinson wrote: My analog card, uses a PCI slot and a 12V power connector, is configured with 2 FXO and 2 FXS modules. I can ring handsets connected to the FXS ports but I can't dial out from them. Is extensions.conf where I need to make changes? If you can't make calls *from* a phone, but you can make calls *to* it, that suggests a problem with its default context. Your configuration snippets shew myphones as the default context in chan_dahdi.conf, but the context in the dialplan was my-phones. Make them match up, reload all configuration files and it should all Just Work. -- AJS Answers come *after* questions. -- _ -- 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] FXS ports on TDM410P card...
Changes so far: chan_dahdi.conf: [my-phones](!) . . . context = my-phones signalling = fxo_ks . . . [phone1](my-phones) . . . [phone2](my-phones) . . . [phone3](my-phones) . . . [phone4](my-phones) And extensions.conf is the same. Seems to be working now, good eyes. I have the O'Reilly book on Asterisk 1.8, though I'm wondering where to look in it to learn how to allow incoming calls from the phone company to ring SIP box and FXS connected handsets? This would be a neat feature, especially if there was a way from the handset to turn it off. Just to make sure I'm clear, imagine you are hooked to a Linksys PAP2 and not only want to call out via the phone company but you want to receive calls there from the phone company as well. Imagine there is an extension to call that toggles this behavior on/off. So say 2025 is the special extension which you call and a voice says relaying phone company on. You hang up then, the phone rings, and you pick up a call from somewhere remote via the phone company. You hang up when you're done and decide the behavior should be turned off, so you dial 2025 again and the voice says relaying phone company off. Now if a call is incoming from the phone company, your phone doesn't ring. You can call all local extensions and even remote numbers, but you can't receive remote calls. Another trick I want to pull is this. I have a few extensions, 2000 to 2011, where I'd like to have an extension someone can call to figure out what these extensions are. Say 1000 or even 0 if that will work. Something easy to remember anyways. Another neat trick would be to list what the extensions are when someone enters an invalid extension. Say someone dials 1011, not one of my extensions and not a remote phone number prefixed by 8 or 9. The last trick I want to pull, I want an extension that will ring inclusively 2000 to 2011, say 2012. How do I set this up by hand? Thank you again for helping me figure out the context problem. -- _ -- 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] FXS ports on TDM410P card...
My analog card, uses a PCI slot and a 12V power connector, is configured
with 2 FXO and 2 FXS modules. I can ring handsets connected to the FXS
ports but I can't dial out from them. Is extensions.conf where I need
to make changes?
[root@robin asterisk]# cat chan_dahdi.conf
[trunkgroups]
[channels]
[phone](!)
usecallerid = yes
hidecallerid = no
callwaiting = yes
usecallingpres = yes
callwaitingcallerid = yes
threewaycalling = yes
transfer = yes
canpark = yes
cancallforward = yes
callreturn = yes
echocancel = yes
echocancelwhenbridged = yes
relaxdtmf = yes
rxgain = 0.0
txgain = 0.0
group = 1
callgroup = 1
pickupgroup = 1
immediate = no
context = myphones
signalling = fxo_ks
[phone1](phone)
signalling = fxs_ks
callerid = Andrew F Robinson (503)543-2338
dahdichan = 1
[phone2](phone)
signalling = fxs_ks
callerid = Michael C Robinson (503)987-1322
dahdichan = 2
[phone3](phone)
callerid = 2010 2010
dahdichan = 3
[phone4](phone)
callerid = 2011 2011
dahdichan = 4
[root@robin asterisk]#
extensions.conf:
[globals]
CENTURYLINK=DAHDI/1
COMCAST=DAHDI/2
ANDREWROOM=DAHDI/3
SERVERROOM=DAHDI/4
[external]
exten = _9NXXNXX,1,Dial(${CENTURYLINK}/${EXTEN:1})
exten = _8NXXNXX,1,Dial(${COMCAST}/${EXTEN:1})
[my-phones]
exten = 2000,1,Dial(SIP/2000,40)
same = n,VoiceMail(2000,u)
exten = 2002,1,Dial(SIP/2002,40)
same = n,VoiceMail(2002,u)
exten = 2004,1,Dial(SIP/2004,40)
same = n,VoiceMail(2004,u)
exten = 2006,1,Dial(SIP/2006,40)
same = n,VoiceMail(2006,u)
exten = 2007,1,Dial(SIP/2007,40)
same = n,VoiceMail(2007,u)
exten = 2008,1,Dial(SIP/2008,40)
same = n,VoiceMail(2008,u)
exten = 2009,1,Dial(SIP/2009,40)
same = n,VoiceMail(2009,u)
exten = 2010,1,Dial(${SERVERROOM},40)
same = n,VoiceMail(2010,u)
exten = 2011,1,Dial(${ANDREWROOM},40)
same = n,VoiceMail(2011,u)
exten = 2999,1,VoiceMailMain(${CALLERID(num)},s)
include = external
--
_
-- 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
