Hi Tzafrir,
Some more background...I have a comcast phone line
which I have connected to my FXO port. When I call my
number, it goes directly to comcast voicemailin other words,
there is no ringing tone and pickup by asterisk.
See inline below with NB...
On Sat, Jan 9, 2010 at 11:56
On Sun, Jan 10, 2010 at 12:25:10AM -0800, Nitin Bahadur wrote:
Hi Tzafrir,
Some more background...I have a comcast phone line
which I have connected to my FXO port. When I call my
number, it goes directly to comcast voicemailin other words,
there is no ringing tone and pickup by
| What's the output of:
dialplan show internal
in the asterisk CLI?
NB
jserver*CLI dialplan show internal
[ Context 'internal' created by 'pbx_config' ]
'_X.' = 1. Dial(Zap/1)
[pbx_config]
-= 1 extension (1 priority) in 1 context. =-
jserver*CLI dialplan show default
[
On Sun, 2010-01-10 at 00:25 -0800, Nitin Bahadur wrote:
Hi Tzafrir,
Some more background...I have a comcast phone line
which I have connected to my FXO port. When I call my
number, it goes directly to comcast voicemailin other words,
there is no ringing tone and pickup by asterisk.
Some more background...I have a comcast phone line
which I have connected to my FXO port. When I call my
number, it goes directly to comcast voicemailin other words,
there is no ringing tone and pickup by asterisk.
That would suggest the card is looping the line (busying it out).
On Sun, Jan 10, 2010 at 12:49:24AM -0800, Nitin Bahadur wrote:
| What's the output of:
dialplan show internal
in the asterisk CLI?
NB
jserver*CLI dialplan show internal
[ Context 'internal' created by 'pbx_config' ]
'_X.' = 1. Dial(Zap/1)
[pbx_config]
-= 1
On Sun, Jan 10, 2010 at 1:48 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Sun, Jan 10, 2010 at 12:49:24AM -0800, Nitin Bahadur wrote:
| What's the output of:
dialplan show internal
in the asterisk CLI?
NB
jserver*CLI dialplan show internal
[ Context 'internal'
Dear All
You are not willing to help me anymore ?
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Sunday, January 10, 2010, 11:24:22 AM, hadi wrote:
You are not willing to help me anymore ?
Why do you think this?
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On Sun, Jan 10, 2010 at 10:58 AM, Gergo Csibra csi...@gmail.com wrote:
Sunday, January 10, 2010, 11:24:22 AM, hadi wrote:
You are not willing to help me anymore ?
Why do you think this?
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Best regards,
Gergomailto:csi...@gmail.com
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why don't you post your question
On Sun, Jan 10, 2010 at 4:42 PM, hadi motamedi motamed...@gmail.com wrote:
On Sun, Jan 10, 2010 at 10:58 AM, Gergo Csibra csi...@gmail.com wrote:
Sunday, January 10, 2010, 11:24:22 AM, hadi wrote:
You are not willing to help me anymore ?
Why do you
ABBAS SHAKEEL wrote:
why don't you post your question
On Sun, Jan 10, 2010 at 4:42 PM, hadi motamedi motamed...@gmail.com
mailto:motamed...@gmail.com wrote:
On Sun, Jan 10, 2010 at 10:58 AM, Gergo Csibra csi...@gmail.com
mailto:csi...@gmail.com wrote:
Sunday, January 10,
On Sunday, January 10, 2010, Francesco Peeters wrote:
Yes, post your question clear and consicely, include all relevant
information and snip all unneccessary history.
Note that: no reply != not wanting to help...
It *is* obviously possible people just do not KNOW the answer!... (Oh
what
If xlite subscribes on a hint, and the phone is offline, xlite says so
(not online)
If SPA942 does the same, the led is green for available. The other
hints work: blink red for ringing and red for busy.
I seem to remember the led once showed amber for subscribed phone offline.
The SPA extended
Hello all,
I've noticed a few differences in my recent upgrade from 1.4 to 1.6.2 that
have me baffled. I thought I'd write to the list and see if anyone has any
ideas.
- In 1.4, app Directory matched users based on the name listed in their
voicemail.conf entry. Now it appears that 1.6 matches
Alternatively, since your iphone will have a web browser you could possibly
meet your needs going a web development route instead.
Asterisk and telephony is a very interesting project though.
On Thu, Jan 7, 2010 at 11:20 AM, UIT DEVELOPMENT uit...@gmail.com wrote:
Hello Tiago, I think that
Kevin P. Fleming schrieb:
Rick Green wrote:
'dash dash space CR'. A compliant MUA will strip that
line and everything after it when quoting for a reply or forward. Note
for the list admin: Please preceed your message-footer with a sigdashes
line!
Good idea, done!
A big thank you!
Anyone using the above mentioned SIP Gateway made by grandstream?
I would like to hear some feedback on real life experience using this gateway.
TIA
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Hi!
I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my
dialplan has an error but my extensions doesnt have the error that show me
asterisk.
I dont know from where asterisk take extension 8 and how is playing
ss-noservice because in my dialplan is not exist.
Any help
With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you
enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF}
is [un]set in an odd way.
for example consider:
999,1,Swift(some long message that you dont want to wait for|5000|5)
999,n,NoOp(DTMF: ${SWIFT_DTMF})
if while
On Sun, Jan 10, 2010 at 1:17 PM, C F shma...@gmail.com wrote:
Anyone using the above mentioned SIP Gateway made by grandstream?
I would like to hear some feedback on real life experience using this gateway.
I have a few that I used for about 2 days before I replaced them with
AudioCodes
Hi,
I am seeking help with the installation of a Soundpoint 650 desk phone.
Although I have some experience (and a good one! no single issue so
far, besides the problem I am trying to solve...) installing a few SP
320/330 units, I am having several issues with my first SP 650.
Polycom SP
Bon journo Aldo.
I am having several issues with my first SP 650.
* Assembly: 2345-12600-001 Rev.G
I have deployed more than 200 IP650 with the same assembly as yours and
so far there are no problems.
The first thing I have noticed is that I was not able to upgrade the
unit's
The 8 probably comes from the T1, does the telephone number end with an 8?
The playback of ss-noservice might be a fallback ensuring that
*something* happens when a call comes in
On Sun, Jan 10, 2010 at 1:31 PM, Edwin Quijada
listas_quij...@hotmail.com wrote:
Hi!
I have an T1 line for using
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hi,
i want to use $agi - exec_dial() to dial .
this is in extention.conf
[tutorial]
exten = 1234,1,Dial(SIP/ivan)
is that i use
$agi - exec_dial(SIP,tutorial|1234|1)
can dial ?
BTW, i want to know some turorial on how to use PHPAGI funtions? can
you tell me some?
Thanks!
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Best
hi, all
i want to test if a telephone is busy now in agi php script?
Could you tell me how to do that judgement？
example：
if( ivan is not busy)
{
$agi - exec_dial(SIP,ivan);
}
else if (test is not busy)
{
$agi - exec_dial(SIP,test);
}
Thanks very much!
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Best regards,
Sucan
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On Sun, Jan 10, 2010 at 2:43 PM, Geoff Lane ge...@gjctech.co.uk wrote:
On Sunday, January 10, 2010, Francesco Peeters wrote:
Yes, post your question clear and consicely, include all relevant
information and snip all unneccessary history.
Note that: no reply != not wanting to help...
It
At 15:33 1/7/2010, Tzafrir Cohen wrote:
On Thu, Jan 07, 2010 at 12:50:03AM -0600, Doug wrote:
At 00:22 1/7/2010, Tzafrir Cohen wrote:
On Wed, Jan 06, 2010 at 11:41:54PM -0600, Doug wrote:
At 16:49 1/5/2010, Tzafrir Cohen wrote:
On Tue, Jan 05, 2010 at 04:24:37PM -0600, Doug wrote:
On 01/10/2010 11:38 PM, hadi motamedi wrote:
FWIW, he did post his question yesterday. I've just taken a look and
one potential issue I've spotted is that the external server he
mentions is 192.168.0.139, which is part of the 192.168.0.0/16
http://192.168.0.0/16
netblock
in your dialplan ,did you add area code automaticly? when dial out.
2010/1/11 hadi motamedi motamed...@gmail.com:
On Sun, Jan 10, 2010 at 2:43 PM, Geoff Lane ge...@gjctech.co.uk wrote:
On Sunday, January 10, 2010, Francesco Peeters wrote:
Yes, post your question clear and consicely,
you'd better paste your dialplan snip here, in order to get specific help.
2010/1/11 Darrick Hartman dhart...@djhsolutions.com:
On 01/10/2010 11:38 PM, hadi motamedi wrote:
FWIW, he did post his question yesterday. I've just taken a look and
one potential issue I've spotted is that
On Mon, Jan 11, 2010 at 6:23 AM, Zhang Shukun bit...@gmail.com wrote:
you'd better paste your dialplan snip here, in order to get specific help.
2010/1/11 Darrick Hartman dhart...@djhsolutions.com:
On 01/10/2010 11:38 PM, hadi motamedi wrote:
FWIW, he did post his question
On 1/10/2010 5:33 PM, Jeremy Kister wrote:
With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you
enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF}
is [un]set in an odd way.
The problem lies within f-subclass inside the else if of line 436.
the code seems to
Hi,
I recently upgraded our asterisk server from some 1.4 version to version
1.6.0.15. From this point on my AGI scripts aren't working anymore, here
is a simple example:
[isdin]
exten = 83086921,1,AGI(test.php)
exten = 83086921,2,NOOP(MARKE1)
exten = 83086921,3,WAIT(2)
exten =
On 1/10/2010 5:33 PM, Jeremy Kister wrote:
With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you
enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF}
is [un]set in an odd way.
I fixed it up by ignoring the f-subclass and starting the
dtmf_listener right away.
http://lists.digium.com/pipermail/asterisk-users/
Trusting user-generated date fields? sweet. :D
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Jeremy Kister
http://jeremy.kister.net./
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