Re: [asterisk-users] Any help with these error messages???
Moved to Asterisk 1.8.7, most of the watnings/errors are gone. I have a new error though: [Oct 21 13:40:40] ERROR[15709] ais/clm.c: Could not initialize cluster membership service: Try Again And I get a warning that no music on hold classes are configured. Never mind. The chan_dahdi warnings about the Pseudo channel was already fixed in -r331955 of the v1.8 SVN branch and is in v1.8.7. Richard -- _ -- 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
Re: [asterisk-users] Any help with these error messages???
[trunkgroups] [channels] [my-phones](!) 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 = my-phones signalling = fxo_ks [phone1](my-phones) signalling = fxs_ks callerid = Andrew F Robinson (503)543-2338 dahdichan = 1 [phone2](my-phones) signalling = fxs_ks callerid = Michael C Robinson (503)987-1322 dahdichan = 2 [phone3](my-phones) callerid = 2010 2010 dahdichan = 3 [phone4](my-phones) callerid = 2011 2011 dahdichan = 4 I don't see anywhere in the above file that I deal with pseudo. It looks like you are attempting to manually configure the pseudo channel multiple times in chan_dahdi.conf. You do not need to explicitly configure the pseudo channel. The pseudo channel is always created and has no settable configuration parameters as far as I know. Richard -- _ -- 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
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
--
_
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Re: [asterisk-users] Emulate and script emulation of users calling in/receiving calls, transferring calls etc
On Sat, 2011-10-15 at 20:12 +1100, Alec Taylor wrote: Good evening, If asterisk or freeswitch would be taught in a classroom environment, is there someway to emulate and script emulation of users calling in/receiving calls, transferring calls etc? The plan is to have each student setup there own Asterisk or FreeSwitch box, and measure handling efficiency, and communicate between the servers (by transferring calls from server to server). Thanks for all suggestions, Alec Taylor Are you needing to emulate telephone company analog service? There have been devices on EBay that do up to four analog lines which work like telephone company lines most likely for around $300. Pretty expensive, but probably cheaper than paying for telephone company lines and running the risk of someone placing a long distance phone call. http://www.ebay.com/itm/TELTONE-4-PORT-FXS-TELEPHONE-LINE-SIMULATOR-TLS-5-/150672380134?pt=LH_DefaultDomain_0hash=item2314c610e6 If the above link works, it is for a $500+ unit. I looked in the $300 and less bracket and mostly what the auctions said is untested. Search EBay with the phrase telephone line simulator. I hope this helps ;-) -- _ -- 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] Any help with these error messages???
[Oct 15 22:44:31] ERROR[29013] res_config_pgsql.c: PostgreSQL RealTime: Failed to connect database asterisk on 127.0.0.1: [Oct 15 22:44:31] WARNING[29013] res_config_pgsql.c: PostgreSQL RealTime: Couldn't establish connection. Check debug. [Oct 15 22:44:31] ERROR[29013] res_config_ldap.c: No directory URL or host found. [Oct 15 22:44:31] ERROR[29013] res_config_ldap.c: Cannot load LDAP RealTime driver. [Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Attempt to configure channel -2 with signaling Unknown signalling -1 ignored because it is already configured to be Pseudo. [Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Attempt to configure channel -2 with signaling Unknown signalling -1 ignored because it is already configured to be Pseudo. [Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Attempt to configure channel -2 with signaling Unknown signalling -1 ignored because it is already configured to be Pseudo. [Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Attempt to configure channel -2 with signaling Unknown signalling -1 ignored because it is already configured to be Pseudo. [Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Ignoring any changes to 'userbase' (on reload) at line 23. [Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Ignoring any changes to 'vmsecret' (on reload) at line 31. [Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Ignoring any changes to 'hassip' (on reload) at line 35. [Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Ignoring any changes to 'hasiax' (on reload) at line 39. [Oct 15 22:44:31] WARNING[29013] chan_dahdi.c: Ignoring any changes to 'hasmanager' (on reload) at line 47. [Oct 15 22:44:32] WARNING[29013] cel_pgsql.c: CEL pgsql config file missing global section. [Oct 15 22:44:33] ERROR[29013] ais/clm.c: Could not initialize cluster membership service: Try Again I'm too much of a newbie to use a database to hold the configuration files and besides that I want to manage the files manually until I know what I'm doing in a gui/database environment. I don't want to set up LDAP either at this time, I'm not sure it would give me anything. I'm not sure what the last error pertains to, again I probably need to shut something off. I'm running Asterisk 1.8.3. I'm concerned about the signaling errors and not sure what is causing them as my chan_dahdi.conf appears to be correct. -- _ -- 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
--
_
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