Re: [Asterisk-Users] Calls not incoming to any extension from remoteproxy server
On Thursday 22 December 2005 22:13, abhishek wrote:
Thanks a lot for the reply. But i am sucessfully getting registered on the
remote proxy, so that i am getting right outputs as u said. I think that is
why only i am able to route calls outside to remote proxy,
The problem is when i am writing
register = user:[EMAIL PROTECTED]/1234
, the outside calls are not coming to 1234 extension , which is a Xlite
client.
The files configuration are as
sip.conf
register = user:[EMAIL PROTECTED]/1234
[1234]
type=friend
host=dynamic
context=test_in
user=phone
regexten=1234
extensions.conf
[test_in]
exten= 1236,1,Dial(SIP/sandhu)
exten= 1235,1,Dial(SIP/1235)
exten= _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten= 1234,1,Dial(SIP/1234)
I would try to separate incoming and outgoing extensions
to different contexts, for instance:
[test_in]
exten= 1234,1,Dial(SIP/1234)
[test_out]
exten= _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
and make include = both to you [default] context and put
context = default in your [1234] definition
I think this was important in order to follow the correct dialing priority.
To see the difference you could type now: show dialplan test_in
and after : show dialplan default
Also when forming a dial string keep in mind that
X = any digit from 0-9, Z = any digit from 1-9,
N = any digit from 2-9
means to use:
exten= _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
when dialing US/Canada and:
exten= _9011N.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
when dialing other desitinations...
Another thing I can see now is that there isn't a peer
(or you don't show it?) for the remote proxy i.e.:
[remote_proxy]
type=peer (or friend)
host=proxy-ip
context=whatever_they_say
etc
Your [1234] is for the Xlite
and [remote_proxy] for your provider.
Hope that helps,
benchev
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RE: [Asterisk-Users] Tracing a crash with CAPI calls
On Wed, 21 Dec 2005, Andrew Gough wrote: On Wed, 21 Dec 2005, Andrew Gough wrote: I have just setup asterisk on a debian sarge box. I am running Asterisk 1.21 with AMP and chan_capi_cm 0.6.1 using a BT Speedway (AVM Fritz) ISDN card, connected to a BT ISDN2e line. Currently we have 6 extensions configured all using CounterPath(Xten) eyebeam softphone. After many hours of Googling I have finally got it all setup and working. We can transfer calls internally and make and receive external calls. Its all great except for stability issues!! Essentially every now and again, in the middle of a call (so far only external CAPI calls) asterisk simply dies. No warning, no error, just my console session outputs a disconnected from console message. The server is a brand new AMD 3400+ with 512Mb RAM. The other issue experienced is occasional break up on inbound sound quality. I don't expect anyone to be able to solve this one straight away but I am at a loss where to look, I have tried /var/logs/messages Please try latest CVS sources of chan_capi from sourceforge. There are some bugfixes which may help you. If this doesn't help, please provide a log (set verbose 5 , capi debug) of the error. Armin Ok got the lastest sources from cvs for chan_capi and installed it. Left console seesion running with setting suggested, Crashed again twice this afternoon. Below is the output, not sure if it tells you anything. The CAPI details look good, there is no error and the call is released correctly. It looks like the crash happens after the other lines when the SIP call is released: -- Executing Hangup(SIP/113-a753, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/113-a753' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/113-a753' So at this point it doesn't look like a chan_capi problem. Can you create a backtrace (with a core dump)? Armin - == Spawn extension (macro-dialout-trunk, s, 21) exited non-zero on 'SIP/113-a753' in macro 'dialout-trunk' == Spawn extension (from-internal, 907950846621, 1) exited non-zero on 'SIP/113-a753' -- Executing Macro(SIP/113-a753, hangupcall) in new stack -- Executing ResetCDR(SIP/113-a753, w) in new stack INFO_IND ID=001 #0x3473 LEN=0017 Controller/PLCI/NCCI= 0x101 InfoNumber = 0x8 InfoElement = 80 90 INFO_RESP ID=001 #0x3473 LEN=0012 Controller/PLCI/NCCI= 0x101 -- ISDN1: info element CAUSE 80 90 DISCONNECT_B3_CONF ID=001 #0x2927 LEN=0014 Controller/PLCI/NCCI= 0x10101 Info= 0x0 DISCONNECT_B3_IND ID=001 #0x3474 LEN=0015 Controller/PLCI/NCCI= 0x10101 Reason_B3 = 0x3301 NCPI= default DISCONNECT_B3_RESP ID=001 #0x3474 LEN=0012 Controller/PLCI/NCCI= 0x10101 DISCONNECT_REQ ID=001 #0x2928 LEN=0017 Controller/PLCI/NCCI= 0x101 AdditionalInfo BChannelinformation= default Keypadfacility = default Useruserdata = default Facilitydataarray = default DISCONNECT_CONF ID=001 #0x2928 LEN=0014 Controller/PLCI/NCCI= 0x101 Info= 0x0 -- Executing NoCDR(SIP/113-a753, ) in new stack -- Executing Wait(SIP/113-a753, 5) in new stack DISCONNECT_IND ID=001 #0x3475 LEN=0014 Controller/PLCI/NCCI= 0x101 Reason = 0x3490 DISCONNECT_RESP ID=001 #0x3475 LEN=0012 Controller/PLCI/NCCI= 0x101 CAPI INFO 0x3490: Normal call clearing == ISDN1: Interface cleanup PLCI=0x101 -- Executing Hangup(SIP/113-a753, ) in new stack == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/113-a753' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/113-a753' rapid*CLI Disconnected from Asterisk server Regards Andrew Gough ___ --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
Re: [Asterisk-Users] chan_capi-cm 0.6.1 won't load
On Wed, 21 Dec 2005, Johan Helsingius wrote: Asterisk 1.2.1 on gentoo. Trying to use chan_capi-cm 0.6.1 results in: WARNING[11724] loader.c: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: use_ast_mutex_init_instead_of_pthread_mutex_init WARNING[11724] loader.c: Loading module chan_capi.so failed! Sounds like some version incompatibility - any ideas? Please use latest sources from CVS on sourceforge for chan_capi-cm Armin ___ --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] How to record a call
On Thursday 22 December 2005 07:36, Stefan Reuter wrote: http://www.voip-info.org/wiki-Asterisk+cmd+Monitor For Asterisk 1.2: http://www.voip-info.org/wiki/view/MixMonitor ___ --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] Problem with octobri and x100p clone
Hi everybody,
I have a problem with my *..
I have an Octobri Card working good..
and 2 x100p clones ..
the fact is that * modeprobes ok and load drivers
ok.
but when I want to make a call outside, I get this
error
-- Executing Dial("SIP/203-7f0a",
"Zap/g1/69X65|30") in new stackDec 19 12:33:51 NOTICE[3042]:
app_dial.c:759 dial_exec: Unable to create channel of type 'Zap'
the group 1are8 lines of the
Octobri
the group 2are2 x100p
clones
zapata.conf
file-
[EMAIL PROTECTED] asterisk]# vi
zapata.conf
;; Default
context;context=enlacesignalling=fxs_ksechocancel=yesechocancelwhenbridged=yesechotraining=800relaxdtmf=yesrxgain=4.0txgain=4.0busydetect=nocallprogress=nomusiconhold=defaultusecallerid=yescallerid=asreceivedchannel=25-26group=2
switchtype = euroisdn
signalling = bri_cpe; p2mp NT mode (for connecting ISDN phones in
point-to-multipoint mode);signalling = bri_net_ptmp; p2p NT mode (for
connecting an ISDN pbx in point-to-point mode);signalling = bri_net
pridialplan = localprilocaldialplan = localnationalprefix
=internationalprefix = 0
echocancel = yes
context=incominggroup = 1; S/T port 1channel = 1-2
group = 1; S/T port 2channel = 4-5
so on to latest Octobri port
group = 1; S/T port 8channel = 22-23
---end of zapata.conf file
zaptel.conf file-
[EMAIL PROTECTED] etc]# vi
zaptel.confloadzone=esdefaultzone=es# qozap span definitions#
most of the values should be bogus because we are not really zaptel
span=1,1,3,ccs,amispan=2,0,3,ccs,amispan=3,0,3,ccs,amispan=4,0,3,ccs,amispan=5,1,3,ccs,amispan=6,0,3,ccs,amispan=7,0,3,ccs,amispan=8,0,3,ccs,ami
bchan=1,2dchan=3bchan=4,5dchan=6bchan=7,8dchan=9bchan=10,11dchan=12bchan=13,14dchan=15bchan=16,17dchan=18bchan=19,20dchan=21bchan=22,23dchan=24fxsks=25-26--end
of zaptel.conf file -
someone could help me?
thanks in advance
Bye
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Re: [Asterisk-Users] Problem with octobri and x100p clone
have you checked the order the modules are loaded and that this matches the zaptel.conf ? [EMAIL PROTECTED] wrote on 22.12.2005 09:48:55: Hi everybody, I have a problem with my *.. I have an Octobri Card working good.. and 2 x100p clones .. the fact is that * modeprobes ok and load drivers ok. but when I want to make a call outside, I get this error -- Executing Dial(SIP/203-7f0a, Zap/g1/69X65|30) in new stack Dec 19 12:33:51 NOTICE[3042]: app_dial.c:759 dial_exec: Unable to create channel of type 'Zap' the group 1 are 8 lines of the Octobri the group 2 are 2 x100p clones zapata.conf file- [EMAIL PROTECTED] asterisk]# vi zapata.conf ; ; Default context ; context=enlace signalling=fxs_ks echocancel=yes echocancelwhenbridged=yes echotraining=800 relaxdtmf=yes rxgain=4.0 txgain=4.0 busydetect=no callprogress=no musiconhold=default usecallerid=yes callerid=asreceived channel=25-26 group=2 switchtype = euroisdn signalling = bri_cpe ; p2mp NT mode (for connecting ISDN phones in point-to-multipoint mode) ;signalling = bri_net_ptmp ; p2p NT mode (for connecting an ISDN pbx in point-to-point mode) ;signalling = bri_net pridialplan = local prilocaldialplan = local nationalprefix = internationalprefix = 0 echocancel = yes context=incoming group = 1 ; S/T port 1 channel = 1-2 group = 1 ; S/T port 2 channel = 4-5 so on to latest Octobri port group = 1 ; S/T port 8 channel = 22-23 ---end of zapata.conf file zaptel.conf file- [EMAIL PROTECTED] etc]# vi zaptel.conf loadzone=es defaultzone=es # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami span=5,1,3,ccs,ami span=6,0,3,ccs,ami span=7,0,3,ccs,ami span=8,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 bchan=13,14 dchan=15 bchan=16,17 dchan=18 bchan=19,20 dchan=21 bchan=22,23 dchan=24 fxsks=25-26 --end of zaptel.conf file - someone could help me? thanks in advance Bye___ --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: Postgres
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi, I am new to Asterisk problems. Could anyone tell me how to install asterisk with postgres cdr feature. Because I install asterisk 1.2 from newest Bristuff and I do not have it Thanks in advance http://www.voip-info.org/wiki/view/Asterisk+cdr+pgsql I use MySQL so this is the most I can help you. -- Tomislav Parcina [EMAIL PROTECTED] ___ --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: Asterisk Call Forwarding
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi Tim, probably my information are not quite clear; 3473774567 is a mobile phone and 105 is an extension. I would like to forward any outside calling from this mobile (3473774567) to the extension 105. When you talk about DB, what do you mean exactly? Could you be so kind to post some examples so the * forward calling function will be more clear. Thanks a lot. Do your self a favor, read some Asterisk tutorial. You will start using Asterisk much faster then asking evry simple thing on mailing list. On this pages http://www.voip-info.org/tiki-index.php?page=Asterisk you have a list of Howtos and Tutorials. -- Tomislav Parcina [EMAIL PROTECTED] ___ --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] Asterisk server to provide virtuals IPBX
Have you test it for virtuals IPBX ? --- C F [EMAIL PROTECTED] a écrit : The workaround for the parking limitation is app_valetparking.so from http://www.pbxfreeware.org/app_valetparking.c instructions on how to install is on the wiki. On 12/21/05, Olle E Johansson [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: Hello, Is Asterisk able to provide virtuals IPBX ? I mean one hardware server which handle one IPBX per enterprise . A lot of service providers do that. One caveat is the parking function, that only supports one parking lot for all virtual PBXs. /O ___ --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 ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --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: Aastra 9133i directory list downloading
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... How do you configure aastra.cfg to download directory list entries to each phone? The Aastra documentation is very sketchy. Anyone have an example??? Please stop replaying to mesage. If you plan to open thread do so by writing mail to this address asterisk-users@lists.digium.com -- Tomislav Parcina [EMAIL PROTECTED] ___ --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] php agi problem (perhaps problem..)
Thank you for your answer. I tried your suggestion, but probably in the wrong way, and I had no success. I tried: asteriskge03:/etc/asterisk # cat phpagi.conf debug=false [festival] text2wave=/usr/src/festival/bin/text2wave tempdir=/var/lib/asterisk/sounds/tmp/ and also asteriskge03:/etc/asterisk # cat phpagi.conf [general] debug=false [festival] text2wave=/usr/src/festival/bin/text2wave tempdir=/var/lib/asterisk/sounds/tmp/ What am I doing wrong ? It was impossible (for me...) to find any sample about phpagi.conf, both in wiki and in google. I only found: http://phpagi.sourceforge.net/phpagi2/docs/ric_README.html wich speak about a phpagi.conf sample file, but no chance to find a link thanks in advance, Andrea trixter aka Bret McDanel [EMAIL PROTECTED] To ad.com Asterisk Users Mailing List - Sent by: Non-Commercial Discussion asterisk-users-bo asterisk-users@lists.digium.com [EMAIL PROTECTED] cc m.com Subject Re: [Asterisk-Users] php agi 21/12/2005 14.44 problem (perhaps problem..) Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com On Wed, 2005-12-21 at 14:15 +0100, [EMAIL PROTECTED] wrote: My /var/log/messages log is very full of a lot of line regarding php agi scripts, i.e Dec 21 10:36:00 asteriskge03 php: agi Object Dec 21 10:36:00 asteriskge03 php: ( Dec 21 10:36:00 asteriskge03 php: [request] = Array Dec 21 10:36:00 asteriskge03 php: ( these are caused when phpagi.conf has debug=true (technically anything that is not false). Dec 21 10:36:01 asteriskge03 php: Could not parse /etc/asterisk/localprefixes.conf that is a problem with your agi and does not appear to be related to phpagi. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group (See attached file: signature.asc) ___ --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 signature.asc Description: Binary data ___ --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] Asterisk server to provide virtuals IPBX
Does these providers use blabe servers for reliability and scalability ? Harry --- Olle E Johansson [EMAIL PROTECTED] a écrit : [EMAIL PROTECTED] wrote: Hello, Is Asterisk able to provide virtuals IPBX ? I mean one hardware server which handle one IPBX per enterprise . A lot of service providers do that. One caveat is the parking function, that only supports one parking lot for all virtual PBXs. /O ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --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] chan_capi-cm 0.6.1 won't load
Please use latest sources from CVS on sourceforge for chan_capi-cm Seems to have helped! Thanks! Julf ___ --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] Problem with octobri and x100p clone
the thing is: I first had problems loading the drivers and was the zaptel
init file that I will post below
that thing... the order the modules are loaded was the first problem that I
solved and asterisk starts perfectly, the error commented still showing.
here is the zaptel init file --- just a part including the loading of
modules and the ztcfg start
--
action Loading zaptel framework: modprobe zaptel loads
zaptel first
# sleep 10
this was my first try with errors - is commented now
# action Loading zaptel second: modprobe wcfxo
echo -n Waiting for zap to come online ...
TMOUT=10 # max secs to wait
while [ ! -d /dev/zap ] ; do
sleep 1
TMOUT=`expr $TMOUT - 1`
if [ $TMOUT -eq 0 ] ; then
echo Error: missing /dev/zap!
exit 1
fi
done
echo OK
echo -n Loading zaptel hardware modules:
for x in $MODULES; do
if insmod ${x} ${ARGS} /dev/null; then
echo -n $x
fi
done
echo
sleep 5;
action Loading wcfxo: modprobe wcfxo
I put the wcfxo here - loads OK
sleep 5;
action Running ztcfg: /sbin/ztcfg
RETVAL=$?
[ $RETVAL -eq 0 ] touch /var/lock/subsys/zaptel
;;
stop)
---end of part of zaptel init file
Any hint?
Thanks in advance...
Agustin
PS: excuse my english
- Original Message -
From: [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, December 22, 2005 9:55 AM
Subject: Re: [Asterisk-Users] Problem with octobri and x100p clone
have you checked the order the modules are loaded and that this matches
the zaptel.conf ?
[EMAIL PROTECTED] wrote on 22.12.2005 09:48:55:
Hi everybody,
I have a problem with my *..
I have an Octobri Card working good..
and 2 x100p clones ..
the fact is that * modeprobes ok and load drivers ok.
but when I want to make a call outside, I get this error
-- Executing Dial(SIP/203-7f0a, Zap/g1/69X65|30) in new stack
Dec 19 12:33:51 NOTICE[3042]: app_dial.c:759 dial_exec: Unable to create
channel of type 'Zap'
the group 1 are 8 lines of the Octobri
the group 2 are 2 x100p clones
zapata.conf
file-
[EMAIL PROTECTED] asterisk]# vi zapata.conf
;
; Default context
;
context=enlace
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
relaxdtmf=yes
rxgain=4.0
txgain=4.0
busydetect=no
callprogress=no
musiconhold=default
usecallerid=yes
callerid=asreceived
channel=25-26
group=2
switchtype = euroisdn
signalling = bri_cpe
; p2mp NT mode (for connecting ISDN phones in point-to-multipoint mode)
;signalling = bri_net_ptmp
; p2p NT mode (for connecting an ISDN pbx in point-to-point mode)
;signalling = bri_net
pridialplan = local
prilocaldialplan = local
nationalprefix =
internationalprefix = 0
echocancel = yes
context=incoming
group = 1
; S/T port 1
channel = 1-2
group = 1
; S/T port 2
channel = 4-5
so on to latest Octobri port
group = 1
; S/T port 8
channel = 22-23
---end of zapata.conf file
zaptel.conf file-
[EMAIL PROTECTED] etc]# vi zaptel.conf
loadzone=es
defaultzone=es
# qozap span definitions
# most of the values should be bogus because we are not really zaptel
span=1,1,3,ccs,ami
span=2,0,3,ccs,ami
span=3,0,3,ccs,ami
span=4,0,3,ccs,ami
span=5,1,3,ccs,ami
span=6,0,3,ccs,ami
span=7,0,3,ccs,ami
span=8,0,3,ccs,ami
bchan=1,2
dchan=3
bchan=4,5
dchan=6
bchan=7,8
dchan=9
bchan=10,11
dchan=12
bchan=13,14
dchan=15
bchan=16,17
dchan=18
bchan=19,20
dchan=21
bchan=22,23
dchan=24
fxsks=25-26
--end of zaptel.conf file -
someone could help me?
thanks in advance
Bye___
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[Asterisk-Users] unplugging E1 cables while asterisk running
Yesterday I've had to unplug one cable coming from a TE410 card to plug it in another hole, due to provider's changes in the patch panel. The calls on that span stopped working (can't create zap channel), the problem was solved restarting asterisk. Note that the PRI termination hasn't changed, only moved the cables connecting the card to it from one patch panel to another. The cable's guy told me that unplugging and quickly replugging E1 cables isn't a problem on traditional systems, anyone konws the reason or if it is a bug that should be reported ? ___ --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] realtime SIP
Hello, Is there a dedicated GUI to manage sip buddies with realtime ? I looked at voip-info I don't find it . Asterisk===WEB/odbc=SQL database with sip_buddies table || GUI Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --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] Asterisk - Gizmo
Original Message From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, December 21, 2005 9:14 AM Subject: Re: [Asterisk-Users] Asterisk - Skype anywhere/anyhow? On Tue, 20 Dec 2005, AR Tarzi wrote: could you please tell how it interfaces with Asterisk? Could I receive calls into Asterisk? send calls out? I've just downloaded it and am searching (unsuccessfully) for these on Gizmo's site/software. Gizmo isn't just a soft phone. Like Skype, its a service. Unlike Skype, though, the service is open to the rest of the SIP world. So - to call your Asterisk system from Gizmo, simply tell Gizmo to dial [EMAIL PROTECTED] To call Gizmo from Asterisk, simply tell it to dial SIP/[EMAIL PROTECTED] It 'sort of works'. I can call from gizmo to my *, but the url for incoming is SIP/[EMAIL PROTECTED] DTMF from gizmo does not work If gizmo is dialing into the queue, gizmo doesn't notice the prompts from * (which I can see in the *log), but keeps playing ringtones. But when the phone is answered, gizmo knows. and the connection is made. (The queue works as expected, when I call from eg my cellphone to *) So an Answer() is needed before queue(). Leif ___ --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] IAX No Authority found
Guys,
I,m facing a little tricky issue here, is there anybody that faced the same
issue or knows how to solve this?
I have 2 *, trunked with IAX
From ServerA I can call ServerB without any problems
If I call from ServerB to ServerA i get the following message :
ServerB
Dec 21 11:12:23 WARNING[2849]: chan_iax2.c:6967 socket_read: Call rejected
by 10.0.100.125: No authority found
-- Hungup 'IAX2/campinas-16384'
== No one is available to answer at this time (1:0/0/0)
== Auto fallthrough, channel 'IAX2/8512-7' status is 'NOANSWER'
-- Hungup 'IAX2/8512-7'
ServerA
Dec 21 13:08:12 NOTICE[2420]: chan_iax2.c:6772 socket_read: Rejected connect
attempt from 10.20.0.20, who was trying to reach '[EMAIL PROTECTED]'
The iasx.conf is as follows:
serverB
[campinas]
qualify=yes
type=friend
auth=rsa
;username=campinas
;secret=campinasvoip
host=10.0.100.125
trunk=yes
notransfer=yes
disallow=all
allow=speex
serverA
[saopaulo]
qualify=yes
type=friend
auth=rsa
;username=saopaulo
;secret=saopaulovoip
host=10.20.0.20
trunk=yes
notransfer=yes
The Dial string is this one:
On serverA
exten = _85XX,1,Dial(IAX2/saopaulo/${EXTEN},60,t)
On serverB
exten = _74XX,1,Dial(IAX2/campinas/${EXTEN},60,t)
Is there anything missing ???
Happy holidays to you all !!!
Leandro Martini
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[Asterisk-Users] *1.2.1 setcidnum from Zap
Hello
I'm testing asterisk 1.2.1 now and I have a problem with change
CALLERIDNUM on calls incommings from Zap channel.
My * is connected to our Ericsson PBX over Digium E1 (module
wcte11xp). The calls from Ericsson have CLI without prefix, only
extension. So I need to add our prefix to this calls. The calls go out
over a next E1 or an IAX2 trunk.
If I use
Set(CALLERID(number)=531011${CALLERIDNUM})
or
Setcidnum(531011${CALLERIDNUM})
the CLI isn't changed.
When I used *1.0.x with Setcidnum(531011${CALLERIDNUM}), everything
was working great.
Do you have any idea ?
Thank you
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[Asterisk-Users] Re: Semi OT - SuperMicro config question for the Linux/Hardware jedi's - $50 bounty!
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... That motherboard has an Adaptec/Marvell SATA controller, which does not have a open source driver in those distributions. There is an open-source driver in the very latest 2.6 kernel releases, but it won't be included in the installer kernels for any of those distros. Is there a list of controller's that are supported by specific version of kernel? I have Intel MB D865 and kernel 2.6.11 doesn't support it. Where can I see does any other kernel supports it? -- Tomislav Parcina [EMAIL PROTECTED] ___ --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: How to record a call
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... For Asterisk 1.2: http://www.voip-info.org/wiki/view/MixMonitor Can this one be done on demand? Like, I dial *1 and it starts recording. -- Tomislav Parcina [EMAIL PROTECTED] ___ --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] Problem with octobri and x100p clone
On Thu, Dec 22, 2005 at 11:06:50AM +0100, Agustin Gudiño wrote: the thing is: I first had problems loading the drivers and was the zaptel init file that I will post below What's wrong with a simple 'modprobe qozap; modprobe wcfxo; ztcfg' ? (With no sleeps). Just remove the post-install executions of ztcfg from /etc/modprobe.conf -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] Semi OT - SuperMicro config question for the Linux/Hardware jedi's - $50 bounty!
On Wed, Dec 21, 2005 at 10:10:02PM -0600, Kevin P. Fleming wrote: That motherboard has an Adaptec/Marvell SATA controller, which does not have a open source driver in those distributions. There is an open-source driver in the very latest 2.6 kernel releases, but it won't be included in the installer kernels for any of those distros. You will either have to: A) use the Intel SATA controller and skip the Adaptec one until you get the distro up and running and can update the kernel B) build your own installer kernel using a more up-to-date kernel source tarball C) build a module you could load from a 'driver disk' for the distros that support them, using a backport of the open-source driver D) (variant of (A)) Find a linux system that can boot (e.g: some latest knoppix) and instsll the target distro from the current distro. Installing from a running distro is a concept that Gentoo users are probably very familiar with. It is also officially supported (documented in the installation guide) in Debian. I dont know about other distros. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] Problem with octobri and x100p clone
On Thu, 22 Dec 2005, Tzafrir Cohen wrote: On Thu, Dec 22, 2005 at 11:06:50AM +0100, Agustin Gudiño wrote: the thing is: I first had problems loading the drivers and was the zaptel init file that I will post below What's wrong with a simple 'modprobe qozap; modprobe wcfxo; ztcfg' ? (With no sleeps). I need even longer sleeps (TDM11B only box), if I don't use the sleeps my RedHat Enterprise Linux 4.x rebuild will fail to load the second module after zaptel. I don't know why, it seems that the second module is very impatient and also that RHEL is slow setting up the proper udev stuff. ___ --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] PRI problems: B-Channel restart
I have a TE205P card with sapn 1 connected to the TELCO and span 2 connected to a PBX, The 2 spans have the following configuration: span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 The problem is that I get periodic B-Channel restart on both spans (altough it is much more frequent on span 1 connected to the telco) during calls which result in call drops --B-channel 0/1 successfully restarted on span1 --B-channel 1/1 successfully restarted on span1 --B-channel 2/1 successfully restarted on span1 . . . --B-channel 31/1 successfully restarted on span1 The channel restarts are less frequent when I restart Asterisk and then grow in frequency. Does anyone have an idea how to solve this problem? Thanks, Antoine Megalla. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Codec selection in dialplan
Is is possible to select (preferred) codec in dialplan using extensions.ael? For example, use 711 for extension 6004 (which is not physical extension) and 729 for anything else? ___ --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: How to record a call
On Thu, December 22, 2005 12:54, Tomislav Parcina said: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... For Asterisk 1.2: http://www.voip-info.org/wiki/view/MixMonitor Can this one be done on demand? Like, I dial *1 and it starts recording. http://www.voip-info.org/wiki-Asterisk+config+features.conf BTW: Please let me know when you've got this working 100%... I keep having issues with it! Most notably when dialling OUTBOUND with IAX softphone (tried borg DIAX and IDEFISK) Last time I checked, it worked for some of my DECT ISDN phones on ZAP (only the ones supporting 'dialpad mode') Looks to me like (*) has some issues with inband DTMF on outbound calls, but I need to test more before I can put together an exact description of the problem... (Next step is to test SIP phones with both RFC and inband DTMF) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] Problem with octobri and x100p clone
On Thu, Dec 22, 2005 at 01:15:31PM +0100, Remco Barende wrote: On Thu, 22 Dec 2005, Tzafrir Cohen wrote: On Thu, Dec 22, 2005 at 11:06:50AM +0100, Agustin Gudiño wrote: the thing is: I first had problems loading the drivers and was the zaptel init file that I will post below What's wrong with a simple 'modprobe qozap; modprobe wcfxo; ztcfg' ? (With no sleeps). I need even longer sleeps (TDM11B only box), if I don't use the sleeps my RedHat Enterprise Linux 4.x rebuild will fail to load the second module after zaptel. I don't know why, it seems that the second module is very impatient and also that RHEL is slow setting up the proper udev stuff. Yet another reason to cancel the automatic execution of ztcfg in modprobe.conf and just run ztcfg once. After a short sleep, if you must. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] snom Firmware 5.0.
Title: snom Firmware 5.0. Hi, Snom phones firmware 5.0 is now out. Try it if you like: http://www.snom.com/wiki/index.php/Main_Page. Regards, - Usman Tahir snom technology AG www.snom.com - ___ --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] Anyone doing NAT through m0n0Wall?
Hi Folks, I've just built myself a m0n0Wall based around a WRAP board and whilst it work really well for everything else I'm having some issues with Asterisk's NAT abilities. Here's my setup, A bunch of hardphones (various types) littered around the house. SPA-3000 handles the house POTS line which forwards to extention 2005. X-Ten Pro on my laptop for when I'm out and about. Grandstream BT-101 at my dad's house via our cable modems. Until replacing the Linksys with the m0n0Wall everything was working fine and dandy. I have externip=g7ltt.dyndns.org set in my sip.conf file. Without it I could not make my dad's phone work. With the m0n0Wall in place and the externip setting set I can make no calls internally but all the external phones work just fine. The reverse is true when I remove the externip setting; the internal phones work but the external ones don't. I've done some tracing with both firewalls and have noted the following; Linksys: externip set all SIP and IAX2 frames from * have my public address as the reply-to regardless of the NAT requirement of the phone in use. In other words it offers up the external address for internal calls. All data flows through the Linksys when addressed to the public IP address and is then forwarded back to the * server. m0n0Wall: externip set as above and the firewall drops the packets. externip not set and the * NAT doesn't work. I know that the m0n0Wall requires a rule to be added to make it work as before but what I don't understand is why is Asterisk forcing all calls to use its public IP address when externip is set? Surely this doubles network traffic; one packet goes to the router. another goes from the router to the internal host. Why doesn't go directly over the LAN for internal stuff? I had assumed that the addition of a nat=yes statement in the relevant phone stanza would turn on or off the NAT reqirement for that phone device but this doesn't seem to be the case. Any ideas would be greatly appreciated. Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --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] DTMF - FSK CallerID problems
Dear List, First, Seasons Greetings and all the best for the coming year. I hope this telecommunication revolution that is Asterisk, and which we are all involved gains its deserved position. I have a installation of an * in Brazil, and as you might know, we have a weird DTMF CallerID in the analog side that sends before the ring a DTMF String starting with A and ending in C without any warning at all (no polarity, ring, nothing). This has been discussed not only here in this list, but also in the local Brazilian lists, and no solution has yet been implemented. Hence, the option we have is to install a converter that takes this before-ring DTMF stream and converts it to the Bellcore FSK standard. Since the process is that the DTMF stream is immediately followed by the first ring, the converter is able to capture the DTMF stream and convert it into FSK after the first ring. Now, I have * 1.0.9 installed from [EMAIL PROTECTED] 1.5, but as I got better acquainted with * I dwelled more and into the configs directly, but the core [EMAIL PROTECTED] configs are still there. After installing the converter the setup still fails to get the CID, and there is not even a peep in the full log of asterisk about CID success/errors after the Simple Switch is started. One doubt I have is that as * answers the line only after the second ring, is it missing the CID? Any other ideas? I even went into chan_zap and read it all over (phew!!) but since my C knowledge is inexistent, I doubt I have understood it all, but as far as I gathered, the Simple Switch (ss_thread?) where the CID detection is, starts only after the second ring, long after the FSK has been transmitted. Is this correct? Man, I am at a loss, and apologize already, for I have this foreboding feeling that something basic is missing, but after three days of scouring, still cannot pinpoint it. Thank you for all your time! Cheers! oZ PS, below is my Zapata.conf. [channels] language=us context=from-pstn signalling=fxs_ks ;rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; usedistinctiveringdetection=no cidsignalling=bell cidstart=ring usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no jitterbuffers=12 callprogress=yes busydetect=yes ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;callerid=asreceived ; Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 signalling=fxo_ks ; Note: this is an extension. Create a ZAP extension in AMP for Channel 1 context=from-internal group=1 channel = 1 signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 2 context=from-pstn group=0 channel = 2 signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 3 context=from-pstn group=0 channel = 3 signalling=fxs_ks ; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 4 context=from-pstn group=0 channel = 4 ; Span 2: WCTDM/1 Wildcard TDM400P REV I Board 2 signalling=fxo_ks ; Note: this is an extension. Create a ZAP extension in AMP for Channel 5 context=from-internal group=1 channel = 5 signalling=fxo_ks ; Note: this is an extension. Create a ZAP extension in AMP for Channel 6 context=from-internal group=1 channel = 6 signalling=fxo_ks ; Note: this is an extension. Create a ZAP extension in AMP for Channel 7 context=from-internal group=1 channel = 7 signalling=fxo_ks ; Note: this is an extension. Create a ZAP extension in AMP for Channel 8 context=from-internal group=1 channel = 8 ;[204] signalling=fxo_ks record_out=On-Demand record_in=On-Demand [EMAIL PROTECTED] echotraining=800 echocancelwhenbridge=no echocancel=yes context=from-internal callprogress=no callerid=Ossi Sariola 204 busydetect=no busycount=7 channel=8 ___ --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] IAX No Authority found
I've run into this in the past, and it's usually been a context mismatch issue. Could you post the relevant portions of extensions.conf?On 12/22/05, Leandro Martini - ISAT DGL
[EMAIL PROTECTED] wrote:
Guys,I,m facing a little tricky issue here, is there anybody that faced the sameissue or knows how to solve this?I have 2 *, trunked with IAXFrom ServerA I can call ServerB without any problems
If I call from ServerB to ServerA i get the following message :ServerBDec 21 11:12:23 WARNING[2849]: chan_iax2.c:6967 socket_read: Call rejectedby 10.0.100.125
: No authority found-- Hungup 'IAX2/campinas-16384'== No one is available to answer at this time (1:0/0/0)== Auto fallthrough, channel 'IAX2/8512-7' status is 'NOANSWER'-- Hungup 'IAX2/8512-7'
ServerADec 21 13:08:12 NOTICE[2420]: chan_iax2.c:6772 socket_read: Rejected connectattempt from 10.20.0.20, who was trying to reach '[EMAIL PROTECTED]'The iasx.conf is as follows:
serverB[campinas]qualify=yestype=friendauth=rsa;username=campinas;secret=campinasvoiphost=10.0.100.125trunk=yesnotransfer=yesdisallow=all
allow=speexserverA[saopaulo]qualify=yestype=friendauth=rsa;username=saopaulo;secret=saopaulovoiphost=10.20.0.20trunk=yesnotransfer=yes
The Dial string is this one:On serverAexten =_85XX,1,Dial(IAX2/saopaulo/${EXTEN},60,t)On serverBexten = _74XX,1,Dial(IAX2/campinas/${EXTEN},60,t)Is there anything missing ???
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Re: [Asterisk-Users] IAX No Authority found
On Thursday 22 December 2005 13:40, Leandro Martini - ISAT DGL wrote:
Guys,
I,m facing a little tricky issue here, is there anybody that faced the same
issue or knows how to solve this?
I have 2 *, trunked with IAX
From ServerA I can call ServerB without any problems
If I call from ServerB to ServerA i get the following message :
ServerB
Dec 21 11:12:23 WARNING[2849]: chan_iax2.c:6967 socket_read: Call rejected
by 10.0.100.125: No authority found
-- Hungup 'IAX2/campinas-16384'
== No one is available to answer at this time (1:0/0/0)
== Auto fallthrough, channel 'IAX2/8512-7' status is 'NOANSWER'
-- Hungup 'IAX2/8512-7'
ServerA
Dec 21 13:08:12 NOTICE[2420]: chan_iax2.c:6772 socket_read: Rejected
connect attempt from 10.20.0.20, who was trying to reach '[EMAIL PROTECTED]'
The iasx.conf is as follows:
serverB
[campinas]
qualify=yes
type=friend
auth=rsa
If you use auth=rsa
then must show where is it:
inkeys=campinasOrwhatever
benchev
;username=campinas
;secret=campinasvoip
host=10.0.100.125
trunk=yes
notransfer=yes
disallow=all
allow=speex
serverA
[saopaulo]
qualify=yes
type=friend
auth=rsa
;username=saopaulo
;secret=saopaulovoip
host=10.20.0.20
trunk=yes
notransfer=yes
The Dial string is this one:
On serverA
exten = _85XX,1,Dial(IAX2/saopaulo/${EXTEN},60,t)
On serverB
exten = _74XX,1,Dial(IAX2/campinas/${EXTEN},60,t)
Is there anything missing ???
Happy holidays to you all !!!
Leandro Martini
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[Asterisk-Users] need help in building dynamic conference
hi all, can any one helpme in how to invite a user(exisiting person) to an already started conference, by using meetme app. in asterisk. hope every got what i mean. with regards asteriskuser ___ --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] agentcallbacklogin
Hi, On of my agents made a mistake while logging in to the Queue system, and entered another agent's extension. Asterisk allowed that, and the first agent was then able to receive two calls from the queue, on that was actually for him, and the other one that was on behalf of the agent that made the mistake. Shouldn't Asterisk block the second agent in case he tries to login using an extension that is already in use by other agent? Thank you Dov ___ --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: Re: How to record a call
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... http://www.voip-info.org/wiki-Asterisk+config+features.conf BTW: Please let me know when you've got this working 100%... I keep having issues with it! Most notably when dialling OUTBOUND with IAX softphone (tried borg DIAX and IDEFISK) I have tried it with softphones but it didn't work. Now I have three diferent SIP phones and Digium TDM22P card. So I can try diferent sort of combinations. Hopefully one of them will work :)) -- Tomislav Parcina [EMAIL PROTECTED] ___ --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] unplugging E1 cables while asterisk running
unplugging PRI cables generates major alarms on ALL telecom systems... in our lab, we do that every day and don't have to restart anything... On Thu, 2005-12-22 at 11:51 +0100, Simone Cittadini wrote: Yesterday I've had to unplug one cable coming from a TE410 card to plug it in another hole, due to provider's changes in the patch panel. The calls on that span stopped working (can't create zap channel), the problem was solved restarting asterisk. Note that the PRI termination hasn't changed, only moved the cables connecting the card to it from one patch panel to another. The cable's guy told me that unplugging and quickly replugging E1 cables isn't a problem on traditional systems, anyone konws the reason or if it is a bug that should be reported ? ___ --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] anybody getting No authority found with teliax now?
Everything was working great until last night. All calls since last night are getting No Authority Found message. I am using IAX2 Is anybody else having this problem? Thx, Tom __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] anybody getting No authority found with teliaxnow?
This is an authentication problem. Check the username, password, number and context being sent across to see if they are correct. Post your iax debug info for the call if you can. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Thomas Miller Sent: Thursday, December 22, 2005 8:58 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] anybody getting No authority found with teliaxnow? Everything was working great until last night. All calls since last night are getting No Authority Found message. I am using IAX2 Is anybody else having this problem? Thx, Tom __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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
Re: [Asterisk-Users] PRI problems: B-Channel restart
Adding this to zapata.conf may help resetinterval=never Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 Antoine Megalla wrote: I have a TE205P card with sapn 1 connected to the TELCO and span 2 connected to a PBX, The 2 spans have the following configuration: span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 The problem is that I get periodic B-Channel restart on both spans (altough it is much more frequent on span 1 connected to the telco) during calls which result in call drops --B-channel 0/1 successfully restarted on span1 --B-channel 1/1 successfully restarted on span1 --B-channel 2/1 successfully restarted on span1 . . . --B-channel 31/1 successfully restarted on span1 The channel restarts are less frequent when I restart Asterisk and then grow in frequency. Does anyone have an idea how to solve this problem? Thanks, Antoine Megalla. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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
Re: [Asterisk-Users] Re: Re: How to record a call
I'm running AAH 2.2 and *1 works from my eyebeam sip phones to do on demand recording. You need to set the DIAL_OPTIONS of wW in order to utilize this feature. lower case w means called person can initiate, upper case means callee can initiate, I think that is the order. They show up as auto-timestamp-src-dst.wav in /var/spool/asterisk/monitor However, they will NOT show up in ARI, I modified the code to show them and sent the modification to Dan to implement if he chooses. -BlakeOn 12/22/05, Tomislav Parcina [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED],[EMAIL PROTECTED] says... http://www.voip-info.org/wiki-Asterisk+config+features.conf BTW: Please let me know when you've got this working 100%... I keep having issues with it! Most notably when dialling OUTBOUND with IAX softphone (tried borg DIAX and IDEFISK)I have tried it with softphones but it didn't work. Now I have threediferent SIP phones and Digium TDM22P card. So I can try diferent sort of combinations. Hopefully one of them will work :))--Tomislav Parcina[EMAIL PROTECTED]___ --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: Can't pass variables using Originate in PHPAGI 2.14
What version of Asterisk are you using? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] chan_oss.so
What does this channel do? Today I installed * 1.2.1 for the first time and I needed to put noload = chan_oss.so in modules section of modules.conf file. Will I miss some Asterisk functionality now? -- Tomislav Parcina [EMAIL PROTECTED] ___ --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: Can't pass variables using Originate in PHPAGI 2.14
I am using Asterisk 1.20. -Original Message- From: Matt Riddell [mailto:[EMAIL PROTECTED] Sent: Thursday, December 22, 2005 9:41 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RE: Can't pass variables using Originate in PHPAGI 2.14 What version of Asterisk are you using? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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: Re: How to record a call
On Thu, December 22, 2005 15:27, Blake Krone said: I'm running AAH 2.2 and *1 works from my eyebeam sip phones to do on demand recording. Like I said SIP phones are next on the list to try! ;-) You need to set the DIAL_OPTIONS of wW in order to utilize this feature. lower case w means called person can initiate, upper case means callee can initiate, I think that is the order. Changed DIAL_OPTIONS in the database to read 'tTrwW' They show up as auto-timestamp-src-dst.wav in /var/spool/asterisk/monitor However, they will NOT show up in ARI, I modified the code to show them and sent the modification to Dan to implement if he chooses. -Blake Could you send me (off-list) the diff to look at? I am using AAH2.2 as well ;-) On 12/22/05, Tomislav Parcina [EMAIL PROTECTED] wrote: TIA! -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --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] Weird Echo Problem
Just wondering if someone can explain this scenario: phone--iaxy--interweb--(asterisk)--IAX2trunk--(asterisk)--SNOM360(SIP) calls work like a charm. The iaxy user (in canada) hears very clear audio, no hiss drops or echos. The snom user (me in uk) has fine audio from the iaxy but always hears an echo of himself. This is only heard on this call, all other calls are fine. So where is the echo coming from?? Thanks Bails ___ --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] asterisk AVM C2 again
Hi all, finally i think i find where is my trouble :-)
i can't get the card fonctionning in P2P mode, actually i work in p2mp
but i have only 2 Bchan. I think i forgot something (or definitly AVM C2
does not work whith *)
* 1.0.9
chan_capi_cm 0.5.4
the dial sttring for outbound call:
_0X.,4,Dial(CAPI/g1/b${EXTEN}/bo)
default extension for inbouond call:
exten = s,1,Answer
exten = s,2,Dial(${INCOMING},30,tT)
/etc/capi.conf (if i put P2P, i cant have inbound call):
c2 c2.bin DSS1- - - -
c2 - DSS1- - - -
/etc/asterisk/capi.conf
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
isdnmode=msn
incomingmsn=*
controller=1
group=1
softdtmf=1
accountcode=
context=capi-in
devices=2
isdnmode=msn
incomingmsn=*
controller=2
group=1
softdtmf=1
accountcode=
context=capi-in
devices=2
--
Stephane Plichon | HASGARD
~
___
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RE: [Asterisk-Users] Anyone doing NAT through m0n0Wall?
I am. Setup exactly as you describe, in a corporate environment. No problem whatsoever. Do you have port forwarding rules to your Asterisk server from the WAN interface specifically for 5060 UDP and RTP 1-2? Also Monowall 1.2 was flaky for me, I'm running 1.1 -Original Message- From: Mark Phillips [mailto:[EMAIL PROTECTED] Sent: Thursday, December 22, 2005 5:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Anyone doing NAT through m0n0Wall? Hi Folks, I've just built myself a m0n0Wall based around a WRAP board and whilst it work really well for everything else I'm having some issues with Asterisk's NAT abilities. Here's my setup, A bunch of hardphones (various types) littered around the house. SPA-3000 handles the house POTS line which forwards to extention 2005. X-Ten Pro on my laptop for when I'm out and about. Grandstream BT-101 at my dad's house via our cable modems. Until replacing the Linksys with the m0n0Wall everything was working fine and dandy. I have externip=g7ltt.dyndns.org set in my sip.conf file. Without it I could not make my dad's phone work. With the m0n0Wall in place and the externip setting set I can make no calls internally but all the external phones work just fine. The reverse is true when I remove the externip setting; the internal phones work but the external ones don't. I've done some tracing with both firewalls and have noted the following; Linksys: externip set all SIP and IAX2 frames from * have my public address as the reply-to regardless of the NAT requirement of the phone in use. In other words it offers up the external address for internal calls. All data flows through the Linksys when addressed to the public IP address and is then forwarded back to the * server. m0n0Wall: externip set as above and the firewall drops the packets. externip not set and the * NAT doesn't work. I know that the m0n0Wall requires a rule to be added to make it work as before but what I don't understand is why is Asterisk forcing all calls to use its public IP address when externip is set? Surely this doubles network traffic; one packet goes to the router. another goes from the router to the internal host. Why doesn't go directly over the LAN for internal stuff? I had assumed that the addition of a nat=yes statement in the relevant phone stanza would turn on or off the NAT reqirement for that phone device but this doesn't seem to be the case. Any ideas would be greatly appreciated. Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --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
Re: [Asterisk-Users] Weird Echo Problem
bails wrote: The snom user (me in uk) has fine audio from the iaxy but always hears an echo of himself. This is only heard on this call, all other calls are fine. It's probably coming from the analog hybrid in the IAXy. Since there is no echo canceler in the path between you and the IAXy, any echo generated there is likely to make its way to you undisturbed :-) ___ --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] Anyone doing NAT through m0n0Wall?
Hi Colin, You should use externhost=yourhost.somethingddns.com and you should put the local network parameter in your sip.conf. This will identify that your local lan doesn't need to use the externhost parameter when you try to connect internally- and asterisk should just work fine. regards, Francis On 12/22/05, Colin Anderson [EMAIL PROTECTED] wrote: I am. Setup exactly as you describe, in a corporate environment. No problemwhatsoever. Do you have port forwarding rules to your Asterisk server from the WAN interface specifically for 5060 UDP and RTP 1-2?Also Monowall 1.2 was flaky for me, I'm running 1.1-Original Message-From: Mark Phillips [mailto: [EMAIL PROTECTED]]Sent: Thursday, December 22, 2005 5:48 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Anyone doing NAT through m0n0Wall?Hi Folks,I've just built myself a m0n0Wall based around a WRAP board and whilst it work really well for everything else I'm having some issues withAsterisk's NAT abilities.Here's my setup,A bunch of hardphones (various types) littered around the house.SPA-3000 handles the house POTS line which forwards to extention 2005. X-Ten Pro on my laptop for when I'm out and about.Grandstream BT-101 at my dad's house via our cable modems.Until replacing the Linksys with the m0n0Wall everything was workingfine and dandy.I have externip= g7ltt.dyndns.org set in my sip.conf file. Without it Icould not make my dad's phone work.With the m0n0Wall in place and the externip setting set I can make nocalls internally but all the external phones work just fine. The reverse is true when I remove the externip setting; the internal phones work butthe external ones don't.I've done some tracing with both firewalls and have noted the following;Linksys: externip set all SIP and IAX2 frames from * have my public address as the reply-to regardless of the NAT requirement of the phonein use. In other words it offers up the external address for internalcalls. All data flows through the Linksys when addressed to the public IP address and is then forwarded back to the * server.m0n0Wall: externip set as above and the firewall drops the packets.externip not set and the * NAT doesn't work.I know that the m0n0Wall requires a rule to be added to make it work as before but what I don't understand is why is Asterisk forcing all callsto use its public IP address when externip is set?Surely this doubles network traffic; one packet goes to the router.another goes from the router to the internal host. Why doesn't go directly over the LAN for internal stuff?I had assumed that the addition of a nat=yes statement in the relevantphone stanza would turn on or off the NAT reqirement for that phonedevice but this doesn't seem to be the case. Any ideas would be greatly appreciated.Mark--Mark, G7LTT/KC2ENIRandolph, NJhttp://www.g7ltt.com___ --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 -- Regards,Francis BallaresE-mail: ballares (at) gmail.comwww.VoIPware.ca ___ --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] unplugging E1 cables while asterisk running
What version are you running? In 1.0.9 and CVS HEAD of the 1.2 branch I do it all the time and I don't have to restart. On 12/22/05, Simone Cittadini [EMAIL PROTECTED] wrote: Yesterday I've had to unplug one cable coming from a TE410 card to plug it in another hole, due to provider's changes in the patch panel. The calls on that span stopped working (can't create zap channel), the problem was solved restarting asterisk. Note that the PRI termination hasn't changed, only moved the cables connecting the card to it from one patch panel to another. The cable's guy told me that unplugging and quickly replugging E1 cables isn't a problem on traditional systems, anyone konws the reason or if it is a bug that should be reported ? ___ --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
Re: [Asterisk-Users] chan_oss.so
Tomislav Parcina wrote: What does this channel do? Today I installed * 1.2.1 for the first time and I needed to put noload = chan_oss.so in modules section of modules.conf file. Will I miss some Asterisk functionality now? Just dial from the console check that your Dial cmd disappear from ur CLI. regards saul ___ --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] chan_oss.so
Tomislav Parcina wrote: What does this channel do? Today I installed * 1.2.1 for the first time and I needed to put noload = chan_oss.so in modules section of modules.conf file. Will I miss some Asterisk functionality now? It's for the Open Sound System. If you are using Alsa, you'll need to comment this out. Doug ___ --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] TDM2400
Hi all, I was checking the TDM2400 features and seems to me very interesating. I think is that I need :) I want to know your experience with this card and if you know abouts bugs, configuration and everithing thah I need to know before acquire it :) The http://www.voipsupply.com/product_info.php?products_id=1115 is necesary ? Best regards, -- Guillermo Salas M. Telconet S.A. Manta Calle 15 y Av. 24 Esq. Phone : 593 5 262 8071 Mobile: 593 9 985 5138 SIP : [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] www : http://www.telconet.net http://www.telcocarrier.net Linux User: 255902 Soporte en Linea en http://www.manta.telconet.net Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html ___ --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] TDM2400
On 12/22/05, Guillermo Salas M [EMAIL PROTECTED] wrote: Hi all, I was checking the TDM2400 features and seems to me very interesating. I think is that I need :) I want to know your experience with this card and if you know abouts bugs, configuration and everithing thah I need to know before acquire it :) The http://www.voipsupply.com/product_info.php?products_id=1115 is necesary ? We have put a number of them into production for our clients already and they are working very well. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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] Anyone doing NAT through m0n0Wall?
Thanks Francis!!! You were right on the nail with the local network parameter. I had localnet = 192.168.201.0 255.255.255.0 set rather than localnet = 192.168.201.0/255.255.255.0 All is working as it should! Thanks for all the responses. Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com Francis Ballares (VoIPware.ca) wrote: Hi Colin, You should use externhost=yourhost.somethingddns.com http://yourhost.somethingddns.com and you should put the *local network parameter *in your sip.conf. This will identify that your local lan doesn't need to use the externhost parameter when you try to connect internally- and asterisk should just work fine. regards, Francis On 12/22/05, *Colin Anderson* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I am. Setup exactly as you describe, in a corporate environment. No problem whatsoever. Do you have port forwarding rules to your Asterisk server from the WAN interface specifically for 5060 UDP and RTP 1-2? Also Monowall 1.2 was flaky for me, I'm running 1.1 -Original Message- From: Mark Phillips [mailto: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] Sent: Thursday, December 22, 2005 5:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Anyone doing NAT through m0n0Wall? Hi Folks, I've just built myself a m0n0Wall based around a WRAP board and whilst it work really well for everything else I'm having some issues with Asterisk's NAT abilities. Here's my setup, A bunch of hardphones (various types) littered around the house. SPA-3000 handles the house POTS line which forwards to extention 2005. X-Ten Pro on my laptop for when I'm out and about. Grandstream BT-101 at my dad's house via our cable modems. Until replacing the Linksys with the m0n0Wall everything was working fine and dandy. I have externip= g7ltt.dyndns.org http://g7ltt.dyndns.org set in my sip.conf file. Without it I could not make my dad's phone work. With the m0n0Wall in place and the externip setting set I can make no calls internally but all the external phones work just fine. The reverse is true when I remove the externip setting; the internal phones work but the external ones don't. I've done some tracing with both firewalls and have noted the following; Linksys: externip set all SIP and IAX2 frames from * have my public address as the reply-to regardless of the NAT requirement of the phone in use. In other words it offers up the external address for internal calls. All data flows through the Linksys when addressed to the public IP address and is then forwarded back to the * server. m0n0Wall: externip set as above and the firewall drops the packets. externip not set and the * NAT doesn't work. I know that the m0n0Wall requires a rule to be added to make it work as before but what I don't understand is why is Asterisk forcing all calls to use its public IP address when externip is set? Surely this doubles network traffic; one packet goes to the router. another goes from the router to the internal host. Why doesn't go directly over the LAN for internal stuff? I had assumed that the addition of a nat=yes statement in the relevant phone stanza would turn on or off the NAT reqirement for that phone device but this doesn't seem to be the case. Any ideas would be greatly appreciated. Mark -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ --Bandwidth and Colocation provided by Easynews.com http://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 http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Francis Ballares E-mail: ballares (at) gmail.com http://gmail.com www.VoIPware.ca http://www.VoIPware.ca ___ --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
Re: [Asterisk-Users] agentcallbacklogin
On 12/22/05, Dov Bigio [EMAIL PROTECTED] wrote: Hi, On of my agents made a mistake while logging in to the Queue system, and entered another agent's extension. Asterisk allowed that, and the first agent was then able to receive two calls from the queue, on that was actually for him, and the other one that was on behalf of the agent that made the mistake. Shouldn't Asterisk block the second agent in case he tries to login using an extension that is already in use by other agent? This is a good suggestion. I'd recommend that you post a feature request to bugs.digium.com so that it might become an option to protect from this happening. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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] Weird Echo Problem
The analog phone on the iaxy? Tried a different phone? -Original Message- From: bails [mailto:[EMAIL PROTECTED] Sent: Thursday, December 22, 2005 9:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Weird Echo Problem Just wondering if someone can explain this scenario: phone--iaxy--interweb--(asterisk)--IAX2trunk--(asterisk)--SNOM360(SIP) calls work like a charm. The iaxy user (in canada) hears very clear audio, no hiss drops or echos. The snom user (me in uk) has fine audio from the iaxy but always hears an echo of himself. This is only heard on this call, all other calls are fine. So where is the echo coming from?? Thanks Bails ___ --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] recording queue calls
Hi, When I set "monitor-format=wav49" on file queues.conf for a queue, Asterisk records callsat /var/spool/asterisk/monitor. But the file names it users are the call-ids of the calls. Is there a way to change that, and use information such as date, time, agent and queue to "build" the filename? It would make the localization of such files much more easy. Other useful that I miss is the capability to to allow the files to be stored in different directories, such as /var/spool/asterisk/monitor/queue1, /var/spool/asterisk/monitor/queue2, and so on, based on the queuename. Is this possible by any means? Thank you Dov ___ --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] 1.2.0 PRI dropping calls occasionally...
Did you ever solve this? I have noticed this also. I made a few changes at the same time I moved to 1.2.0. One being that I switched providers and thought it may be them. Please let me know. Thanks! ___ --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] Realtime Config
I'm a little confused about something with Realtime. It isn't clear to me what order Asterisk prefers to read the config. If we are using realtime, do we have to completely throw away the use of the .conf files? Sometimes not it appears. Extensions.conf lets you have a switch command to call into Realtime. For other conf files, you can use the realtime static table to load the general sections, or can you? I guess this question doesn't make much sense because the docs don't make much sense to me. My preference is to have static stuff in the config files and have dynamic portions, ie bits that might change, in realtime. Doug. ___ --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: Realtime Config
Actually, this is weird too... I have in my res_mysql.conf file: queues = mysql,voxdb,ast_queues queue_members = mysql,voxdb,ast_queue_members and I an connecting to realtime. I removed the agents.conf file. Upon load, Asterisk reports: Dec 22 09:56:40 NOTICE[18475]: chan_agent.c:1033 read_agent_config: No agent configuration found -- agent support disabled I run a network trace when I try to call AgentCallbackLogin and Asterisk isn't even querying the database. So, it appears that with extconfig configured, and no agents.conf file, Asterisk just disables the feature. What am I missing? Thanks, Doug -Original Message- From: Douglas Garstang Sent: Thursday, December 22, 2005 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Realtime Config I'm a little confused about something with Realtime. It isn't clear to me what order Asterisk prefers to read the config. If we are using realtime, do we have to completely throw away the use of the .conf files? Sometimes not it appears. Extensions.conf lets you have a switch command to call into Realtime. For other conf files, you can use the realtime static table to load the general sections, or can you? I guess this question doesn't make much sense because the docs don't make much sense to me. My preference is to have static stuff in the config files and have dynamic portions, ie bits that might change, in realtime. Doug. ___ --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] SIP - SIP bridge dropping calls?
In addition to having this with my SIP phones, I have also experienced it with SCCP. It started when I updated to the 1.2 release of asterisk. At the time I updated I also switched VoIP providers and thought it was them. Did you file this as a bug or find a solution to it? Thanks! ___ --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] Realtime Config
As far as I can tell with our systems, the config files are read first, then the realtime db. We've got a few static servers that never change, so I hardcode those in case something goes wrong with the DB, and the DB contains any other configurations that will be dynamic. I'm not sure if realtime has any support for the basic general information at the top of the config files, so I think you need to have the files to convey that information. Aaron Douglas Garstang wrote: I'm a little confused about something with Realtime. It isn't clear to me what order Asterisk prefers to read the config. If we are using realtime, do we have to completely throw away the use of the .conf files? Sometimes not it appears. Extensions.conf lets you have a switch command to call into Realtime. For other conf files, you can use the realtime static table to load the general sections, or can you? I guess this question doesn't make much sense because the docs don't make much sense to me. My preference is to have static stuff in the config files and have dynamic portions, ie bits that might change, in realtime. Doug. ___ --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
RE: [Asterisk-Users] Realtime Config
Hi Aaron. Well, there's 'realtime static' which it supposedly uses. It's table structure is: CREATE TABLE `ast_config` ( `id` int(11) NOT NULL auto_increment, `cat_metric` int(11) NOT NULL default '0', `var_metric` int(11) NOT NULL default '0', `commented` int(11) NOT NULL default '0', `filename` varchar(128) NOT NULL default '', `category` varchar(128) NOT NULL default 'default', `var_name` varchar(128) NOT NULL default '', `var_val` varchar(128) NOT NULL default '', PRIMARY KEY (`id`), KEY `filename_comment` (`filename`,`commented`) ) TYPE=MyISAM; and you can use it to store information in the [general] section and so on. I know this works because I've used it before. It just isn't clear if all the config files use it or not. Doug. -Original Message- From: Aaron Daniel [mailto:[EMAIL PROTECTED] Sent: Thursday, December 22, 2005 9:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime Config As far as I can tell with our systems, the config files are read first, then the realtime db. We've got a few static servers that never change, so I hardcode those in case something goes wrong with the DB, and the DB contains any other configurations that will be dynamic. I'm not sure if realtime has any support for the basic general information at the top of the config files, so I think you need to have the files to convey that information. Aaron Douglas Garstang wrote: I'm a little confused about something with Realtime. It isn't clear to me what order Asterisk prefers to read the config. If we are using realtime, do we have to completely throw away the use of the .conf files? Sometimes not it appears. Extensions.conf lets you have a switch command to call into Realtime. For other conf files, you can use the realtime static table to load the general sections, or can you? I guess this question doesn't make much sense because the docs don't make much sense to me. My preference is to have static stuff in the config files and have dynamic portions, ie bits that might change, in realtime. Doug. ___ --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 ___ --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] Zap Error
I recently installed a 1 port Fxo card It detected the card when it was booting the Zaptel hardware was being detected upon bootup. I did a yum on Centos and then did a rebuild And then did an autoconfigure everything was working fine. Now when I reboot the zaptel is not coming on-line. Any suggestions. ___ --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] TDM2400
Guillermo Salas M ha scritto: Hi all, I was checking the TDM2400 features and seems to me very interesating. I think is that I need :) I want to know your experience with this card and if you know abouts bugs, configuration and everithing thah I need to know before acquire it :) The http://www.voipsupply.com/product_info.php?products_id=1115 is necesary ? Best regards, Works perfectly out of the box, almost for my customers :-) The only note is to disable echo training. ___ --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] Make Asterisk explicitly UNREGISTER from a SIP service?
I have an Asterisk box registered with an external SIP provider. I want to make a different Asterisk box take over that registration. I removed the register statement from the sip.conf on the first box, did a sip reload, then enabled the register statement in the sip.conf on the second box, followed by a sip reload on there. The second box is getting a 403 Login limit exceeded from the SIP provider, presumably because the original registration is still active. How can I make the first Asterisk explicitly unregister? Failing that, I'm hoping that the original registration at the provider times out, although it hasn't done so yet, an hour after the change. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Manager API connections - crashes?
Is 1.2.0/1 still having problems with crashes due to having too many connections to the manager api or has that been solved? If it is, does anyone know roughly how many connections cause the crash or is it seemingly random ___ --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] how to follow a call in the console
Hi Is there a way to easily follow a call in the console log Maybe by adding a unique call ID or something Thanks Patrick ___ --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] asterisk AVM C2 again
On Thu, 22 Dec 2005, stéphane plichon wrote:
Hi all, finally i think i find where is my trouble :-)
i can't get the card fonctionning in P2P mode, actually i work in p2mp
but i have only 2 Bchan. I think i forgot something (or definitly AVM C2
does not work whith *)
* 1.0.9
chan_capi_cm 0.5.4
the dial sttring for outbound call:
_0X.,4,Dial(CAPI/g1/b${EXTEN}/bo)
This dial string does not make sense. With version 0.5.4 you use
_0X.,4,Dial(CAPI/g1/b${EXTEN})
but for 0.6.x
_0X.,4,Dial(CAPI/g1/${EXTEN}/bo)
/etc/asterisk/capi.conf
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
isdnmode=msn
incomingmsn=*
controller=1
group=1
softdtmf=1
accountcode=
context=capi-in
devices=2
isdnmode=msn
incomingmsn=*
controller=2
group=1
softdtmf=1
accountcode=
context=capi-in
devices=2
When using PtP, you need to set isdnmode=ptp (or isdnmode=did when using
0.6.x)
Read the README !
Armin
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[POSSIBLE SPAM] RE: [Asterisk-Users] Identifying Frame Slips from PRI debug
Our MailScanner believes that the attachment to this message sent to you From: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Identifying Frame Slips from PRI debug is Unsolicited Commercial Email (spam). Unless you are sure that this message is incorrectly thought to be spam, please delete this message without opening it. Opening spam messages might allow the spammer to verify your email address. If you believe that this message has been incorrectly marked as spam, please forward this email to postmaster. pts rule name description -- -- 1.0 NO_REAL_NAME From: does not include a real name 1.1 HELO_EQ_LT4_SA HELO_EQ_LT4_SA -2.6 BAYES_00 BODY: Bayesian spam probability is 0 to 1% [score: 0.] 3.9 RCVD_IN_XBLRBL: Received via a relay in Spamhaus XBL [208.54.94.73 listed in sbl-xbl.spamhaus.org] ---BeginMessage--- Can someone help me understand how to identify frame slips from pri debug, pri intense debug, or any other method? I am familiar with zttest, avoiding interrupt sharing, mucking with ACPI, making sure DMA is on, etc... I have a list of changes I intend to make to the server(s), but don't know how to actually watch the output to see if my changes are affecting the # of frame slips. First things first; Check your timing param in zaptel.conf, you almost certainly want it set to 1 if you are connecting that span to a phone company and it is the only such connection on that board. The timing is already set to 1 for each of the cards. Ah, worth asking. Also, just in case, have you powercycled those boxes since you set the timing source ? Yes. These boxes were all set correctly during their original install, and have been power cycled several times since then. I had no end of trouble 'till I noticed that a cold-start seemed to be the only thing that ensured that both ends of the link _knew_ that I'd changed the timing settings. I've mentioned this on-list a while back and got mixed reactions, some folk felt cold start is overkill and a mod/load/unload is enough, but everyone agrees that an asterisk reload isn't enough. The only other thing I can advise is that you look at running the patch that Matt (Fredrikson?) from Digium put out a month or so back, it allows the low level HDLC protocol to run on the chip, not in the kernel. I haven't looked at it as I don't think it is supported on the older cards. I am aware of this patch (5313), but it appears to only work for TE4XX/TE2XX cards. Since the two cards I am using are T100P and TE110P I don't think it is currently an option. Eric ---End Message--- ___ --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: how to follow a call in the console
Just to back up your need, I posted this on Nov.1, but didn't get any replies. ref: Is there a way to get a thread ID (???) in the log file? I see the process ID, but I think some way to correlate which items are tied to which items would be helpful for troubleshooting. When there are multiple simultaneous calls going on, It takes a lot more effort to correlate hang-ups and errors. (etc.) If I could grep for a phone number in the log, get an ID tied to that thread (???) , then grep for that ID, I could see only what I want to see much faster. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Patrick Fortin [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi Is there a way to easily follow a call in the console log Maybe by adding a unique call ID or something Thanks Patrick ___ --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] Asterfax beta4, Asterisk 1.2.0 and issue sending FAX
Hi,
I have Fedora Core 4, Asterisk 1.2.0, SpandSp0.0.2 e Asterfax beta 4.
When I try to send a FAX the remote part respond and then the line hangs up.
The error here is TIFFOpen: ${FAXFILE}: Cannot open. But I don't
understand why:
1 Message type: CONNECT (7)
1 COLP (len=13) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony
Numbering Plan (E.164/E.163) (1)
1Presentation: Presentation allowed of network
provided number (3) '956545646' ]
1 -- Processing IE 76 (cs0, Connect Line ID Presentation)
1 Protocol Discriminator: Q.931 (8) len=4
1 Call Ref: len= 1 (reference 3/0x3) (Originator)
1 Message type: CONNECT ACKNOWLEDGE (15)
Channel Zap/1-1 was answered.
Launching txfax(${FAXFILE}|caller) on Zap/1-1
Dec 22 20:02:04 NOTICE[3223]: channel.c:2450 __ast_request_and_dial: Unable
to request channel ZAP/1/0957270289
TIFFOpen: ${FAXFILE}: Cannot open.
1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect
Request
1 Protocol Discriminator: Q.931 (8) len=8
1 Call Ref: len= 1 (reference 3/0x3) (Originator)
1 Message type: DISCONNECT (69)
regards
Gianrico Fichera
Itesys srl
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[Asterisk-Users] CVS problem?
cvs checkout: failed to obtain dir lock in repository `/usr/cvsroot/zaptel' Anyone else seen this? ___ --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] CVS problem?
That is because they switched over to svn I belive. Darren Wiebe Colin Anderson wrote: cvs checkout: failed to obtain dir lock in repository `/usr/cvsroot/zaptel' Anyone else seen this? ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] CVS problem?
I thought they were going to run CVS concurrently for a while?? -Original Message- From: Darren Wiebe [mailto:[EMAIL PROTECTED] Sent: Thursday, December 22, 2005 11:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CVS problem? That is because they switched over to svn I belive. Darren Wiebe Colin Anderson wrote: cvs checkout: failed to obtain dir lock in repository `/usr/cvsroot/zaptel' Anyone else seen this? ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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] wav to g729
hello, how can I convert my existing wav file to g729. Currently, i have all of them converted to gsm. Isn't it right, If I had all my sound files in g729 format, my server would use less resource and less channels. I have couple of g729 liscences from digium. Thanks, -- You don't have any choice, you already made it before you came here.___ --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] CVS problem?
Colin Anderson wrote: I thought they were going to run CVS concurrently for a while?? We are; I'm trying to fix the problem right now. ___ --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] CVS problem?
Aha here it is: For the near future, we will continue to provide access to source code via CVS using the same servers/paths that you have previously been using; once every day, the relevant Subversion branches will be copied over into CVS and brought up to date. We expect to keep updating CVS HEAD this way for three to six months; the other branches will be maintained for six to nine months. However the CVS repositories will be updated in a single commit each day and will not contain any detailed revision history for the changes that are made. We encourage all users to transition to using Subversion for tracking development as soon as possible. So, anyone still using CVS have a problem checking out Zaptel? -Original Message- From: Darren Wiebe [mailto:[EMAIL PROTECTED] Sent: Thursday, December 22, 2005 11:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CVS problem? That is because they switched over to svn I belive. Darren Wiebe Colin Anderson wrote: cvs checkout: failed to obtain dir lock in repository `/usr/cvsroot/zaptel' Anyone else seen this? ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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
Re: [Asterisk-Users] anybody getting No authority found with teliaxnow?
maybe your account was disbaled due to non payment ? also check if you are sending auth in the features area.. On 12/22/05, Jonathan k. Creasy [EMAIL PROTECTED] wrote: This is an authentication problem. Check the username, password, numberand context being sent across to see if they are correct.Post your iax debug info for the call if you can.-Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED]] On Behalf Of Thomas Miller Sent: Thursday, December 22, 2005 8:58 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] anybody getting No authority found with teliaxnow? Everything was working great until last night. All calls since last night are getting No Authority Found message. I am using IAX2 Is anybody else having this problem? Thx, Tom __ Do You Yahoo!? Tired of spam?Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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 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] wav to g729
Innocent Evil wrote: hello, how can I convert my existing wav file to g729. Currently, i have all of them converted to gsm. Isn't it right, If I had all my sound files in g729 format, my server would use less resource and less channels. I have couple of g729 liscences from digium. http://www.asteriskguru.com/tools/audio_conversion.php -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] CVS problem?
You know, that's right. I thought so too. I've been entirely unsuccessful getting cvs downloads but that could just be my luck. Merry Christmas Everyone, Darren Wiebe [EMAIL PROTECTED] Colin Anderson wrote: I thought they were going to run CVS concurrently for a while?? -Original Message- From: Darren Wiebe [mailto:[EMAIL PROTECTED] Sent: Thursday, December 22, 2005 11:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CVS problem? That is because they switched over to svn I belive. Darren Wiebe Colin Anderson wrote: cvs checkout: failed to obtain dir lock in repository `/usr/cvsroot/zaptel' Anyone else seen this? ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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 -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --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: Teliax billing question
FYI 60/1 measn first 60 seoncd billed then each 1 /60th of a minute so 1minute 25 second call is billed as 1 minute 25 45 second as one minute wher the first number is minimum seconds so 6/6 is first 6 seconds no matter what then every block of 6 seconds.. 6/1 well you get the point.. its like cell carriers.. where some bill per minute others like canada fido per second.. On 12/19/05, trixter aka Bret McDanel [EMAIL PROTECTED] wrote: On Mon, 2005-12-19 at 10:13 -0800, Wolfgang S. Rupprecht wrote: from: http://www.trac.org/news/2005/tracnotes-vol-3-22.html The scam isn't new, and its certainly not limited to home 800 numbers. The same basic principles were used by some of the 900 number folks a few years ago as well. My fear wasn't that someone would stuff phony charges on my bill (like charges for 900 calls that were never made).I was more afraid of the case where someone in bad faith war-dials the 800 number so they can collect the 60-cent (???) per call payphone charge.Will VOIP providers let your dispute this charge because the calls were made in bad-faith or is this simply a grin-and-bear it type situation?That could be covered under 18 USC 1343 (wire fraud).afaik there has not been a single case that was prosecuted, and for the payphoneoperator (providing they meet the compensation requirements of the FCCrules - 13.65 comes to mind but I havent owned a payphone business since 1998 so I may not remember correctly) to make up some wild story abouthow it was a kid or something (which doesnt negate the payphoneoperators claim to compensation).An elligible payphone must beavailable for the general public to get access to it. All payments are typically made through clearinghouses as opposed toinidvidual carriers processing the billing.This makes fraud trackingslightly easier since all the calls are there.They have kept averages of total calls by a payphone to compensatable numbers, carrier averages(ie mci, sprint, att, etc) and stuff that way.If someone were to use an auto dialer to call a tollfree they violate atleast 47 CFR 64.1200 and I think a criminal statute too (I dont rememberwhere in the USC it is anymore, but there is one for that).According to the FCC rules back in 1997-98 on this matter even if fraudis suspected you must pay the payphone operator.They also talk about civil damages being sought, but that doesnt preclude criminal charges,only gives you easy rights to sue, which of course costs money and theburden of proof is then upon you. I understand that within the PSTN there is a 2-bit value associated with the class of phone that the call is placed from (normal, payphone, prison-phone).If voip/pstn gateways started passing this on it might make it easier for folks to guard against payphone scams by configuring their asterisk to only answer the 800 calls made from normal residential phones.Any reasonable provider should be able to block those calls, however ina blocking situation its all or nothing.If you have ani you can look for the same number calling over and over and reject it that way.Youshould have ani with a tollfree.The additional info is commonly not sent and afaik there is no'standard' way to send that.SIP IM might work (that is how verisign sends SS7 info in their SIP-7 product so doing something in this caseshouldnt be *too* hard but the provider has to agree to it).--Trixter http://www.0xdecafbad.com Bret McDanelUK +44 870 340 4605 Germany +49 801 777 555 3402US +1 360 207 0479 or +1 516 687 5200FreeWorldDialup: 635378http://www.sacaug.org/ Sacramento Asterisk Users Group -BEGIN PGP SIGNATURE-Version: GnuPG v1.4.2 (GNU/Linux)iD8DBQBDpvv1+1olxlzQw5cRAn1hAJ401KFBurmj1nvvc/P8hIMjFbhO4ACfSlngyOMzArCFQIRlGtwuwBqXEJg==UsIX-END PGP SIGNATURE- ___--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] CVS problem?
Colin Anderson wrote: So, anyone still using CVS have a problem checking out Zaptel? This has been corrected. One of the CVS mirrors was broken. ___ --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] TDM2400
Massimo De Nadal wrote: Works perfectly out of the box, almost for my customers :-) The only note is to disable echo training. Could you please elaborate which exact model you're using and what are your opinion about the echo can/training quality? Have you tried spandsp faxing? Thanks in advance, Vahan ___ --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] wav to g729
I prefer something 'sox' like program. -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 22 Dec 2005 19:44:36 +0100 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] wav to g729 Innocent Evil wrote: hello, how can I convert my existing wav file to g729. Currently, i have all of them converted to gsm. Isn't it right, If I had all my sound files in g729 format, my server would use less resource and less channels. I have couple of g729 liscences from digium. http://www.asteriskguru.com/tools/audio_conversion.php -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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] ast_sock_cmd: pipe commands to asterisk
If you need to pipe commands to Asterisk without resorting to the
manager interface (e.g: requiring a specific user, hard-wiring password
or other authentication messure. After all: you're root of the machine
and know better than that lousy Asterisk process), you may want to use
the following.
It is a simple C program to pipe commands to asterisk.ctl . It was
written to help me and thus hardwires the path I use
(/var/run/asterisk/asterisk.ctl ) , but I'd welcome any fixes for that.
Usage:
build with:
make ast_sock_cmd
example usage:
echo -e set verbose 3\nset debug 5 | ./ast_sock_cmd
echo -e restart now | ./ast_sock_cmd
I'd also be happy to know of existing alternatives. It looked strange I
could not find such an existing tool to pipe text into a unix-domain
socket.
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il | | a Mutt's
[EMAIL PROTECTED] | | best
ICQ# 16849755 | | friend
/**
*
* ast_sock_cmd: Pipe commands to asterisk through the asterisk control
* socket (asterisk.ctl)
*
* Written by Tzafrir Cohen [EMAIL PROTECTED]
* Copyright (C) 2005, Xorcom
*
* All rights reserved.
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
*
* Asterisk listens on that socket for commands which should be
* terminated with \r\n . The input is sent back in the same connection.
*
* This program reads commands from its standard input and sends them
* to that socket (with \r\n). It does not attempt to read the output
* from the socket and thus will not give any indication of succes or
* failure.
*
* Unlike asterisk -r it starts much faster and can be piped multiple
* commands from its standard input.
*/
#include sys/types.h
#include sys/socket.h
#include sys/un.h
#include stdio.h
#include errno.h
#define MAX_CMD_LEN 80
#define SOCK_FILE /var/run/asterisk/asterisk.ctl
// borrowed from Asterisk's ast_tryconnect
int main () {
char command[MAX_CMD_LEN+2];
struct sockaddr_un sock_addr;
int res;
int sock = socket(PF_UNIX, SOCK_STREAM, 0);
if (sock 0) {
fprintf(stderr,Failed to create socket: %s\n, strerror(errno));
exit(1);
}
memset(sock_addr, 0, sizeof(sock_addr));
sock_addr.sun_family = AF_UNIX;
strcpy(sock_addr.sun_path, SOCK_FILE);
res = connect(sock, (struct sockaddr *)sock_addr, sizeof(sock_addr));
if (res 0) {
fprintf(stderr,Failed to connect to asterisk control socket: %s\n, strerror(errno));
exit(2);
}
while (fgets(command, MAX_CMD_LEN, stdin)) {
// stand at the end of the string and terminate it with \r\n\0
char *p = strchr(command, '\n');
if (p == NULL) { p=strchr(command,'\0');}
p[0]='\r'; // TODO: check if there was already a \r ?
p[0]='\n';
p[0]='\0';
// for the sake of simpliicty: let's assume that if write wrote,
// it wrote everything:
res = write(sock, command, strlen(command));
if (res 0) {
fprintf(stderr,Failed to write to asterisk control socket: %s\n,
strerror(errno));
exit(3); // any point of close()-ing here?
}
}
close(sock);
}
___
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RE: [Asterisk-Users] CVS problem?
Yup working fine now, thanks. -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Thursday, December 22, 2005 11:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CVS problem? Colin Anderson wrote: So, anyone still using CVS have a problem checking out Zaptel? This has been corrected. One of the CVS mirrors was broken. ___ --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
RE: [Asterisk-Users] Asterisk server to provide virtuals IPBX
App_valetparking is a great (and necessary) addition to asterisk. Does app_valetparking.c work with the current release of asterisk? I tried to install it on Asterisk 1.0.9 and I get errors following the instruction in the wiki? app_valetparking.c:678: dereferencing pointer to incomplete type -Original Message- From: Olle E Johansson [mailto:[EMAIL PROTECTED] Sent: Thursday, December 22, 2005 2:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk server to provide virtuals IPBX Christopher L. Wade wrote: On Wed, Dec 21, 2005 at 10:45:29AM -0500, C F wrote: The workaround for the parking limitation is app_valetparking.so from http://www.pbxfreeware.org/app_valetparking.c instructions on how to install is on the wiki. On 12/21/05, Olle E Johansson [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: Hello, Is Asterisk able to provide virtuals IPBX ? I mean one hardware server which handle one IPBX per enterprise . A lot of service providers do that. One caveat is the parking function, that only supports one parking lot for all virtual PBXs. /O There is also a work in progress in svn to add context support to the builtin asterisk parking. I forget which developer is working on it but it should be hard to find if you check the asterisk-commits archive on lists.digium.com. That would be me :-) It is in the multiparking branch. /O ___ --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
Re: [Asterisk-Users] Asterisk server to provide virtuals IPBX
The on on pbxfreeware works with 1.2.1 On 12/22/05, Kevin Kiely [EMAIL PROTECTED] wrote: App_valetparking is a great (and necessary) addition to asterisk. Does app_valetparking.c work with the current release of asterisk? I tried to install it on Asterisk 1.0.9 and I get errors following the instruction in the wiki? app_valetparking.c:678: dereferencing pointer to incomplete type -Original Message- From: Olle E Johansson [mailto:[EMAIL PROTECTED] Sent: Thursday, December 22, 2005 2:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk server to provide virtuals IPBX Christopher L. Wade wrote: On Wed, Dec 21, 2005 at 10:45:29AM -0500, C F wrote: The workaround for the parking limitation is app_valetparking.so from http://www.pbxfreeware.org/app_valetparking.c instructions on how to install is on the wiki. On 12/21/05, Olle E Johansson [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: Hello, Is Asterisk able to provide virtuals IPBX ? I mean one hardware server which handle one IPBX per enterprise . A lot of service providers do that. One caveat is the parking function, that only supports one parking lot for all virtual PBXs. /O There is also a work in progress in svn to add context support to the builtin asterisk parking. I forget which developer is working on it but it should be hard to find if you check the asterisk-commits archive on lists.digium.com. That would be me :-) It is in the multiparking branch. /O ___ --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 ___ --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] App_valetparking
App_valetparking is a great (and necessary) addition to asterisk. Does app_valetparking.c work with the current release of asterisk? I tried to install it on Asterisk 1.0.9 and I get errors following the instruction in the wiki? app_valetparking.c:678: dereferencing pointer to incomplete type -Original Message- From: Olle E Johansson [mailto:[EMAIL PROTECTED] Sent: Thursday, December 22, 2005 2:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk server to provide virtuals IPBX Christopher L. Wade wrote: On Wed, Dec 21, 2005 at 10:45:29AM -0500, C F wrote: The workaround for the parking limitation is app_valetparking.so from http://www.pbxfreeware.org/app_valetparking.c instructions on how to install is on the wiki. On 12/21/05, Olle E Johansson [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote: Hello, Is Asterisk able to provide virtuals IPBX ? I mean one hardware server which handle one IPBX per enterprise . A lot of service providers do that. One caveat is the parking function, that only supports one parking lot for all virtual PBXs. /O There is also a work in progress in svn to add context support to the builtin asterisk parking. I forget which developer is working on it but it should be hard to find if you check the asterisk-commits archive on lists.digium.com. That would be me :-) It is in the multiparking branch. /O ___ --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
Re: [Asterisk-Users] wav to g729
Try the new conversion module from redice li ..it is greate! Miklos IPFONE TELEFONIA IP Rua Caio Graco 735 São Paulo SP IPBX - +55 11 3488-3800 http://www.ipfone.com.br [EMAIL PROTECTED] Balbus balbum intellegit - Original Message - From: Innocent Evil [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 22, 2005 5:00 PM Subject: Re: [Asterisk-Users] wav to g729 I prefer something 'sox' like program. -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 22 Dec 2005 19:44:36 +0100 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] wav to g729 Innocent Evil wrote: hello, how can I convert my existing wav file to g729. Currently, i have all of them converted to gsm. Isn't it right, If I had all my sound files in g729 format, my server would use less resource and less channels. I have couple of g729 liscences from digium. http://www.asteriskguru.com/tools/audio_conversion.php -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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 ___ --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] sill looking for a provider
ahaahah now thats something to be worry about.. that prolly coz they dont want to pay taxes and your invoices serve you as refunds/credits for irs.. BUT it is required by law to give irs that crap.. so i guess they pushing in offshore an will disappear someday with all the service and cash. When something looks suspect it usually is. Ever seen smoke without any fire ? ps'' especialy the payment part USTOMERS MAY NOT DISCLOSE USE OF OR PAYMENTS TO VOIPJET ON PERSONAL, CORPORATE, LEGAL, ACCOUNTING AND lol On 11/7/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: OOPPS!Looks like someone just broke voipjet's tosgw at adcomcorp.com gw at adcomcorp.com wrote onSat Nov 5 11:36:46 CST 2005 I tend to agree with you, my experience with Teliax has been decent,and getting better.If only I could get to them at under 20ms though, right now my latency is about 75ms whereas voipjet comes through at19ms.Greg-- https://www.voipjet.com/tos.phpNON-DISCLOSURE: ALL CUSTOMERS USING VOIPJET'S SERVICE ARE SPECIFICALLY PROHIBITED FROM DISCLOSING TO OTHERS THAT THEY USE VOIPJET'S SERVICE, THIS INCLUDES BUT IS NOT LIMITED TO, END USERS. CUSTOMERS MAY NOT DISCLOSE USE OF OR PAYMENTS TO VOIPJET ON PERSONAL, CORPORATE, LEGAL, ACCOUNTING AND OTHER DOCUMENTS AND COMMUNICATIONS UNLESS SPECIFICALLY REQUIRED TO DO SO BY LAWHas anyone else read these TOS'es???Some are pretty funny.Thomas HerlihyScaletta Moloney ArmoringChicago, IL USA708.924.0099Skype VoIP @ HerlsOne Free World Dialup 647717[EMAIL PROTECTED]www.scaletta.com___--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 provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_sip.c error message
Has anyone seen this? If anyone out there has, please share your experience. I recently installed Asterisk-1.2.1 and I am getting the following error message: Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor Dec 22 15:27:21 WARNING[3315]: chan_sip.c:11046 sipsock_read: Recv error: Bad file descriptor ___ --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] forwarding a caller to a conference room
Hi, I have created a conference # 400 in meetme.conf. Now i have two extension 191 and 200. When 191 calls 200, i want to redirect the call to the conference 400 and join 200 there too so that 191 and 200 can communicate in that conference room. Please let me know how can i do that. Any pointer will be appreciated. Thanks Anup ___ --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] wav to g729
Where can I find it. Thanks, -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 22 Dec 2005 17:20:39 -0200 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] wav to g729 Try the new conversion module from redice li ..it is greate! Miklos IPFONE TELEFONIA IP Rua Caio Graco 735 São Paulo SP IPBX - +55 11 3488-3800 http://www.ipfone.com.br [EMAIL PROTECTED] Balbus balbum intellegit - Original Message - From: Innocent Evil [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 22, 2005 5:00 PM Subject: Re: [Asterisk-Users] wav to g729 I prefer something 'sox' like program. -- You don't have any choice, you already made it before you came here. -Original Message- From: [EMAIL PROTECTED] Sent: Thu, 22 Dec 2005 19:44:36 +0100 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] wav to g729 Innocent Evil wrote: hello, how can I convert my existing wav file to g729. Currently, i have all of them converted to gsm. Isn't it right, If I had all my sound files in g729 format, my server would use less resource and less channels. I have couple of g729 liscences from digium. http://www.asteriskguru.com/tools/audio_conversion.php -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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 ___ --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] Fwd: Legacy PBX - * - Voip Calls problems
If have installed a TE110P and have connected it to my Mitel 200SX. I can dial to the Mitel via the T1 connection but when I dial from the Mitel to try and go out the Asterisk box via Voip it fails. I can see the calls getting to the Asterisk box from the Mitel but it just loops though its Zap channels then fails. Do I have spilt incoming and out going channels on a T1? Thanks, -Scott Scott, Sorry for approaching you personally but not sure you are still subscribed to the list (copy to the list anyway) I have found your efforts about Legacy PBX - * - Voip Calls problems goggling but not a full thread. Your scheme is very intriguing since I intend to interconnect an *+TE110P -(PRI)-Siemens HiPath3750. Asterisk+TE110P will provide only VOIP dialing to the Siemens HiPath3750 members(about 400). Mitel is more more IP oriented than Siemens but any info you could point me to, or share would be of great value. Thanks and Merry Xmas everyone, benchev --- ___ --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] wav to g729
Innocent Evil wrote: I prefer something 'sox' like program. -- You don't have any choice, you already made it before you came here. redice.krisk.org Do it from Asterisk! -- Kristian Kielhofner ___ --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] CALLED NUMBER in IAX2
I am trying to determine the number that was called in via an IAX2 channel. When using debug: IAX2 Debugging Enabled Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00013ms SCall: 00814 DCall: 0 [66.234.228.170:4569] VERSION : 2 CALLED NUMBER : 609XXX CALLING NUMBER : 347XXX CALLING NAME: 347XX LANGUAGE: en USERNAME: voicepulse-in-01 FORMAT : 4 CAPABILITY : 1086 ADSICPE : 2 DATE TIME : 194420173 What is the variable that CALLED NUMBER is stored in? I would like to access through php_agi. $agi-request['agi_dnid'] works for SIP channels but not IAX. Thanks! Michael ___ --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
