hello
i want to kick participant in a meeting by pressing the digit on sip
phone.when i entry the meeting ,no matter how i press the button,the dtmf does
not work.
here is my dialplan and my agi script,and sip.conf
[from-internal]
exten =121,1,MeetMeCount(900,CONFCOUNT)
exten
hello
i want to kick participant in a meeting by pressing the digit on sip
phone.when i entry the meeting ,no matter how i press the button,the dtmf does
not work.
here is my dialplan and my agi script,and sip.conf
[from-internal]
exten =121,1,MeetMeCount(900,CONFCOUNT)
exten
I wanna connect proxy server.
my IP Phone - my asterisk - service provider's proxy server - extern PSTN
phone
but asterisk server can't register to proxy server.
I think that configuration is right.
When asterisk send to register request, proxy server don't response.
I did
HI
I am Using asterisk-1.6.0.5
i cannot originate call from AMI interface here are my Originate action
Packet
Action: Originate
Channel: SIP/111
Context: default
Exten: 8135551212
Priority: 1
Callerid: 3125551212
Timeout: 3
Variable: var1=23|var2=24|var3=25
ActionID: ABC45678901234567890
Hi All,
Is it possible to make a IAX2 connection between asterisk 1.6.1.0 , and
asterisk 1.2.14 ?
i tried to use a IAX2 connection between version 1.2.14 and 1.6.1.0 but
it gave an error -
1.2.14 End - Error Msg
WARNING[8313]: chan_iax2.c:7103 socket_read: Call rejected by
147.120.203.71:
Hi,
I use realtime and I found that changing accountcode needed to
restart asterisk to activate that code and shown in CDR. Does it has
a way to update accountcode without restart asterisk?
ango
___
-- Bandwidth and Colocation Provided by
I am using 1.4.24 with realtime.
On Fri, May 29, 2009 at 5:21 PM, Rilawich Ango maillist...@gmail.com wrote:
Hi,
I use realtime and I found that changing accountcode needed to
restart asterisk to activate that code and shown in CDR. Does it has
a way to update accountcode without restart
Hi all,
If anyone interested, it is a DUNDI bug which makes Asterisk crash with
a segfault: disabling dundi fix the problem.
Giorgio Incantalupo wrote:
Hi all,
I was playing with top on my Asterisk 1.4.24 server when I noticed
this strange thing:
PID USER PR NI VIRT RES SHR S
2009/5/29 김무성 ki...@infosec.co.kr
I wanna connect proxy server.
my IP Phone - my asterisk - service provider's proxy server - extern
PSTN phone
but asterisk server can't register to proxy server.
I think that configuration is right.
When asterisk send to register request, proxy
Asterisk versions may differ. I do IAX trunk successfully even
between Asterisk 1.0.2 and 1.4.xx
please post your Dial command.
On Fri, May 29, 2009 at 11:33 AM, Tharanga thara...@roomsnet.com wrote:
Hi All,
Is it possible to make a IAX2 connection between asterisk 1.6.1.0 , and
asterisk
On Thu, May 28, 2009 at 12:10 AM, David Backeberg dbackeb...@gmail.com wrote:
On Tue, May 26, 2009 at 8:46 PM, Mikel Lindsaar raasd...@gmail.com wrote:
Does anyone know of a way to have tones played during the call
progress stage of the call?
You could detect what was dialed and route
accountcode is a setting you add to your SIP peer.. so it doesn't require
restarting Asterisk.. only restart the SIP module..
sip reload will be enough my friend..
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE, RHCE, VoIP
Syria: +963 944 618286
USA: +1 347 562 2308
Date:
You could do this with a class specific Music on hold.
- exten = s,1,SetMusicOnHold(mytone)
- exten = s,1,Dial(tech/1,,m)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mikel Lindsaar
Sent: Friday, May 29,
i'm not so familiar with what youa re talking about .. but i beleive i've seen
something like that in FreePBX where you can setup a failover trunk for a
context.. try to have a look at it. and i hope it's what you are looking for
--
AHD Tarek Sawah
Integrated Digital Systems
CCNA, MCSE,
some how the extension you have identified in your extensions.conf file is
wrong..
you are forwarding your call to an extension @ a local extension??
you can try at least the following
[default]
exten = _X.,1,Dial(SIP/${ext...@proxy.sp.co.kr)
it may work .
let me know
--
AHD Tarek Sawah
On Thu, 2009-05-28 at 19:38 -0500, Andres Gomez wrote:
Please help me, i need transfer a call in asterisk to other telco
number and free the channel. Can i do with any q931 function?.
Asterisk will automatically attempt a Two B-channel Transfer if the
following conditions are met:
1) You must
I beg to differ with you Jared. Since I don't have your email, I'll post
this here. Create this call file.
Channel: DAHDI/g1/5551212
CallerID: SIP/104
MaxRetries: 1
WaitTime: 60
retryTime: 5
Application: playback
Data: /var/lib/asterisk/sounds/you-sound-cute
On my test machine using a TDM400P,
Hi,
In extensions.ael, I can use ${CHANNEL(key)} to get value associated with
key in current channel.
How can I get the value associated with a in another channel ?
Something like ${CHANNEL(channel-name,key)} ?
Regards
___
-- Bandwidth and Colocation
Greetings all, I have an interesting problem I am trying to work around.
I currently have 2 * servers running in separate offices, using IAX2 to trunk
between them, and queues in our main office. I'll call them Office_A and
Office_B. I use Polycom 501s with a primary and secondary server, the
Hi,
How can you add specific statements into Asterisk dialplan (extension.ael,
...) for attented transfers ?
I can see Asterisk sending Transfer or Masquerade events through AMI (in
1.6.1) but I could use an external program to catch those events but I would
prefer to use dialplan instead.
Any
I'm pretty sure that attended transfer is a features function, not a
dialplan one.
On my system I do *2 and asterisk says transfer, then I punch in the new
extension.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Please, does anybody have a good document describes well
the optimum method to achieve Asterisk Redundancy/Clustering on 2 servers.
Thanks.
_
Invite your mail contacts to join your friends list with Windows Live Spaces.
On Fri, May 29, 2009 at 4:22 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
i cannot originate call from AMI interface here are my Originate action
Packet
Channel: SIP/111
where 111 Is my SIP phone number which registered with my asterisk server
I can login with this manager User and
sean darcy wrote:
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,
Tzafrir Cohen wrote:
On Fri, May 08, 2009 at 12:11:40PM -0400, Mike van der Stoop wrote:
I have a Sangoma b600de analog card using dahdi 2.1.0.4 and I get the
following results (same dialplan, config etc):
Asterisk 1.6.0.1 = works fine
Asterisk 1.6.0.9 = can't dial out unless I dial
Hello,
I am trying to make some test in connecting 2 asterisk with trunk.
I managed to call the 2 asterisk eachother via IAX trunk.
The problem is, i have a pri trunk over servera and i want to enable to
call that trunk via serverb too.
When i try to call i got the following error on servera
2009/5/29 robert aofeish...@163.com:
i want to kick participant in a meeting by pressing the digit on sip
phone.when i entry the meeting ,no matter how i press the button,the dtmf
does not work.
Does dtmf work anywhere?
What happens if you try to have the call navigate an IVR?
You
On Fri, May 29, 2009 at 1:22 AM, Tharanga thara...@roomsnet.com wrote:
I managed to register my phone on asterisk. but i cant hear any dial
tone on my phone. these are my configs. it will detect incoming calls
and transfer the call to ext 312. but sip phone users voice is not
clear..., but
I'm not sure if this posting will go to the correct thread or not, as I
am subscribing to make this post, and don't have a message to reply to.
Hose, if this does not end up in the thread can you post in in there?
I am getting the same DAHDI error, under Asterisk 1.4.25, libpri-1.4.10
using a
On Fri, May 29, 2009 at 1:22 AM, Tharanga thara...@roomsnet.com wrote:
I have installed asterisk latest stable version 1.6.1.0, with dahdi
driver (tdm410p). then i try to use my older 1.4 extensions.conf. . now
it wont work with 1.6.
You pasted in sip.conf and dahdi config, but not your
On Fri, May 29, 2009 at 1:22 AM, Tharanga thara...@roomsnet.com wrote:
I managed to register my phone on asterisk. but i cant hear any dial
tone on my phone.
What kind of phone? What kind of channel?
___
-- Bandwidth and Colocation Provided by
1. You should get a dial tone from SIP as soon as you pick up the phone or
press the call button.
2. show us output of dahdi status , dahdi show channels and sip show
peers from your CLI.
This will give important clues.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
On Fri, May 29, 2009 at 09:24:02AM -0500, Danny Nicholas wrote:
I beg to differ with you Jared. Since I don't have your email, I'll post
this here. Create this call file.
Channel: DAHDI/g1/5551212
CallerID: SIP/104
MaxRetries: 1
WaitTime: 60
retryTime: 5
Application: playback
Data:
On Fri, May 29, 2009 at 12:16:55PM -0400, Mike van der Stoop wrote:
I've had the same issue with 1.6.0.9, if you can dial in, then dial out
then you have the same problem. I turned off hardware DTMF and it works
so far.
The issue may be https://issues.asterisk.org/view.php?id=14761.
Are
Anything above 1.6.0.4 I believe. I required another patch (
http://bugs.digium.com/file_download.php?file_id=21859type=bug ) to get
it to work with 1.6.0.9 but that patch had no affect on 1.6.1.0.
It is getting confusing for me.
Thanks,
Tzafrir Cohen wrote:
On Fri, May 29, 2009 at
On Friday 29 May 2009 11:20:31 am David Backeberg wrote:
On Fri, May 29, 2009 at 4:22 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
i cannot originate call from AMI interface here are my Originate action
Packet
Channel: SIP/111
where 111 Is my SIP phone number which registered
Date: Fri, 29 May 2009 14:27:16 +0800
From: robert aofeish...@163.com
Date: Fri, 29 May 2009 14:28:43 +0800
From: robert aofeish...@163.com
Impatient?
On Fri, 29 May 2009, robert wrote:
i want to kick participant in a meeting by pressing the digit on sip
phone.when i entry the meeting ,no
On Thu, May 28, 2009 at 02:00:15PM -0500, resea...@businesstz.com wrote:
Hello
May i please know if asterisk is now supporting sip call encryption. It
has been a requirement from one of my client to ensure that all
conversation is well secured from any potential sniffers or inside
hackers
resea...@businesstz.com wrote:
On Thu, May 28, 2009 at 02:00:15PM -0500, resea...@businesstz.com wrote:
Hello
May i please know if asterisk is now supporting sip call encryption. It
has been a requirement from one of my client to ensure that all
conversation is well secured from any
Hi all,
I ve setup a queue with 2+ agents for managing our inbound calls from
customer. Using Asterisk 1.2.18 in a CentOS box. Agents login using
AgentCallbackLogin application and I use a BASH AGI to accomplish this
as there are some validations done with MySQL DB. Im aware that transfer
could
HI, all!
During compiling chan_h323, i got the error:
ast_h323.cxx: In member function ‘virtual PBoolean
MyH323Connection::OnReceivedSignalSetup(const H323SignalPDU)’:
ast_h323.cxx:1246: error：‘mccSlaveDeterminationProcedure’ undeclared
in the scope
make[2]: *** [ast_h323.o] error 1
I am using:
Please, does anybody have a good document describes well
the optimum method to achieve Asterisk Redundancy/Clustering on 2 servers.
Documentation?!... well... there's not much.
It depends on what you're trying to achieve with your cluster. If you
want a simple active/passive failover
Hi,
I am a newbie to Asterisk; need help understanding three-way conferencing
call-transfer features implemented over standard extensions i.e. on a
TDM800P card (4 FXO + 4FXS)
In Asterisk I have observed that if an extension is already participating in
an active call (e.g. Ext A Ext B
2009/5/29 Danny Nicholas da...@debsinc.com
I’m pretty sure that attended transfer is a “features” function, not a
dialplan one.
Yes, you're right but do you think there's such a big difference between
both that it shouldn't be easy or even possible to add support of attended
transfer in
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