Receiving a fax pstn - pstn with 1.6.2.6-rc2:
-- Executing [...@incoming-pstn-line:1] Answer("DAHDI/4-1", "") in
new stack
-- Executing [...@incoming-pstn-line:2] Wait("DAHDI/4-1", "3") in new
stack
-- Executing [...@incoming-pstn-line:3] Dial("DAHDI/4-1",
"DAHDI/g0,36") in new s
Fred Posner wrote:
> On Mar 5, 2010, at 1:01 PM, sean darcy wrote:
>
>> The issues are that sip doesn't work,
>
>
> What does "doesn't work" mean? In / Out? Both? Do you have a sip trace?
>
>> even though this same set up
>> worked with
Anthony Messina wrote:
> On Sunday 07 March 2010 09:16:55 am sean darcy wrote:
>> Well, I've figured it out, at least for me.
>>
>> Another driver was grabbing the TDM400P: netjet.
>>
>> added netjet to /etc/modprobe.d/blacklist.conf.
>>
>> I think
Anthony Messina wrote:
> On Saturday 06 March 2010 09:18:13 pm sean darcy wrote:
>> I have a TDM400. Just updated Fedora 12 to kernel 2.6.32. Rebuilt and
>> installed dahdi-2.2.1.
>>
>> kernel modules loaded.
>> lsmod | grep wctdm
>> wctdm 3
I have a TDM400. Just updated Fedora 12 to kernel 2.6.32. Rebuilt and
installed dahdi-2.2.1.
kernel modules loaded.
lsmod | grep wctdm
wctdm 37233 0
dahdi 194985 1 wctdm
lsmod | grep dahdi
dahdi 194985 1 wctdm
crc_ccitt 1549 2
On Wed, Mar 3, 2010 at 1:23 PM, Fred Posner wrote:
>
> On Mar 3, 2010, at 1:03 PM, sean darcy wrote:
>
>> Well at least my RG doesn't let you use DMZplus _unless_ you've chosen
>> dhcp. So I did. And the RG shows the router as DMZplus. And I can ssh
>
Warren Selby wrote:
> You need to set your firewall public ip to dhcp in order for Uverse
> dmz to work.
>
>
>
> Thanks,
> --Warren Selby
>
> On Mar 2, 2010, at 8:53 PM, sean darcy wrote:
>
>> Fred Posner wrote:
>>> On Mar 2, 2010, at 6:27
Fred Posner wrote:
> On Mar 2, 2010, at 6:27 PM, sean darcy wrote:
>
>> I've just got Uverse installed. I had dsl, but ATT insisted I couldn't
>> keep my old dsl, but had to switch to Uverse internet - vdsl.
>>
>> My setup:
>>
>> l
I've just got Uverse installed. I had dsl, but ATT insisted I couldn't
keep my old dsl, but had to switch to Uverse internet - vdsl.
My setup:
linux box as router : 10.10.11.252
asterisk box: 10.10.11.180
10.10.11.252 is multihomed and connected to the Uverse Residential
Gateway. I've
Håkon Nessjøen wrote:
> You only need to answer() the call when you want to play audio, or
> music on hold, receive dtmf, etc.
> If you are just sending the incoming call to a Dial() or Queue(without
> music on hold), you don't need to answer.
> The receiving party will do the answering. This way t
Kevin P. Fleming wrote:
> sean darcy wrote:
>
>> OK, now clear on suffix v. prefix ( Doh! ) and having RTFM,
>>
>> I have extensions.conf:
>>
>> [general]
>>
>> #include exts/gvoice.exten.conf
>>
>>
>> s
Olle E. Johansson wrote:
> 11 feb 2010 kl. 08.49 skrev Ron Arts:
>
>> Op 11-02-10 03:42, sean darcy schreef:
>>> Kevin P. Fleming wrote:
>>>> sean darcy wrote:
>>>>> I found out that the [globals] section in extensions.conf is ignored if
>>&g
sean darcy wrote:
> 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
Kevin P. Fleming wrote:
> sean darcy wrote:
>> I found out that the [globals] section in extensions.conf is ignored if
>> an #include 'd file has a [globals] section. Is this intended?
>>
>> In this particular case, the #include 'd file has a number of c
I found out that the [globals] section in extensions.conf is ignored if
an #include 'd file has a [globals] section. Is this intended?
In this particular case, the #include 'd file has a number of contexts
for googlevoice. I'd put various googlevoice variables in there to use
in all those cont
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
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
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
>&g
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 '
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] Answer("DAHD
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
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&qu
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", "Co
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 b
On Sun, Dec 6, 2009 at 12:28 PM, Thomas Perron 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 generally
> echo bo
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
&
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'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.
>
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 Teliax.
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 '' failed for
'10.10.11.98' - No matching peer found
So I changed the sip.conf context to [%201995]
Then:
[2009-11-19 20:44:28] WARNING[1
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 happ
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 prio
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
fax-tx-testESC[0
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
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]
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
defaultuser
On Tue, Sep 29, 2009 at 2:48 PM, David Backeberg wrote:
> On Mon, Sep 28, 2009 at 10:08 PM, sean darcy wrote:
>> On Mon, Sep 28, 2009 at 1:09 PM, David Backeberg
>> wrote:
>>> Have you tried using ps2tiff?
>> I looked up ps2tiff. That seems to be a windows program
On Mon, Sep 28, 2009 at 1:09 PM, David Backeberg wrote:
> On Mon, Sep 28, 2009 at 12:30 PM, sean darcy 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
>> a sp
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
>>
>> ${uniquefile}0
>>
>> Martin
> But I generate another unique file
On 1.6.0.16-rc1:
I'm using app_fax.so to send a fax, and then send a confirm.
'send' => 1.
Set(UniqueFile=/var/spool/asterisk/outgoing/call-${UNIQUEID}) [pbx_config]
2. System(env echo -e
"Channel:DAHDI/g0/\\nContext:fax-tx\\nExtension: s\\nPriority:
1\\n
nfiguration: single image plane
DocumentName: Standard Input
ImageDescription: converted PNM file
sean
>
> On Sat, Sep 26, 2009 at 8:05 PM, sean darcy wrote:
>> In my quest to actually send a fax, I'm now stuck trying to send the
>> confirm.
>>
>> First
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" >/var/spool/asterisk/outgoing/call-1254
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 transmi
n
>
> On Wed, Sep 23, 2009 at 7:45 PM, sean darcy wrote:
>> Martin wrote:
>>> well maybe it doesn't work as it should ... anyways like the other
>>> poster said that's not the way you use it ...
>>>
>>> either call the sendfax app direc
te app executable from dialplan ...
> But since 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
ay be
> specified.
>Otherwise, 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
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. SendFax(/var/spool/asterisk/fax/20090922_
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: ../staging/ech
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]
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, I'm
On Wed, Jul 15, 2009 at 5:23 PM, Wayne 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 don't get Asterisk
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] Dial("DAHDI
On Tue, Jun 16, 2009 at 10:13 AM, Miguel Molina 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
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]
exten
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 trai
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
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
> li
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 caller
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 inter
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 "DAHD
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"/1646xxxy
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
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. I'
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
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]
language=en
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 Network
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
> som
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 fin
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
Asterisk comes with ABSOLUTELY N
Philipp Kempgen wrote:
> sean darcy schrieb:
>> On 1.6.1, I must be losing my eyesight:
>
>> exten => _6000XXXNXXX,n,Set(Time-in-secs="${STRFTIME(${EPOCH},,%s}" )
>
> exten => _6000XXXNXXX,n,
On 1.6.1, I must be losing my eyesight:
[internal]
include => outbound-pstn
.
include => meetme; 2663
include => setup-meetme-conf-room ; 6000xxx
[setup-meetme-conf-room]
exten => _6000XXXNXXX,n,Set(Time-in-secs="${STRFTIME(${EPOCH},,%s}" )
CLI:
On Tue, May 12, 2009 at 3:05 PM, James A. Shigley
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: asterisk-users-boun...@li
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.
>
>
>
>>>> I do
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)
>
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
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
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/asteri
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.
>>
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
>> S
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-
>&
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 for conf-getconfno
>
> With 1.6.1, I'm trying to set up a test of m
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 co
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 hi
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 =
Rilawich Ango wrote:
> Can you try to disable call waiting in your phone?
>
> On Fri, May 15, 2009 at 6:44 AM, sean darcy wrote:
>> sean darcy wrote:
>>> I have two internal analogue extensions off a TDM400P. If the first is
>>> busy, I'd like to ring the se
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
`/home/asterisk/
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(${mainline},60)
> exten =>s,n,ExecIf($["${DIALSTATUS}"
I have two internal analogue extensions off a TDM400P. If the first is
busy, I'd like to ring the second. So:
[incoming]
exten =>s,1,Answer()
exten =>s,n,Dial(${mainline},60)
exten =>s,n,ExecIf($["${DIALSTATUS}" = "BUSY"]?Dial(${secondline},30))
But it doesn't work because * first tries Call Wai
David Backeberg wrote:
> On Mon, May 4, 2009 at 10:52 PM, sean darcy 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(&
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 sta
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'
[2009-04-26
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-b
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
>&
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
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
aste
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 Begin)
On Sat, Apr 11, 2009 at 12:04 PM, David fire 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?
> sorry for the
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
___
-- B
On Sat, Mar 28, 2009 at 11:25 PM, Zvonimir Mileta 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. nothing happens and
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