YC Nyon wrote:
hi,
I'm been successful in making calls to another local extension using
Nokia E71. However calling the E71 from another ext. (X-lite) is not
successful. There is a ringing tone from the caller side but the E71 is
silent.
Tried disabling the NAT (dunno whether that
Tzafrir Cohen wrote:
On Fri, Feb 05, 2010 at 01:55:03PM -0500, sean darcy wrote:
sean darcy wrote:
Using 1.6.2.1 with a TDM400, attached to internal analog phones and
PSTN. When I dial out to PSTN, I cannot send tones, like push 1 for
something stupid. The call itself works, but the DTMF
Nikhil Nair wrote:
Hi,
I'm getting some strange behaviour on Asterisk 1.4 running on Debian
Stable (Lenny). I suspect it's something to do with my setup, rather than
a bug, but I'm struggling to see it, and would appreciate any input.
Thanks for posting this. And for persistently
sean darcy wrote:
Using 1.6.2.1 with a TDM400, attached to internal analog phones and
PSTN. When I dial out to PSTN, I cannot send tones, like push 1 for
something stupid. The call itself works, but the DTMF tones fail.
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [6258
Using 1.6.2.1 with a TDM400, attached to internal analog phones and
PSTN. When I dial out to PSTN, I cannot send tones, like push 1 for
something stupid. The call itself works, but the DTMF tones fail.
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [6258...@internal:1]
sean darcy wrote:
Calling from my home using Asterisk 1.6.2.1 to an office extension
(Asterisk 1.6.1.13) the callerid is not honored:
Home:
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [...@internal:1] Answer(DAHDI/1-1, ) in new stack
-- Executing [...@internal:2
Steve Howes wrote:
On 31 Jan 2010, at 16:24, sean darcy wrote:
-- Executing [...@internal:3] Set(DAHDI/1-1, CALLERID=Test
447) in new stack
Why isn't the office asterisk picking up the callerid from the home
asterisk?
You're making up the syntax?
http://www.voip-info.org/wiki/view
listu...@spamomania.co.uk wrote:
On Thu, 2010-01-28 at 23:11 -0600, Karl Fife wrote:
Appears completely resolved!
No more home-spun patches!
Thanks!
-K
It's *not* fixed here:
DAHDI Version: 2.2.1 Echo Canceller: MG2
But as is depressingly the 'norm' for Asterisk it comes back to
Calling from my home using Asterisk 1.6.2.1 to an office extension
(Asterisk 1.6.1.13) the callerid is not honored:
Home:
-- Starting simple switch on 'DAHDI/1-1'
-- Executing [...@internal:1] Answer(DAHDI/1-1, ) in new stack
-- Executing [...@internal:2] NoOp(DAHDI/1-1, Context:
On Sun, Dec 6, 2009 at 12:28 PM, Thomas Perron thomas.per...@gmail.com wrote:
I am reading a lot of the material but need your input to help me
understand what you mean.
System(echo body of message | mail -s subject line
${the_caller_...@tmobile.net)
I understand the System application
I've got a single TDM 400P board with two internal ports and 1 external.
chan_dahdi.conf:
context=internal ; Uses the [internal] context in extensions.conf
signalling=fxo_ks ; fxo_ks not auto Use FXO signalling for FXS
group=0
channel = 1 ; Telephone attached to port 1
channel
sean darcy wrote:
I've got a single TDM 400P board with two internal ports and 1 external.
chan_dahdi.conf:
context=internal ; Uses the [internal] context in extensions.conf
signalling=fxo_ks ; fxo_ks not auto Use FXO signalling for FXS
group=0
channel = 1 ; Telephone
I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk
B. Both are behind NAT, but port forwarded. I get the connection, but no
voice - either in or out.
I can call on SIP from A to B (and from B to A). Do it all the time.
Asterisk A receives SIP calls from Junction and
sean darcy wrote:
I'm trying to connect a sip call from sipgate to Asterisk A to Asterisk
B. Both are behind NAT, but port forwarded. I get the connection, but no
voice - either in or out.
I can call on SIP from A to B (and from B to A). Do it all the time.
Asterisk A receives SIP calls
In SIP setting on the e71 I set the public user name as
1...@10.10.11.180. There is a sip.conf context [1995]
On the asterisk CLI I get:
Registration from 'sip:%201...@10.10.11.180:5060' failed for
'10.10.11.98' - No matching peer found
So I changed the sip.conf context to [%201995]
Then:
sean darcy wrote:
Leif Madsen wrote:
sean darcy wrote:
On 1.6.0.18-rc3 using app_fax.so, spandsp-0.0.5, anytime I use SendFAX
asterisk restarts:
Before I file a bug, is there anything I'm missing?
Does this happen on earlier versions of the 1.6.0 series prior to this
release
candidate
On 1.6.0.18-rc3 using app_fax.so, spandsp-0.0.5, anytime I use SendFAX
asterisk restarts:
[Nov 15 19:00:36] VERBOSE[17013] logger.c: -- Executing
[...@fax-tx-test:1] ESC[1;36;40mNoOpESC[0;37;40m(ESC[1;35;40mSIP/nhi-rive
rside-sip-ESC[0;37;40m, ESC[1;35;40mContext
Leif Madsen wrote:
sean darcy wrote:
On 1.6.0.18-rc3 using app_fax.so, spandsp-0.0.5, anytime I use SendFAX
asterisk restarts:
Before I file a bug, is there anything I'm missing?
Does this happen on earlier versions of the 1.6.0 series prior to this
release
candidate? I'm curious
Karl Fife wrote:
Question about the proper way to update LibPRI:
'Bouncing' asterisk after an installing the new LibPRI version does
indeed reflect the update:
asterisk*CLI pri show version
libpri version: 1.4.10.2
Hmm. What asterisk version are you running?
On 1.6.0.18-rc2:
pbx*CLI
I'm trying to setup sipgate on 1.6.1. Following the instructions on the
site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk,
I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf:
[sipgate]
type=friend
secret= ;;SIP_PASSWORD
insecure=port,invite
sean darcy wrote:
I'm trying to setup sipgate on 1.6.1. Following the instructions on the
site: http://www.sipgate.com/faq/article/394/How_do_I_configure_Asterisk,
I created [sipgate] in sip.conf and a [sipgate-in] in extensions.conf:
[sipgate]
type=friend
secret= ;;SIP_PASSWORD
On Tue, Sep 29, 2009 at 2:48 PM, David Backeberg dbackeb...@gmail.com wrote:
On Mon, Sep 28, 2009 at 10:08 PM, sean darcy seandar...@gmail.com wrote:
On Mon, Sep 28, 2009 at 1:09 PM, David Backeberg dbackeb...@gmail.com
wrote:
Have you tried using ps2tiff?
I looked up ps2tiff. That seems
sean darcy wrote:
Martin wrote:
u don't change the ${uniquefile} for the second System/Originate
try to add a string to the ${uniquefile} ...
eg
${uniquefile}0
Martin
But I generate another unique file in [fax-tx] just before I try to send
the confirm.
Here's the first call
On Mon, Sep 28, 2009 at 1:09 PM, David Backeberg dbackeb...@gmail.com wrote:
On Mon, Sep 28, 2009 at 12:30 PM, sean darcy seandar...@gmail.com wrote:
Well one of the problems it that SendFax doesn't like the tiff file(BTW,
SendFax from app_fax.so gives you clue what the problem is). It requires
Photometric Interpretation: min-is-black
FillOrder: msb-to-lsb
Orientation: row 0 top, col 0 lhs
Samples/Pixel: 1
Rows/Strip: 109
Planar Configuration: single image plane
DocumentName: Standard Input
ImageDescription: converted PNM file
sean
On Sat, Sep 26, 2009 at 8:05 PM, sean
On 1.6.0.16-rc1:
I'm using app_fax.so to send a fax, and then send a confirm.
'send' = 1.
Set(UniqueFile=/var/spool/asterisk/outgoing/call-${UNIQUEID}) [pbx_config]
2. System(env echo -e
Channel:DAHDI/g0/\\nContext:fax-tx\\nExtension: s\\nPriority:
1\\n
In my quest to actually send a fax, I'm now stuck trying to send the
confirm.
First I send the fax:
-- Executing [s...@outbound-fax:2] System(Console/dsp, env echo
-e Channel:DAHDI/g0/12036378447\\nContext:fax-tx\\nExtension:
s\\nPriority: 1\\n
Martin wrote:
if you're trying to send the same fax to both parties, then do
exten = s,1,System()
exten = s,2,Sendfax()
step1 will spool the call to dial a number and send a fax
step2 will transmit the fax to the incoming call
Martin
On Wed, Sep 23, 2009 at 7:45 PM, sean darcy
sean darcy wrote:
Martin wrote:
if you're trying to send the same fax to both parties, then do
exten = s,1,System()
exten = s,2,Sendfax()
step1 will spool the call to dial a number and send a fax
step2 will transmit the fax to the incoming call
Martin
On Wed, Sep 23, 2009 at 7:45 PM
, the current extension is used. You cannot use
any additional
action post answer options in conjunction with this option.
your priority+1 is Hangup ...
is that it ?
Martin
On Tue, Sep 22, 2009 at 7:32 PM, sean darcy seandar...@gmail.com wrote:
Using Digium fax I've tried
there's none you can do
exten = _X.,n,System(echo -e Channel: SIP/num...@gateway\\ncontext:
send\\nExtension: s\\nPriority: 1\\n
/var/spool/asterisk/outgoing/call-${UNIQUEID})
and at send,s,1 call sendfax
Martin
On Wed, Sep 23, 2009 at 1:44 AM, sean darcy seandar...@gmail.com wrote:
Martin
Using Digium fax I've tried a simple dialplan:
'8447' = 1. Answer() [pbx_config]
2. Set(CALLERID(num)=xxxyyy) [pbx_config]
3. Dial(DAHDI/g0/1bbbccc,,G(send))[pbx_config]
[send]4.
In my attempts to set up digium fax I get an odd warning:
-- Executing [...@capture-fax:2] ReceiveFAX(SIP/173-b53023e8,
/var/spool/asterisk/fax/20090921_1806.tif) in new stack
-- Channel 'SIP/173-b53023e8' receiving fax
'/var/spool/asterisk/fax/20090921_1806.tif'
[Sep 21 18:06:37]
Trying to build oslec. Following dahdi-linux README I copy
drivers/staging/echo to dahdi-linux/drivers/staging. I uncomment the 2
oslec lines in drivers/dahdi/Kbuild.
That doesn't work:
/home/asterisk/build/dahdi/svn/dahdi-linux/drivers/dahdi/dahdi_echocan_oslec.c:33:35:
error:
My set up is 1.6.0.15 with the digium fax modules. I want to capture a
fax from the internal analog fax machine (using an SPA2102), and then
resend it. I know the internal extension of the fax machine, and for now
I'm just testing it to one outside fax machine if I dial 8447.
In particular,
On Wed, Jul 15, 2009 at 5:23 PM, Waynewa...@planetwayne.com wrote:
Hi all,
Just a quickie to say that this has been solved now - real simple -
downloaded '*current*' rather than the versions from the home page of
Astrisk.org. (didn't realise there was a 'current' version tbh.
Anyways - I
I've * in a small office with 10 internal sip extensions on aastra's.
Outgoing is pstn over dahdi, voip over teliax and iax to another office.
This morning no calls could be made: iax to branch offfices, voip iax
over teliax, pstn, or even internal extensions. The aastra's showed Not
in Service.
Trying to set up Junction Networks for outgoing on 1.6.1:
extensions.conf:
exten = _99X.,n,Dial(IAX2/jnctn_out/${called-num})
iax.conf
[jnctn_out]
type=peer
host=iax.jnctn.net
username=
secret=
qualify=yes
I'm not using realtime.
But CLI:
-- Executing [99xxxy...@internal:3]
On Tue, Jun 16, 2009 at 10:13 AM, Miguel Molinammol...@millenium.com.co wrote:
sean darcy escribió:
For our internal fax machines, I'm checking if the faxes are going to
branch offices. If they are, I want to capture and email them to the
branches. I've set up extension 8447 to test
For our internal fax machines, I'm checking if the faxes are going to
branch offices. If they are, I want to capture and email them to the
branches. I've set up extension 8447 to test this.
A fax machines is connected via an SPA 2102 on 173. Any calls from 173
are sent to:
[outbound-fax]
Tzafrir Cohen wrote:
On Sun, Jun 07, 2009 at 10:15:42PM -0500, Tilghman Lesher wrote:
IAXmodem is a completely different ball of wax, and I think you would agree
that if the builtin FAX support in spandsp provided excellent support, there
never would have been a reason for IAXmodem to be
Jared Smith wrote:
On Sat, 2009-06-06 at 21:24 +0200, Philipp Kempgen wrote:
exten = s,n,Set(CALLERID(num)=${IF($[0${CALLERID(num)} 140] ?
${MAINSTUB}${CALLERID(num)}:${MAINNUMBER} )} )
^ ^
remove the trailing spaces
Philipp Kempgen wrote:
sean darcy schrieb:
I'm having trouble setting callerid with teliax. I use a simple dial-out
subroutine to set the callerid depending on the calling extension, and
then dial out. Teliax is saying they're not seeing any callerid info.
exten = s,n,Set(CALLERID(num
Steve Underwood wrote:
Elliot Murdock wrote:
Hello!
I have a 64 bit Asterisk system and am wondering how to use Digium's
32 bit fax driver. Is there some kind of emulation that can be used?
Thanks!
Elliot
Use the FAX support built into Asterisk 1.6 and you won't have that
limitation.
I'm having trouble setting callerid with teliax. I use a simple dial-out
subroutine to set the callerid depending on the calling extension, and
then dial out. Teliax is saying they're not seeing any callerid info.
[DialOut] ; subroutine for dialing out.
exten = s,1,NoOp(Context: DialOut called
David Backeberg wrote:
You don't say the kind of call you're making, but if you're using
MeetMe() I have more advice regarding voice quality with conference
rooms.
I don't know about the OP, I'd sure appreciate any advice regarding
voice quality with MeetMe(). When we have 2 -3 internal
Still trying to upgrade to 1.6.0.9 for 1.4.
It worked - it worked all day yesterday, but this morning:
-- Executing [646xxxy...@longdistance:1]
Answer(SIP/172-08276a60, ) in new stack
..
-- Executing [646xxx...@longdistance:6] Dial(SIP/172-08276a60,
DAHDI/g2/1646xxx) in
Jared Smith wrote:
On Wed, 2009-05-27 at 10:46 -0400, sean darcy wrote:
-- Executing [646xxx...@longdistance:6] Dial(SIP/172-08276a60,
DAHDI/g2/1646xxx) in new stack
It appears you're attempting to dial DAHDI/g2/1646xxxyyy instead of
DAHDI/g2/1646xxx... Did you mean to put
The local telco is now going 10 digit dialing even for local (free)
calls which used to be 7 digit. For a while no problem, everyone will
continue to dial 7 digits, and I'll add the area code. But pretty soon
everyone will become used to 10 digits.
There are about 40 3 digit local exchanges.
Barry L. Kline wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
sean darcy wrote:
I've looked at the Berkeley DB. That works pretty well, if the exchanges
are all stored. But it looks like the exchanges have to be entered 1 by
1 from the CLI. And can only be reviewed, corrected
Pascal Bruno wrote:
It seems like a few people including me DID understand what Dhaval
meant, or maybe some people used they common sense and their
intelligence to understand what somebody who's english is not the
primary language wanted to say and put some effort to guide or help
someone
I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 over this weekend.
I'm getting:
[May 23 10:56:33] ERROR[26017]: chan_sip.c:2922 __sip_reliable_xmit:
Serious Network Trouble; __sip_xmit returns error for pkt data
[May 23 10:56:33] ERROR[26017]: chan_sip.c:2922 __sip_reliable_xmit:
Serious
I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 with a TE120P card.
I can't make any connection over the T1.
From CLI:
ERROR[26017]: chan_dahdi.c:14300 process_dahdi: Unknown signalling
method 'pri_cpe' at line 37.
cat chan_dahdi.conf
cat chan_dahdi.conf
[trunkgroups]
[channels]
Tzafrir Cohen wrote:
On Sat, May 23, 2009 at 12:23:50PM -0400, sean darcy wrote:
I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 with a TE120P card.
I can't make any connection over the T1.
From CLI:
ERROR[26017]: chan_dahdi.c:14300 process_dahdi: Unknown signalling
method 'pri_cpe
Markus Weiler wrote:
Hi,
In VI:
In 'vi', moving the cursor over any bracket, brace, etc, and then
pressing '%' moves the cursor to the 'matching' bracket/brace character.
That can be very useful when programming, to find missing/extra brackets
and braces. It even seems to find
I'm testing an upgrade of an i686 production machine running 1.4.24.1 to
1.6.0.9. I've installed dahdi-linux-2.1.0.4.
But:
asterisk -cvvv
Asterisk 1.6.0.9, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer marks...@digium.com
Asterisk
Dan Austin wrote:
Sean wrote:
Tilghman Lesher wrote:
On Saturday 16 May 2009 08:21:43 sean darcy wrote:
With 1.6.1, I'm trying to set up a test of meetme for creating dynamic
conferences.
Trimmed
I don't want the conference to stay up forever, since I'd like new pin's
each time
On Tue, May 12, 2009 at 3:05 PM, James A. Shigley j...@answeringserv.com
wrote:
Unless there is a new feature or your making a new system. Don’t fix it if
it aint broke.
BUT do stay current on reading about new feature and things in the releases.
James Shigley
From:
On 1.6.1, I must be losing my eyesight:
[internal]
include = outbound-pstn
.
include = meetme; 2663
include = setup-meetme-conf-room ; 6000xxx
[setup-meetme-conf-room]
exten = _6000XXXNXXX,n,Set(Time-in-secs=${STRFTIME(${EPOCH},,%s} )
CLI:
Philipp Kempgen wrote:
sean darcy schrieb:
On 1.6.1, I must be losing my eyesight:
exten = _6000XXXNXXX,n,Set(Time-in-secs=${STRFTIME(${EPOCH},,%s} )
exten = _6000XXXNXXX,n,Set(Time_in_secs=${STRFTIME(${EPOCH},,%s)})
^
CLI
sean darcy wrote:
sean darcy wrote:
Mark Michelson wrote:
sean darcy wrote:
Danny Nicholas wrote:
You lost conf-getconfno.gsm . Asterisk is trying to play that file to
let
you pick a conference number to use. It goes in /var/lib/asterisk/sounds.
Grep for it.
-Original Message
With 1.6.1, I'm trying to set up a test of meetme for creating dynamic
conferences.
extensions.conf:
[meetme]
exten = 2663,1,MeetMe(,De)
exten = 2663,n,Hangup()
exten = 2666,1,MeetMe()
exten = 2666,n,Hangup()
What I'm expecting is to dial 2663, get a conference room number ( 600,
I suppose
Tilghman Lesher wrote:
On Saturday 16 May 2009 08:21:43 sean darcy wrote:
With 1.6.1, I'm trying to set up a test of meetme for creating dynamic
conferences.
extensions.conf:
[meetme]
exten = 2663,1,MeetMe(,De)
exten = 2663,n,Hangup()
exten = 2666,1,MeetMe()
exten = 2666,n,Hangup
Yehavi Bourvine wrote:
You check for BUSY. Check for IN_USE instead. That's what I do here (on
1.4, but I guess that 1.6 behaves similarly).
When an extension is in IN_USE state I have a decision tree after
consulting a database:
* If the user wants waiting call - dial him/her/
With 1.6.1, I'm trying to set up a test of meetme for creating dynamic
conferences.
cat meetme.conf
[rooms]
conf = 600
extensions.conf:
[meetme]
exten = 2663,1,MeetMe(,D)
exten = 2663,n,Hangup()
exten = 2666,1,MeetMe()
exten = 2666,n,Hangup()
What I'm expecting is to dial 2663, get a
...@lists.digium.com] On Behalf Of sean darcy
Sent: Friday, May 15, 2009 12:39 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] meetme dies looking for conf-getconfno
With 1.6.1, I'm trying to set up a test of meetme for creating dynamic
conferences.
cat meetme.conf
[rooms
sean darcy wrote:
Danny Nicholas wrote:
You lost conf-getconfno.gsm . Asterisk is trying to play that file to let
you pick a conference number to use. It goes in /var/lib/asterisk/sounds.
Grep for it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Steve Edwards wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Friday, May 15, 2009 12:39 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] meetme dies looking
Mark Michelson wrote:
sean darcy wrote:
Danny Nicholas wrote:
You lost conf-getconfno.gsm . Asterisk is trying to play that file to let
you pick a conference number to use. It goes in /var/lib/asterisk/sounds.
Grep for it.
-Original Message-
From: asterisk-users-boun
sean darcy wrote:
Mark Michelson wrote:
sean darcy wrote:
Danny Nicholas wrote:
You lost conf-getconfno.gsm . Asterisk is trying to play that file to let
you pick a conference number to use. It goes in /var/lib/asterisk/sounds.
Grep for it.
-Original Message-
From: asterisk-users
I have two internal analogue extensions off a TDM400P. If the first is
busy, I'd like to ring the second. So:
[incoming]
exten =s,1,Answer()
exten =s,n,Dial(${mainline},60)
exten =s,n,ExecIf($[${DIALSTATUS} = BUSY]?Dial(${secondline},30))
But it doesn't work because * first tries Call Waiting
sean darcy wrote:
I have two internal analogue extensions off a TDM400P. If the first is
busy, I'd like to ring the second. So:
[incoming]
exten =s,1,Answer()
exten =s,n,Dial(${mainline},60)
exten =s,n,ExecIf($[${DIALSTATUS} = BUSY]?Dial(${secondline},30))
But it doesn't work because
On Fedora 11, gcc-4.4, I'm trying to build oslec in dahdi-linux, but:
[aster...@asterisk dahdi-linux]$ make
make -C drivers/dahdi/firmware firmware-loaders
make[1]: Entering directory
`/home/asterisk/build/dahdi/svn/dahdi-linux/drivers/dahdi/firmware'
make[1]: Leaving directory
Rilawich Ango wrote:
Can you try to disable call waiting in your phone?
On Fri, May 15, 2009 at 6:44 AM, sean darcy seandar...@gmail.com wrote:
sean darcy wrote:
I have two internal analogue extensions off a TDM400P. If the first is
busy, I'd like to ring the second. So:
[incoming]
exten
Tzafrir Cohen wrote:
On Thu, May 14, 2009 at 06:37:53PM -0400, sean darcy wrote:
I have two internal analogue extensions off a TDM400P. If the first is
busy, I'd like to ring the second. So:
[incoming]
exten =s,1,Answer()
exten =s,n,Dial(${mainline},60)
exten =s,n,Dial(DAHDI/g5,60
David Backeberg wrote:
On Mon, May 4, 2009 at 10:52 PM, sean darcy seandar...@gmail.com wrote:
Receiving a fax with 1.6.1:
== Spawn extension (incoming-pstn-line, fax, 1) exited non-zero on
'DAHDI/4-1'
-- Executing [...@incoming-pstn-line:1] NoOp(DAHDI/4-1, Fax
Detected) in new stack
Receiving a fax with 1.6.1:
== Spawn extension (incoming-pstn-line, fax, 1) exited non-zero on
'DAHDI/4-1'
-- Executing [...@incoming-pstn-line:1] NoOp(DAHDI/4-1, Fax
Detected) in new stack
-- Executing [...@incoming-pstn-line:2] Goto(DAHDI/4-1,
incoming-fax,s,1) in new stack
1.6.1 svn 190575:
CC=cc CXX=g++ LD= AR= RANLIB= CFLAGS= make -C menuselect
CONFIGURE_SILENT=--silent menuselect
make[1]: Entering directory
`/home/asterisk/rpmbuild/BUILD/asterisk-1.6.1/menuselect'
gcc -m64 -march=native -mtune=native -floop-interchange
-floop-strip-mine -floop-block -c -o
With 1.6.1 svn:
[2009-04-26 15:01:00] NOTICE[1844]: chan_sip.c:9927 sip_reg_timeout:
-- Registration for '17470121...@proxy01.sipphone.com' timed out, trying
again (Attempt #30)
[2009-04-26 15:01:00] WARNING[1844]: acl.c:376 ast_get_ip_or_srv: Unable
to lookup 'proxy01.sipphone.com'
We're getting a new server. I'm considering installing 64bit fedora
rather than the 32bit we use now. Is 64 bit a problem with asterisk? Any
issues we should expect?
sean
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John Novack wrote:
Suggest you use CentOS rather than Fedora.
CentOS has a longer support life, with the same cost.
JMO
John Novack
sean darcy wrote:
We're getting a new server. I'm considering installing 64bit fedora
rather than the 32bit we use now. Is 64 bit a problem
I've a TDM400 with dahdi 2.1.0.4, asterisk 1.6.1-rc5. Asterisk is
randomly hanging up calls coming over the pstn. Often it happens right
as the call is answered:
-- Starting simple switch on 'DAHDI/4-1'
[Apr 22 17:09:38] NOTICE[20123]: chan_dahdi.c:7505 ss_thread: Got event
18 (Ring
On Sat, Apr 11, 2009 at 12:04 PM, David fire ddf...@gmail.com wrote:
hi
there are a lot of virtualization solution out there and every one is the
best and has some pro and some cons...
wich one do you recomend?
the idea to isolate diferents servers asterisk apache ... it is a good idea?
I've been trying to build h323plus (both the release and svn) for
chan_h323 on Fedora 10. No joy. I posted on the h323plus ml, but no
response.
Anybody here actually built it on Fedora? Wanna share your secrets, or
even better a specfile?
sean
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On Sat, Mar 28, 2009 at 11:25 PM, Zvonimir Mileta zmil...@hotmail.com wrote:
hi folks, Im pretty sure this has been covered before but I just wasnt able
to find any answer.
Im having troubles with the call pickup feature, is just not working for me.
whenever I press *8 or 200 or anyother.
I'm setting up dialplans to deal with 800 dialing through a different
channel than regular long distance in the US.
The regular long distance is set up so users can but don't have to
dial one. That's pretty easy, just one more exten statement. But it's
a pain dealing with all the 8xx area codes
I posted this before, but it didn't show up. So if it's a dup...
I'm setting up dialplans to deal with 800 dialing through a different
channel than regular long distance in the US.
The regular long distance is set up so users can but don't have to
dial one. That's pretty easy, just one more
Tilghman Lesher wrote:
On Tuesday 24 February 2009 13:44:25 Barry L. Kline wrote:
Here's one that may be of interest to any upgraders. If you rely on the
behavior of gosub you may want to make note of this change.
I have an incoming call context:
exten =
Tilghman Lesher wrote:
On Wednesday 25 February 2009 09:51:23 Klaus Darilion wrote:
Tilghman Lesher schrieb:
On Tuesday 24 February 2009 16:07:52 Klaus Darilion wrote:
Barry L. Kline wrote:
that is supposed to gosub into the incoming extension at priority 1.
Versions before 1.6.0.6 would
Tilghman Lesher wrote:
.
... but I absolutely
defend fixing this bug in Gosub, given that I'm the designer of it, and it was
never supposed to fail into the i extension.
Wow.
sean
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Doug Bailey wrote:
- sean darcy seandar...@gmail.com wrote:
Doug Bailey wrote:
- sean darcy seandar...@gmail.com wrote:
pstn incoming on a TDM400P, sometimes i* won't answer, going into
a loop like this:
-- Starting simple switch on 'DAHDI/4-1'
[Jan 16 10:38:55] NOTICE[5808
Tilghman Lesher wrote:
On Friday 16 January 2009 20:27:57 sean darcy wrote:
Tilghman Lesher wrote:
On Friday 16 January 2009 17:43:21 sean darcy wrote:
Danny Nicholas wrote:
Why not do a zap restart instead of restarting asterisk? You could
write an AGI to do the ZR when the condition
Doug Bailey wrote:
- sean darcy seandar...@gmail.com wrote:
Doug Bailey wrote:
- sean darcy seandar...@gmail.com wrote:
pstn incoming on a TDM400P, sometimes i* won't answer, going into
a loop like this:
-- Starting simple switch on 'DAHDI/4-1'
[Jan 16 10:38:55] NOTICE[5808
sean darcy wrote:
Doug Bailey wrote:
- sean darcy seandar...@gmail.com wrote:
Doug Bailey wrote:
- sean darcy seandar...@gmail.com wrote:
pstn incoming on a TDM400P, sometimes i* won't answer, going into
a loop like this:
-- Starting simple switch on 'DAHDI/4-1'
[Jan 16 10:38
pstn incoming on a TDM400P, sometimes i* won't answer, going into
a loop like this:
-- Starting simple switch on 'DAHDI/4-1'
[Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event
18 (Ring Begin)...
[Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2
Doug Bailey wrote:
- sean darcy seandar...@gmail.com wrote:
pstn incoming on a TDM400P, sometimes i* won't answer, going into
a loop like this:
-- Starting simple switch on 'DAHDI/4-1'
[Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event
18 (Ring Begin)...
[Jan
Danny Nicholas wrote:
Why not do a zap restart instead of restarting asterisk? You could write
an AGI to do the ZR when the condition occurred and lines where empty.
Yes, a cron job to restart zaptel would cut off any call then existing.
But how would I test for it? I can imagine:
Tilghman Lesher wrote:
On Friday 16 January 2009 17:43:21 sean darcy wrote:
Danny Nicholas wrote:
Why not do a zap restart instead of restarting asterisk? You could
write an AGI to do the ZR when the condition occurred and lines where
empty.
Yes, a cron job to restart zaptel would cut off
I'm trying to capture faxes on 1.6.1-beta4. AFAICT, app_fax is working
OK. I'm then using fax2mail to send the fax. That wasn't working, so i
posted for help using the System() cmd, since fax2mail did work from the
command line. But now I realize it's fax2mail and mime-construct itself.
I set
Joseph L. Casale wrote:
Have you tried your system stuff under su - asterisk? Once it works that
way, the system() command will work.
asterisk is running as root, I run the command at the terminal as root.
I am guessing he doesn't even have an asterisk user.
Well I do have an asterisk
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