Am Mittwoch 21 Dezember 2011, 07:04:03 schrieb Matt Hamilton:
I have a queue that distributes calls among 3 phones. When a phone is in
use (including on hold), queue skips that device and sends the call to the
next available one as expected. On the other hand, if a call comes in
while one of
Hi list,
Is there any way in asterisk by which I make a call from server and then
dialplan(IVR system) gets DTMF from it. I mean to say that automatically
DTMF is sended by channels as per user defined,
I read there is an application sendDTMF but I don't know how we can used it?
like A script
Create a callfile with local channel and once first call leg is answered,
use wait() and senddtmf() application on second call leg.
CALLFILE sample:
Channel: LOCAL/1234\@test_ivr
Context: senddtmf
Extension: s
Priority: 1
Extensions.conf sample:
;-- FIRST LEG CALL --;
[test_ivr]
exten =
Hi Satish,
Thank you Satish. I did the same before your e-mail i saw. But i got
another issue in such case.
DTMF is passed to that channels but in case I will make the complete IVR
system for calling server end. and which become so complected to do it.
Is there any alternate way by which I get
Hi,
You can use combination of SendDTMF() and wait() in such a way that you
traverse through the IVR tree just as Satish mentioned.
SendDTMF(1)
Wait(3)
SendDTMF(2)
Wait(2)
SendDTMF(5678123490)
See also:
*WaitForNoise()* , WaitForSilence(), AMD()
Regards,
Sammy.
On Wed, Dec 28, 2011 at 2:32
On 28 December 2011 03:02, Joseph syscon...@gmail.com wrote:
No, it makes no difference, on the other end is asterisk 1.4.39
and 1.8.8 is still giving me:
Executing [4@internal:1] Dial(SIP/11-0003,
IAX2/home_server:@192.168.141.1/4,30,rw) in new stack
-- Called
Hi List,
I would like create a H323 trunk from Avaya IPOFFICE to Asterisk, but i
would like activate a direct media path for the RTP transit directly
between the phone and the Asterisk.
Now,
- H323 Trunk is OK
- RTP from the phone transit directly to Asterisk (activate strictrtp=no
in rtp.conf,
What I understand from your reply is, you also like to have multiple Read()
in 'support' and 'help' extensions as well.
In that case you can have something like this in [senddtmf]
exten = s,1,Noop(# TEST:IVR ##)
; We should wait atleast 'n' of seconds.
Hi team,
Can any one share with me clustering configuration file SS7.conf for single
pointcode with four slc. two different machine each host having 2 slc
respectively.
Thanks
Vinod Dharashive
Sent from BlackBerry® on Airtel
--
Hi,
I do not know, whether this is the best way to use TESTTIME function, but for
me it is working in that way:
exten = 123,n,Set(TESTTIME()=2011/12/25 18:30:00 Europe/Vilnius)
OR
You can use this:
Set(__TESTTIME=${STRPTIME(2011-12-25 18:00:00,Europe/Vilnius,%Y-%m-%d
%H:%M:%S)})
Best regards,
Hi,
Thanks for replying.
I'm afraid this :
[foobar]
exten = 123,1,Verbose(0,Into context ${CONTEXT})
exten = 123,n,Verbose(0,Time is ${STRFTIME()})
exten = 123,n,Set(TESTTIME()=2011/12/25 18:30:00 Europe/Vilnius)
exten = 123,n,Verbose(0,Time is ${STRFTIME()})
exten = 123,n,HangUp()
... gives
On 11-12-28 03:25 AM, virendra bhati wrote:
Hi list,
Is there any way in asterisk by which I make a call from server and then
dialplan(IVR system) gets DTMF from it. I mean to say that automatically
DTMF is sended by channels as per user defined,
I read there is an application sendDTMF but I
On 12/28/11 10:07, Steve Davies wrote:
On 28 December 2011 03:02, Joseph syscon...@gmail.com wrote:
No, it makes no difference, on the other end is asterisk 1.4.39
and 1.8.8 is still giving me:
Executing [4@internal:1] Dial(SIP/11-0003,
IAX2/home_server:@192.168.141.1/4,30,rw) in new
On 12/28/11 10:45, Joseph wrote:
[snip]
Have you tried enabling IAX2 debug at both ends to see if the packet
decode provides any more clues?
Regards,
Steve
I've enabled iax2 debug on both ends and
on asterisk-1.4.39 I get:
NOTICE[2412]: chan_iax2.c:9541 socket_process: Rejected connect
Thanks Sebastian. It was a phone related issue. Factory resetting the phones
and reconfiguring them fixed it. It probably was a CW issue as you suggested.
I think it is up to your phones to allow only one concurrent session,
you could check call-waiting is deactivated on your phones?!
If
Hi everyone,
I see that there was a bug in version 1.8.5.x and people were advised to
move to 1.8.7.1 but now I have 1.8.7.1 and experiencing the same problem.
Here is the output:
*chan_sip.c: Asked to transmit frame type ulaw, while native formats is
0x100 (g729) read/write = 0x100 (g729)/0x100
The issue is not fixed in 1.8.8.0 either.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Wednesday, December 28, 2011 3:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Hi All,
I am trying to record Call, but when the call is done I have one file but the
conversation inside it is separate into calls conversation and receiver
its single file but separate recording,
How can I make it mixed together so the conversation will be normal?
Thanx
--
Suggestion 1 - mixmonitor instead of monitor
Suggestion 2 - SOX.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 2:16 PM
To: asterisk-users@lists.digium.com
This might or might not help, but here is the offending code in 1.8.8
case AST_FRAME_VOICE:
if (!(frame-subclass.codec ast-nativeformats)) {
char s1[512], s2[512], s3[512];
ast_log(LOG_WARNING, Asked to transmit frame type
%s,
I installed SOX( it was not installed before). Will that solve my problem?
if not what are the parameter for the mixMonitor Command
this is how I use Monitor
exten=6500,2,Monitor(wav,${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S)}_${SIP_HEADER(email)},m)
is Mix Monitor will have the
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, December 28, 2011 3:19 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk
According to the monitor documentation, the format you specified should be
calling SOX and mixing on call completion. What versions of SOX and
Asterisk are you using?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Asterisk 1.6.2 but sox I don't know but now it is the latest version, my
problem is not mixing It's the same file but inside that file two seperate
records first callers then reciever
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٤٦ م, Danny Nicholas da...@debsinc.com wrote:
According to
Can you post a CLI output of the Monitor output? I'm supposing that something
in your $(STRFTIME) string might be eating the M option.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent:
I ran into a rare situation today.
A really short message is being played over the ALSA or console channel
from one asterisk box to another. Both running 1.4.30.
the incoming context on the ALSA or Console port box first runs an AGI
before connecting the audio path.
The AGI got hung up for an
On 12/02/2011 11:37 AM, Anthony Messina wrote:
I've just connected my new Android (Motorola RAZR) phone to Asterisk
using CSipSimple and have discovered that on any call between CSipSimple
and an Asterisk DAHDI or IAX2 channel, the 'other' end of the call will
hear a rhythmic tapping as if my
Can u plz tell me how , I forgot how to run asterisk cli
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١٠:٥٢ م, Danny Nicholas da...@debsinc.com wrote:
Can you post a CLI output of the Monitor output? I'm supposing that
something in your $(STRFTIME) string might be eating the M option.
Asterisk -vvvrc
Is how you would get it live
After the fact you might find it in /var/log/asterisk/full
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:08 PM
Can somebody point me to an explanation from Kevin or Tzafir or someone else
up the food chain explaining the differences/benefits of 1.6/1.8 vs
1.4/10.0?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric
My call happens with a queue , there is no full file but there is queue and
queue is useless, can u give me unix command to search all log files and print
moniter line?
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١١:٠٩ م, Danny Nicholas da...@debsinc.com wrote:
Asterisk -vvvrc
Is how you would
I already searched using grep for the monitor word ... It doesn't exists
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١١:١٥ م, Faraj Khasib fkha...@iconnecths.com wrote:
My call happens with a queue , there is no full file but there is queue and
queue is useless, can u give me unix command to
Even using Queue there should still be a /var/log/asterisk/full that records
the Monitor then the following Queue/Dial commands. What is in your
/var/log/asterisk?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
see attached ...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, December 28, 2011 3:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial
I would wager that your setup dumps what would normally be in /v/l/a/full into
/v/l/a/messages
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:20 PM
To:
So, what is really the effect of this and why is it hard to fix? Does this
bug disrupt processing the call? I see the log filled up with this error. I
do have a BUSY showing on forwarding to a number outside and that is what
concerns me. Not sure if caused by this bug. From reading CHANGES log, I
but i tiried these commands and I didnt find anything about Monitor
[root@c-24-1-71-68 asterisk]# grep -R 'Monitor' *
[root@c-24-1-71-68 asterisk]# grep -R 'monitor' *
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com]
On 12/28/2011 03:10 PM, Danny Nicholas wrote:
Can somebody point me to an explanation from Kevin or Tzafir or someone else
up the food chain explaining the differences/benefits of 1.6/1.8 vs
1.4/10.0?
Every branch (1.0, 1.2, 1.4, 1.6.0, 1.6.1, 1.6.2, 1.8, 10) of Asterisk contains
new
Calls to long distance get disconnected before answer.
Telco: Alestra
Country: Mexico
System: Elastix 2.2
Digital Card: Digium TE122
Log:
[Dec 28 14:37:44] VERBOSE[4586] pbx.c: -- Executing
[+525552622900@default:1] Set(SIP/OCS_TRUNK-01bf, EXT=015552622900) in
new
Try
# grep 'onitor' /var/log/asterisk/messages
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, December 28, 2011 3:25 PM
To: Asterisk Users Mailing List - Non-Commercial
I understand the end of life issue. What I fail to understand is that if
1.8 is the Cadillac of Asterisk, why did they make 10.0 and why does 1.8
have so many bugs (just what I read here, not from my actual experience)?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
It got stuck ...
Sent from my iPhone
On ٢٨/١٢/٢٠١١, at ١١:٢٩ م, Danny Nicholas da...@debsinc.com wrote:
Try
# grep 'onitor' /var/log/asterisk/messages
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf
The UPGRADE*.txt files included with the Asterisk tarballs give a nice summery
of the major changes between each Asterisk verison.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker
Sent: Wednesday,
Hi,
This function sets TESTTIME global variable and if TESTTIME variable is set,
then GoToIfTime use time from this variable.
On 2011.12.28, at 17:28, Olivier oza_4...@yahoo.fr wrote:
Hi,
Thanks for replying.
I'm afraid this :
[foobar]
exten = 123,1,Verbose(0,Into context ${CONTEXT})
I attached log, but there is nothing unusual in it ...all normal ...
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com]
Sent: Wednesday, December 28, 2011 4:06 PM
To: Faraj Khasib
Subject: Your message to
Hey All,
Odd thing. I am just trying to return the whole date time stamp from a
SMALLDATETIME field in a MS SQL server.
func_odbc.conf = readsql=SELECT DateCreated FROM [REDACTED] WHERE Code
= '${ARG1}'
Problem is I only get the first 15 back from the field. Like so...
Connected to
Un-top-posting, snarky comments inline...
On Wed, 28 Dec 2011, Faraj Khasib wrote:
I am trying to record Call, but when the call is done I have one file
but the conversation inside it is separate into calls conversation and
receiver its single file but separate recording, How can I make
Hello,
Do you use monitor?, because in asterisk 1.4 to new versions, It's use
mixmonitor, in asterisk 1.2 had this mistake.
Regards
On Wed, Dec 28, 2011 at 10:11 PM, Steve Edwards
asterisk@sedwards.comwrote:
Un-top-posting, snarky comments inline...
On Wed, 28 Dec 2011, Faraj Khasib
I just realized there is no IP (host) in the message line, so no way for
fail2ban to catch it.
Other suggestions? Or will I have to code something into my dialplan
From: asterisk-users-boun...@lists.digium.com
Here is more of a SIP debug log:
As you can see Asterisk retries four times but I assume the softphone is not
responding?
---
Really destroying SIP dialog
'637b0e9777c88caa16a5a70b5a8984fe@172.31.253.4'mailto:'637b0e9777c88caa16a5a70b5a8984fe@172.31.253.4'
Method: OPTIONS
Reliably
The BB is using wifi, on the same subnet as the asterisk server so no need for
NAT.
There is no keep alive option on the softphone (very simplistic settings)
Thanks
--
_
-- Bandwidth and Colocation Provided by
I happened to be in the cli tonight as some (208.122.57.58) initiated a simple
attack - just trying to make long distance calls from outside context.
Although harmless, this went on for several minutes as the idiot just used up
my bandwidth with SIP messages. Here's and example:
[2011-12-28
Yes fail2ban is working fine. I did NOT have a filter for the rejected
because extension not found line yet (I'm still working on it). Hoping for
input on the regex.
Thanks
From: asterisk-users-boun...@lists.digium.com
On Wed, Dec 28, 2011 at 4:33 PM, Danny Nicholas da...@debsinc.com wrote:
I understand the end of life issue. What I fail to understand is that if
1.8 is the Cadillac of Asterisk, why did they make 10.0 and why does 1.8
have so many bugs (just what I read here, not from my actual experience)?
On 29 December 2011 12:07, Michelle Dupuis mdup...@ocg.ca wrote:
I thought that it might be worth adding a line to my fail2ban filter, but am
looking for a hand with the regex. I have come up with:
NOTICE.* .*: Call from '' to extension '.*' rejected because
extension not found
Hello,
Do you set up, your logrotate in /etc/asterisk ?
Do you test that your fail2ban work fine?
Regards
On Wed, Dec 28, 2011 at 11:07 PM, Michelle Dupuis mdup...@ocg.ca wrote:
I happened to be in the cli tonight as some (208.122.57.58) initiated a
simple attack - just trying to make long
I have a softphone I'm trying on a blackberry, that registers on my Asterisk,
can make outgoing calls, but can't receive calls.
There is very little traffic with this phone (see debug below - as the phone
registers), and sip show peers confirms it is unreachable.
Any suggestions? Is this just
Hello,
try to configure keep alive option on Softphone if there is.
Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Hello,
Your blackberry sip client, works in your wifi network? or by blackberry
internet?
do you set nat=yes if your phone, register by internet?
What is your sip.conf?
Regards
On Wed, Dec 28, 2011 at 11:16 PM, Michelle Dupuis mdup...@ocg.ca wrote:
I have a softphone I'm trying on a
On Wed, 2011-12-28 at 23:16 -0500, Michelle Dupuis wrote:
I have a softphone I'm trying on a blackberry, that registers on my
Asterisk, can make outgoing calls, but can't receive calls.
There is very little traffic with this phone (see debug below - as the
phone registers), and sip show
I have been running 1.8.7 with a few fixes back ported from the 1.8.8
release candidate for the last 2.5 months. The system processes around
4,000 calls per day over PRIs for 250 Polycom phones.
Previously I was running 1.6.1.18 with a bunch of back ports for fixes and
features. Overall it
Hi Michelle,
I just realized there is no IP (host) in the message line, so no way for
fail2ban to catch it.
Probably my understanding is limited, but it seems to me that they
have already 'access' to your Asterisk for them to be able to try to
make outgoing calls. Wouldn't it be better to
You mentioned the IP, 208.122.57.58, where did you get that from?
Following are the default for Asterisk 1.8 (It would be great to have
others input on this to strengthen this part of the filter):
failregex = Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' -
Wrong password
OK !
But AEL2's ifTime keyword do not use it, does it ?
2011/12/28, Mindaugas Jasiulis mindaugas.jasiu...@mediafon.lt:
Hi,
This function sets TESTTIME global variable and if TESTTIME variable is set,
then GoToIfTime use time from this variable.
On 2011.12.28, at 17:28, Olivier
AEL2 ifTime use it too. AEL2 ifTime become the same GoToIfTime in the dialplan
:)
On Dec 29, 2011, at 8:40 AM, Olivier wrote:
OK !
But AEL2's ifTime keyword do not use it, does it ?
2011/12/28, Mindaugas Jasiulis mindaugas.jasiu...@mediafon.lt:
Hi,
This function sets TESTTIME global
I originate calls from .call file and 1 channel I have at A server A and
another channel at B server.
*A server code is below:-*
exten = 43689956,1,Answer()
same = n,Wait(5)
same = n,SendDTMF(1)
same = n,NoOp(== ${CHANNEL(state)}== state)
same = n,wait(2)
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