Kevin P. Fleming wrote:
...
The messages in bug 12099 are *not* errors, they are annoyances only.
The latest SVN branch 1.4 code of Asterisk will no longer generate them,
Using today's svn 3915:
..
Answer(Zap/2-1, ) in new stack
--
Tzafrir Cohen wrote:
On Sun, Mar 02, 2008 at 11:36:03AM -0500, sean darcy wrote:
Kevin P. Fleming wrote:
...
The messages in bug 12099 are *not* errors, they are annoyances only.
The latest SVN branch 1.4 code of Asterisk will no longer generate them,
Using today's
Kevin P. Fleming wrote:
sean darcy wrote:
In any event, as least for me the TDM400P seems to have problems with
zaptel svn - not just an annoyance.
As I've mentioned previously, the changes to fix this for good (assuming
they work properly) are in
http://svn.digium.com/svn/zaptel/team
Kevin P. Fleming wrote:
sean darcy wrote:
In any event, as least for me the TDM400P seems to have problems with
zaptel svn - not just an annoyance.
As I've mentioned previously, the changes to fix this for good (assuming
they work properly) are in
http://svn.digium.com/svn/zaptel/team
sean darcy wrote:
Kevin P. Fleming wrote:
sean darcy wrote:
In any event, as least for me the TDM400P seems to have problems with
zaptel svn - not just an annoyance.
As I've mentioned previously, the changes to fix this for good (assuming
they work properly) are in
http://svn.digium.com
Martin wrote:
.
In any event, as least for me the TDM400P seems to have problems with
zaptel svn - not just an annoyance.
As I've mentioned previously, the changes to fix this for good (assuming
they work properly) are in
ekiga registration fails. I've set nat = yes ( also blank ) and i've set
externip. Anybody have a sip.conf that works?
Here's the sip debug:
Reliably Transmitting (NAT) to 86.64.162.35:5060:
REGISTER sip:ekiga.net SIP/2.0
Via: SIP/2.0/UDP 10.10.11.180:5060;branch=z9hG4bK17818198;rport
sean darcy wrote:
ekiga registration fails. I've set nat = yes ( also blank ) and i've set
externip. Anybody have a sip.conf that works?
Here's the sip debug:
Reliably Transmitting (NAT) to 86.64.162.35:5060:
REGISTER sip:ekiga.net SIP/2.0
Via: SIP/2.0/UDP 10.10.11.180:5060;branch
I'm looking for a usb cordless handset to pair with a softphone (
probably ekiga) on a pc linked to an asterisk server. I've loooked at
the bluetooth headsets, but they seem overkill for just home phone
extensions.
I've found a number of handsets that work with skype, but they seem
locked
Gonzalo Servat wrote:
On Thu, Mar 27, 2008 at 1:56 PM, Tzafrir Cohen [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Any suggestions??
I'm using Asterisk 1.6.0-beta4 and Zaptel 1.4.9.2 http://1.4.9.2.
A freshly-built Asterisk? Built vs. zaptel 1.4.9.2
Kevin P. Fleming wrote:
Mojo with Horan Company, LLC wrote:
P.S. If you can't dial seven digit numbers in your area, but you miss
it, you can restore that behavior if you feel like selecting a default
area code:
exten = _NXX,1,Dial(Zap/1/907${EXTEN},,TWK)
Here, if I dial a seven
I'm set up to call 3 digit extensions at the office ( running 1.4.13)
from home ( 1.6.0 beta7) over iax. 1 out of 3 times the call breaks up,
but only in the home - office direction. office - home always sounds good.
If it were a poor internet connection, I'd expect both sides of the
John Beaman wrote:
John Beaman
Telecom Specialist II
Voice Telecommunications Services Department.
Good Samaritan National Campus
605-362-3331
[EMAIL PROTECTED] 4/4/2008 3:23:49 PM
I'm set up to call 3 digit extensions at the office ( running 1.4.13)
from home ( 1.6.0 beta7) over
Using 1.6-rc8.
In iax.conf on the calling box, I have:
[iax-out]
.
callerid = sean 447
I even also put the same on called box.
But I can't seem to set the callerid:
exten =_NXX,1,Answer()
exten =_NXX,n,NoOp(${CALLERID(num)})
Answer(IAX2/iax-in-7, ) in new stack
NoOp(IAX2/iax-in-7,
Tilghman Lesher wrote:
On Tuesday 06 May 2008 12:45:42 sean darcy wrote:
Using 1.6-rc8.
In iax.conf on the calling box, I have:
[iax-out]
.
callerid = sean 447
I even also put the same on called box.
But I can't seem to set the callerid:
exten =_NXX,1,Answer()
exten =_NXX,n
I'm trying to set the outgoing caller id to the DID number, but only if
the extension is greater than 140. MAINSTUB is simply the first 7 digits
of the main number. sip.conf sets the CALLERID(num) to the extension.
exten =_1NXXNXX,n,Set(CALLERID(num)=${MAINSTUB}${CALLERID(num)})
works. But
Barry Miller wrote:
On Fri, May 23, 2008 at 05:08:28PM -0400, sean darcy wrote:
This doesn't work:
exten =_1NXXNXX,n,Set( CALLERID(num) = ${IF ( $[${CALLERID(num)}
140] ? ${MAINSTUB}${CALLERID(num)} : ${MAINNUMBER} )})
Change IF ( to IF(.
Thanks for the response.
Tried
Barry Miller wrote:
On Sat, May 24, 2008 at 12:01:50AM -0400, sean darcy wrote:
Barry Miller wrote:
On Fri, May 23, 2008 at 05:08:28PM -0400, sean darcy wrote:
This doesn't work:
exten =_1NXXNXX,n,Set( CALLERID(num) = ${IF ( $[${CALLERID(num)}
140] ? ${MAINSTUB}${CALLERID(num
Try this. It WFM:
localnet=10.0.0.0/255.255.255.0
nat = yes
stunaddr = stun.ekiga.net ; or some other stun server, e.g.: foo.stun.com:3478
externrefresh = 15
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asterisk-users
iax.conf:
[nhi] ; receives calls
type=friend
secret=password
context=longdistance
qualify=yes
trunk=yes
callerid=test 447
extensions.conf:
[longdistance]
exten =_1NXXNXX,1,Answer()
exten =_1NXXNXX,n,NoOp(${CALLERID(num)} first stanza)
exten =_1NXXNXX,n,Dial(Zap/g0/${EXTEN})
sean darcy wrote:
iax.conf:
[nhi] ; receives calls
type=friend
secret=password
context=longdistance
qualify=yes
trunk=yes
callerid=test 447
extensions.conf:
[longdistance]
exten =_1NXXNXX,1,Answer()
exten =_1NXXNXX,n,NoOp(${CALLERID(num)} first stanza)
exten
sean darcy wrote:
sean darcy wrote:
iax.conf:
[nhi] ; receives calls
type=friend
secret=password
context=longdistance
qualify=yes
trunk=yes
callerid=test 447
extensions.conf:
[longdistance]
exten =_1NXXNXX,1,Answer()
exten =_1NXXNXX,n,NoOp(${CALLERID(num)} first
sean darcy wrote:
sean darcy wrote:
sean darcy wrote:
iax.conf:
[nhi] ; receives calls
type=friend
secret=password
context=longdistance
qualify=yes
trunk=yes
callerid=test 447
extensions.conf:
[longdistance]
exten =_1NXXNXX,1,Answer()
exten =_1NXXNXX,n,NoOp
Great.
But I'm still a little confused.
Does zaptel 1.4.12 work with asterisk-1.6.0-rc4?
It looks like we first upgrade to zaptel 1.4.12, and then to dahdi. We
can go back to this release of zaptel if we have problems with dahdi.
Or if we go back to zaptel, do we go back to 1.6.0-beta9 also?
upgrading from zaptel to dahdi, with a TDM400P:
Is /etc/dahdi/system.conf the same as /etc/zaptel.conf? As I read the
system.conf.sample, no echo canceller need be specified if there's a
hardware ec. Can I just rename zaptel.conf?
What about zapata.conf? Is this just renamed
http://www.asterisk.org/zaptel-to-dahdi is empty. Is there anyplace else
besides the README's and Upgrade.txt's for config info on updating?
sean
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As best i could figure it out, I've installed dahdi and rc4.
My TDM400P doesn't answer fxo or fxs.
/etc/dahdi/system.conf:
loadzone = us
defaultzone=us
fxoks=1,2
fxsks=4
/etc/asterisk/chan_dahdi.conf:
[house-phones]
context=internal ; Uses the [internal] context in extensions.conf
Tzafrir Cohen wrote:
On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote:
As best i could figure it out, I've installed dahdi and rc4.
My TDM400P doesn't answer fxo or fxs.
/etc/dahdi/system.conf:
loadzone = us
defaultzone=us
fxoks=1,2
fxsks=4
echocancel?
I thought
Tzafrir Cohen wrote:
On Fri, Sep 05, 2008 at 09:47:48AM -0400, sean darcy wrote:
Tzafrir Cohen wrote:
On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote:
As best i could figure it out, I've installed dahdi and rc4.
My TDM400P doesn't answer fxo or fxs.
/etc/dahdi/system.conf
Tzafrir Cohen wrote:
On Fri, Sep 05, 2008 at 07:15:52PM -0400, sean darcy wrote:
Tzafrir Cohen wrote:
On Fri, Sep 05, 2008 at 09:47:48AM -0400, sean darcy wrote:
Tzafrir Cohen wrote:
On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote:
As best i could figure it out, I've installed
sean darcy wrote:
Tzafrir Cohen wrote:
On Fri, Sep 05, 2008 at 07:15:52PM -0400, sean darcy wrote:
Tzafrir Cohen wrote:
On Fri, Sep 05, 2008 at 09:47:48AM -0400, sean darcy wrote:
Tzafrir Cohen wrote:
On Thu, Sep 04, 2008 at 10:58:44PM -0400, sean darcy wrote:
As best i could figure it out
Matt Gibson wrote:
I noticed one thing,
/etc/dahdi/system.conf:
loadzone = us
defaultzone=us
fxoks=1,2
fxsks=4
echocancel?
Tzafrir mentioned it earlier, but it may have gotten lost on the thread. I
was having problems with Dahdi until I added echocancel to our system.conf,
could
Matt Gibson wrote:
It seems to me the dahdi driver works. For some reason, however,
chan_dahdi doesn't see the channels the driver set up.
Anybody else using TDM400P with dahdi and rc4?
Hi Sean,
Not sure if it matters, but we're using 2.0R3, noticed you're on 2.0R2
Unfortunately,
Tzafrir Cohen wrote:
On Fri, Sep 05, 2008 at 08:39:16PM -0400, sean darcy wrote:
sean darcy wrote:
Tzafrir Cohen wrote:
On Fri, Sep 05, 2008 at 07:15:52PM -0400, sean darcy wrote:
Tzafrir Cohen wrote:
What messages do you get when you run in the CLI:
dahdi restart
dahdi restart
sean darcy wrote:
Filed bug:
http://bugs.digium.com/view.php?id=13443
sean
For any one else who has this problem:
don't use user-defined sections, i.e. [pstn]
If you use dahdichan = x , chan_dahdi will only use the _last_
dahdichan statement for channel(s).
channels = works
If you use iax, the console will tell you what codec is being used.
But for sip, nothing is shown. With sip debug on, I get:
Capabilities: us - 0x130e (gsm|ulaw|alaw|g729|speex|g722), peer -
audio=0x100e (gsm|ulaw|alaw|g722)/video=0x0 (nothing)/text=0x0
(nothing), combined - 0x100e
From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of sean darcy [EMAIL
PROTECTED]
Sent: Thursday, September 18, 2008 10:49 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] what codec is sip using?
If you use iax, the console will tell you what codec is being
Alex Balashov wrote:
sean darcy wrote:
David Gibbons wrote:
Sean,
Try 'sip show channels' or 'sip show channel channelid' for the drill
down. I believe the codec in use will be displayed with either command.
Dave
Thanks that worked. Now how do I get it show the codec when I'm
I'm using Teliax, and every incoming call has:
Executing [EMAIL PROTECTED]:2] Answer(IAX2/usrname-14376, ) in
new stack
-- Executing [EMAIL PROTECTED]:3] Dial(IAX2/usrname-14376,
DAHDI/1,60) in new stack
-- Called 1
-- DAHDI/1-1 is ringing
-- IAX2/usrname-14376 requested
Remco Barendse wrote:
The information (or lack of it) on upgrading from zaptel to that
@*^QW%^%!!! dahdi is very frustrating.
I cannot find anything on how to uninstall zaptel, i found an earlier post
to this list which suggested make uninstall and make remove in the zaptel
directory
In download dated 10/9.
Bug fix? Mistake?
sean
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Becasue of all the issues with fax over voip, we want to use pstn for
our fax machine, but not dedicate a line just to fax.
I'm thinking of having asterisk answer the pstn line, check for fax
tones, and route appropriately. In zapata ( chan_dahdi ) set
faxdetect=incoming
then the dial plan
I'm trying to set the callerid(name) to Office for all calls from the
main office.
exten =s,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} =
0${REGEX(21245711*)} ] ? Office:${CALLERID(name)} )})
The main office callerid's are all 212 457 11xx. But this statement
seems to match everything,
Anael DIAZ wrote:
Hi!
I have some problem in my asterisk 1.4.2, I've installed it on centOS 5.2
and this didn't accept voip QoS and can't route the packets having voip
QoS.
So I should change voip packets to be routing with centOS.
I want to use iproute2 but i don't what to do after
Jared Smith wrote:
On Mon, 2008-10-20 at 14:10 -0400, sean darcy wrote:
exten =s,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} =
0${REGEX(21245711*)} ] ? Office:${CALLERID(name)} )})
[snip]
What I'd expect is a callerid(num) of 2124571123 to generate an if test
of [02124571123
Atis Lezdins wrote:
On Mon, Oct 20, 2008 at 9:20 PM, Jared Smith [EMAIL PROTECTED] wrote:
On Mon, 2008-10-20 at 14:10 -0400, sean darcy wrote:
exten =s,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} =
0${REGEX(21245711*)} ] ? Office:${CALLERID(name)} )})
[snip]
What I'd expect is a callerid
On Mon, Oct 20, 2008 at 7:38 PM, sean darcy [EMAIL PROTECTED] wrote:
Atis Lezdins wrote:
On Mon, Oct 20, 2008 at 9:20 PM, Jared Smith [EMAIL PROTECTED] wrote:
On Mon, 2008-10-20 at 14:10 -0400, sean darcy wrote:
exten =s,n,Set(CALLERID(name)=${IF($[0${CALLERID(num)} =
0${REGEX(21245711
I've tried to create a subroutine that sets callerid name based on number.
extensions.conf:
...
exten = s,1,Answer()
exten = s,n,GoSub(set-callerid-name,0${CALLERID(num)},1)
exten = s,n,Dial(${mainline},60)
...
[set-callerid-name]
exten = 0,1,NoOp( no CALLERID num set)
exten =
OCG Technical Support wrote:
Take a look at smartCID at www.generationd.com
This tool will set callerid based on number in a database. If not listed
there, it will search 411 for reverse lookup etc.
It will also let you flag calls for blocking, etc..
Interesting. It looks like more than
Trevor Peirce wrote:
sean darcy wrote:
[set-callerid-name]
exten = 0,1,NoOp( no CALLERID num set)
exten = 02025462677,1,Set(CALLERID(name) = Fred )
exten = _X.,2,NoOp(CALLERID: ${CALLERID(name)})
exten = _X.,3,Return()
But it doesn't work. CALLERID(name) isn't changed
C F wrote:
Who you calling? Is it a remote non PSTN phone number? Or a PSTN number?
It's incoming. Both pstn and voip.
sean
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sean darcy wrote:
I've tried to create a subroutine that sets callerid name based on number.
extensions.conf:
...
exten = s,1,Answer()
exten = s,n,GoSub(set-callerid-name,0${CALLERID(num)},1)
exten = s,n,Dial(${mainline},60)
...
[set-callerid-name]
exten = 0,1,NoOp
Daniel Lynes wrote:
You'll need to lose the double quotation marks in the assignment:
Set(CALLERID(name)=Fred) becomes:
Set(CALLERID(name)=Fred)
If it still doesn't work, then it means that your particular provider
does not support the ability to be able to set the caller ID name, or
Doug Lytle wrote:
sean darcy wrote:
Tried it with and with quotes. Same result - exactly. Works with dummy
variable, doesn't if set in subroutine.
It works fine for me, I use the below:
exten = 3175797960,1,Gosub(get_name,s,1)
[get_name
Matt Riddell wrote:
On 13/11/2008 10:23 a.m., sean darcy wrote:
Tried it with and with quotes. Same result - exactly. Works with dummy
variable, doesn't if set in subroutine.
This is incoming. I'm setting the CID(name) based on the incoming
CID(num). Nothing to do with the provider
Installing 1.4.23-rc2, I actually looked at the startup and saw this
warning:
WARNING[10730]: loader.c:359 load_dynamic_module: Error loading module
'app_directory.so': /usr/lib/libc-client.so.2007: undefined symbol: mm_dlog
I'm running Fedora Core 9, with libc-client 2007d. googling didn't
I just want to pdf and email faxes coming in over pstn on a TDM400P.
Outgoing faxes would just go out over pstn, not through asterisk.
All the voipinfo , etc, howto's are quite complicated. And most use
third party apps like Hylafax.
I thought there was a rxfax and txfax in 1.4. And 1.6 had
Philipp Kempgen wrote:
sean darcy schrieb:
I thought there was a rxfax and txfax in 1.4. And 1.6 had app_fax. I'm
now using 1.4.22, but I'd go to 1.6 if it made this easier.
But I've found no docs or sample configs for either 1.4 or 1.6. In fact,
1.4.22 ( nor addons nor 1.4.23 rc2 ) have
starting 161.1-beta3:
chan_iax2.c:10925 build_user: Unable to support trunking on user
'iax-out' without DAHDI timing
But I have these timing modules:
ls /usr/lib/asterisk/modules/res_tim*
/usr/lib/asterisk/modules/res_timing_dahdi.so
/usr/lib/asterisk/modules/res_timing_pthread.so
Do I need
Russell Bryant wrote:
Michiel van Baak wrote:
On 20:24, Sun 14 Dec 08, sean darcy wrote:
starting 161.1-beta3:
chan_iax2.c:10925 build_user: Unable to support trunking on user
'iax-out' without DAHDI timing
But I have these timing modules:
ls /usr/lib/asterisk/modules/res_tim*
/usr/lib
Russell Bryant wrote:
Michiel van Baak wrote:
On 20:24, Sun 14 Dec 08, sean darcy wrote:
starting 161.1-beta3:
chan_iax2.c:10925 build_user: Unable to support trunking on user
'iax-out' without DAHDI timing
But I have these timing modules:
ls /usr/lib/asterisk/modules/res_tim*
/usr/lib
Using 1.6 on Fedora Core 9 I'm trying to receive faxes. I've got this far:
[incoming-fax]
exten =
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-0${CALLERIDNUM})
exten = s,2,ReceiveFAX(${FAXFILE}.tif)
exten = s,3,Hangup()
exten=h,1,System(/usr/local/bin/fax2mail
I've have a simple caller id lookup on incoming:
[teliax-in]
..
exten =s,n,GoSub(set-callerid-name,0${CALLERID(num)},1)
[set-callerid-name]
exten = 0,1,NoOp( no CALLERID num set)
exten = 02135590993,1,Set(CALLERID(name)=Matthew )
On Tue, Dec 23, 2008 at 10:13 AM, Noah Miller noahisaacmil...@gmail.com wrote:
Hi Tzafrir -
I'm wondering if anybody has IMAP Voicemail AND the directory working
together. I haven't had any success. IMAP voicemail works fine, but
when it's active, the Directory does not work. The problem
On Tue, Dec 30, 2008 at 1:30 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Tue, Dec 30, 2008 at 11:15:54AM -0500, sean darcy wrote:
Very interesting. I tried this with 1.6.1-beta4 on Fedora 9. On
startup, asterisk looks in the --with-imap folder which has just a
static lib
On 1.6.1-beta4:
Trying to receive faxes over a pstn line. extensions.conf:
[incoming-pstn-line]
exten = fax,1,NoOp(Fax Detected)
exten = fax,2,GoTo(incoming-fax,s,1)
exten = fax,n,Hangup()
[incoming-fax]
exten =
OCG Technical Support wrote:
Start with your mail log. Any errors visible?
How about system log - PAMpermission errors?
Thanks for the quick response. maillog shows nothing if it's executed
from the System() call. Obviously maillog shows the outgoing if executed
from the terminal,
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
Lyle Giese wrote:
If you are running the script within Asterisk as root, then it's a path
environment issue. My guess(and I run into this with cron jobs all the
time) is that the path is different from the command line than the
environment that the script runs under.
There are times
OCG Technical Support wrote:
If you want to email me your fixed script I'll put it up on the web site...
Well I'd be pleased to have any script of mine put up on any web site,
but the only thing I did was to hard wire my location of mime-construct:
MimeC=/usr/local/bin/mime-construct
and
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
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
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 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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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
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'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, 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 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
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
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'
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
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
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
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/
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