ppens in the CDR
8. 2004-09-14 22:39:00 SIP/2208-a5dd 2208"2208" <2208> NoOp
s ANSWERED10
Any ideas?
Thanks de Mark
--
Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com/
___
ton on every member phone that lights up when line 5 rings.
Then if the call is for Fred rather than me, Fred can press the line 5
button and take the call. This is working on our Lucent currently
Thanks folks.
--
Mark Phillips, G7LTT/KC2ENI
Randolph
s_thread: CallerID feed
failed: Success
Jul 12 16:40:53 WARNING[15374]: chan_zap.c:4980 ss_thread: CallerID
returned with error on channel 'Zap/1-1'
-- Executing Zapateller("Zap/1-1", "nocallerid") in new stack
Even when there is caller ID on the analogue cal
//lists.digium.com/mailman/listinfo/asterisk-users
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>> >
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>> Asterisk-Users mailing list
>&g
Before you all reply that its available via Cisco, I'm not qualified to be
a tech member according to Cisco.
I just bought 4 7960's with which to use with * and I want to load up the
SIP image into them.
Does anyone have it that they can make available to me please?
Thanks
--
Mar
nk you.)
exten => 555,7,Wait(1)
exten => 555,8,Hangup
Thanks again de Mark
> On Thu, 2004-05-13 at 13:41, Mark Phillips wrote:
>> Those of you whom have a free Washington State phone number from
>> ipkall.om
>> will know that one has to use the number at least every 30 da
that I can program asterisk to make a call to my WA numbers
so that they wont get disco'd? I'm thinking of something like a "alrm
call" that one has in a hotel room. YOu pick up the phone and program a
ring back time.
Hope this make sense.
Thanks
G7LTT/
I'm using the guest account that comes in the default setup files and I
don't register any of the machines.
> On Wed, 2004-04-21 at 17:31, Mark Phillips wrote:
>> Hi all,
>>
>> I have 3 * boxes all running the same OS and software version. Machine A
>> has a
B and C require X100P cards
before IAX2 will work correctly? I don't think this is the case because
the call can be passed to them after it has been setup via A.
Its really bugging me. Any ideas?
G7LTT/KC2ENI
Mark Phillips
___
Asterisk-Users ma
icked up but none that could
be done dynamicly.
What I'd like to be able to do is press a button and have * start
recording the call from that moment and then either stop when I hang up or
stop when I press another button.
Ideas?
G7LTT/KC2ENI
Mar
ng list
> [EMAIL PROTECTED]
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>
G7LTT/KC2ENI
Mark Phillips
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sterisk-Users mailing list
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G7LTT/KC2ENI
Mark Phillips
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A
y Hwy.phone: 1-888-941-3282, 1-205-335-8589
> Birmingham, AL 35216 fax: 1-205-823-7838
> ***
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> T
ailing support
crap. http://www.galaxyvoice.com. Say the British guy sent you.
Enough for now
G7LTT/KC2ENI
Mark Phillips
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has a chance to sort itself out before
the tones play?
> On Mon, 12 Apr 2004, Mark Phillips wrote:
>> I tried,
>>
>> exten => s,1,Zapateller(answer|nocallerid)
>> exten => s,2,Privacymanager
>> exten => s,3,Dial(a bunch of SIP extensions)
>>
>
; s,2,Privacymanager
exten => s,3,Dial(a bunch of SIP extensions)
But then every call was answered regardless of CID and the tones were heard.
Any ideas?
G7LTT/KC2ENI
Mark Phillips
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[EMAIL PROTECTED]
http://l
be ztdummy
>
>
>>Thanks,
>>
>>-Steve
>>
>>
>>
>
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>http://lists.digium.com/mailman/listinfo/asterisk-
id work but now it doesn't.
> Asterisk does show that it is playing the file but no audio is heard.
>
> I have audio on regular SIP based calls as well as IAX based ones. I'me
> not getting and audio when I make a ZAP call.
>
> Ideas?
>
&
Is it CAPI compliant? if so yes
>
> Is there any Linux/* support for the TigerJet ISDN card?
>
> -brian
>
G7LTT/KC2ENI
Mark Phillips
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[EMAIL PROTECTED]
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sword. This did work but now it doesn't.
Asterisk does show that it is playing the file but no audio is heard.
I have audio on regular SIP based calls as well as IAX based ones. I'me
not getting and audio when I make a ZAP call.
Ideas?
Mark
G7LTT/KC2ENI
Mark Phillips
G
n => s,5,Answer
exten => s,6,Wait(1)
exten => s,7,Playback(new/hello)
exten => s,8,Playback(new/marisa-john-not-in-momnt)
exten => s,9,Playback(new/theyre-rattlesnake-wrstling)
exten => s,10,Voicemail(u${PHONE1VM})
exten => s,11,Hangup
exten => s,108,Wait(2)
exten => s,109,Voic
Grandstream phones and on the Cisco ATA186 but not with either my Pulver
WiSIP or X-Ten Pro (yes I did register it and no you can't have a copy)
softphones. I'm also having problems with DIAX which uses IAX.
Ideas?
--
Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
http://www
exten => #,2,Hangup
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
Whilst writing this I've had a thought. What would happen if I had an
entry like this?
; transfer to regular extension
;m guessing that the externip thing is starting to work which in turn is
why I think this is my FWD problem/solution.
Is there a way of defining more than one localnet and localmask? Perhaps
something like;
localnet=38.249.233.0,192.168.0,0
localmask=255.255.255.0,255.255.255.0
Folks?
--
Mark Ph
ternip=63.88.139.198
>
> David
> ----- Original Message -
> From: "Mark Phillips" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Thursday, March 18, 2004 3:16 PM
> Subject: [Asterisk-Users] Problems with FWD
>
>
>> Hi Folks,
>&
anreinvite=no
disallow=all
allow=gsm
allow=ulaw
mailbox=3409
My machine is behind a Checkpoint firewall. Its public address
63.88.139.198; private address is 192.168.18.65. All the normal ports are
open. 5000-6000 & 1-2.
Ideas?
Mark
--
Mark Phillips, G7LTT/KC2ENI
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>
>
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t;
> Does anyone know how to enable Dual line support, Hold and Transfer
> functions with this phone via Asterisk.
>
> Thanks,
>
> Regards,
>
> Steven Thomas
>
>
--
Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com/
_
Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of Mark Phillips
>> Sent: Tuesday, March 16, 2004 3:04 PM
>> To: [EMAIL PROTECTED]
>> Subject: [Asterisk-Users] Anyone got their Pulver WiSIP phone working
>> with
>
gestion
If I dial 71612 I expect to get the time service at FWD. I get connected
but I hear nothing.
I have firewall holes for the following ports
5060
5082
1-2
Any ideas?
--
Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com/
_
away.
If I call the phone by its URI it rings but the calling phone never thinks
the call has been answered by the WiSIP.
I have 711 as the codec and inband DTMF (which the WiSIP thinks should be
2833).
Any ideas ...
Thanks
--
Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
had a chance to set it all up yet :-(
>>
>>
>>
> Nildram also support bonding ADSL lines together, I think they currently
> support up to 4 lines (1Mb upstream 2Mb downstream) and they are looking
> at supporting more..
>
> Later..
>
>
> _
Hi Folks,
Rather than have my hold music play from a sound file I'd like to have a
live feed from a sound card input or MP3 stream. Is this doable and if so
how?
--
Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com/
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>
--
Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com/
_
ter.
Any ideas would be greatly appreciated.
--
Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com/
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b/modules/2.4.22-10mdk-p3-smp-64GB/misc/ztd-eth.o
depmod: *** Unresolved symbols in
/lib/modules/2.4.22-10mdk-p3-smp-64GB/misc/ztdynamic.o
Any ideas anyone?
--
Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com/
___
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esterdays CVS build of *, Linksys router with ports
5000-5100 & 1-2 forwarded to the * host.
All other things work like FWD, IAXTel and IPTel.
Any ideas?
Thanks
--
Mark Phillips, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com/
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