Hi,
I am receiving DTMF without any reason after call establishment.
The log as follows, and I suspect something related to directmedia,
[May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is
making progress passing it to SIP/MAN-000a4b48
[May 17 00:33:35] VERBOSE[4238]
hi,
anyone can help me to debug this ??
--
upendar
On Mon, May 27, 2013 at 4:09 PM, upendra uppi...@gmail.com wrote:
hi,
chan_local and res_crypto are building but the chan_sip is not building .
installed openssl also but still the chan_sip not building.
--
Upendra
On Mon, May 27,
Without posting exact error messages, dont expect help !!
Mitul
On May 28, 2013 1:01 PM, upendra uppi...@gmail.com wrote:
hi,
anyone can help me to debug this ??
--
upendar
On Mon, May 27, 2013 at 4:09 PM, upendra uppi...@gmail.com wrote:
hi,
chan_local and res_crypto are building
hi,
there is no build errors , but the thing is that on Elastix Machine i want
to install asterisk1.8.11.0 , while make the chan_sip module is not
building, and when i see in the memuselect the chan_sip module driver
showing as XXX to enable for building.
--
Upendra.
On Tue, May 28, 2013 at
Why not install the updated rpm version?
Mitul
On May 28, 2013 1:12 PM, upendra uppi...@gmail.com wrote:
hi,
there is no build errors , but the thing is that on Elastix Machine i want
to install asterisk1.8.11.0 , while make the chan_sip module is not
building, and when i see in the
On Mon, May 27, 2013 at 04:09:06PM +0530, upendra wrote:
hi,
chan_local and res_crypto are building but the chan_sip is not building .
installed openssl also but still the chan_sip not building.
./menuselect/contrib/menuselect-dummy -c
./menuselect/contrib/menuselect-dummy -m sip -v
What's
hi all,
After installing packages openssl and openssl-devel packages the chan_sip
is building . :) :)
thanks to all for ur help.
--
Upendra.
On Tue, May 28, 2013 at 1:33 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Mon, May 27, 2013 at 04:09:06PM +0530, upendra wrote:
hi,
please provide more information.
how you are try to build asterisk, what is output of configure. witch
headers configure script not found etc.
On Tue, May 28, 2013 at 9:29 AM, upendra uppi...@gmail.com wrote:
hi,
anyone can help me to debug this ??
--
upendar
On Mon, May 27, 2013 at
i had this in past there was an ATA configured to send 9 at the end of
dialing in my case.
On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Hi,
I am receiving DTMF without any reason after call establishment.
The log as follows, and I suspect something
hello,
201.xxx.xxx.xxx = SIP Softphone which originates the call
xxx.xxx.xxx.xxx = Asterisk server
yyy.yyy.yyy.yyy = ITSP
--- SIP read from UDP:201.xxx.xxx.xxx:5060 ---
INVITE sip:12127773...@xxx.xxx.xxx.xxx SIP/2.0
To: sip:12127773...@xxx.xxx.xxx.xxx
From:
So any resolution for this?
I suspect it could be related to RE INVITE
On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad asghar...@gmail.comwrote:
i had this in past there was an ATA configured to send 9 at the end of
dialing in my case.
On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N
work around was block dtmf.
set wrong type of dtmf in incoming trunk.
On Tue, May 28, 2013 at 11:15 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
So any resolution for this?
I suspect it could be related to RE INVITE
On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad
On Sat, May 25, 2013 at 10:32 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Finally got it working with 3 attempts by the fialplan,
exten = 300,1,Playback(letters/a)
exten = 300,n,Set(gottries=0)
exten = 300,n(getmore),Set(rightPIN=1)
exten = 300,n,Read(inPIN,,1,skip,3,3) ;
On Mon, May 27, 2013 at 12:14:41PM -0500, Mitch Claborn wrote:
Asterisk 11.1
We have a situation where one of our incomings POTS lines will not
answer. There are 2 lines configured by the Telco as a rollover
group (rings the line that is not busy) and they feed into a Digium
AEX410 on the
El 27/05/13 01:56, upendra escribió:
Hi,
i am trying to install asterisk newer version on the Elastix machine, but while installing the chan_sip,c module is not building while make. when i see in make menuselect options it showing XXX -- extended , please let me know how to enable it and
make
Greetings-
I've got a curious project that I could use some input on. I'd like to use
Asterisk to record some audio channels via USB 'soundcard'. When audio passes
through the soundcard, Asterisk should grab that audio channel (CONSOLE?), and
write it to a wav file. I'm perfectly competent
What are you trying to accomplish?
What is the USB 'sound card' attached to?
Your description is too cryptic for someone to propose a solution.
Ron
On 28/05/2013 12:45 PM, Tim Nelson wrote:
Greetings-
I've got a curious project that I could use some input on. I'd like to use
Asterisk to
- Original Message -
What are you trying to accomplish?
What is the USB 'sound card' attached to?
Your description is too cryptic for someone to propose a solution.
The target use is to record mic level audio from various devices (could be an
omnidirectional room mike, phone
I am running 2.6.1. I'll give the 2.6.y a try.
Mitch
On 05/28/2013 10:53 AM, Shaun Ruffell wrote:
On Mon, May 27, 2013 at 12:14:41PM -0500, Mitch Claborn wrote:
Asterisk 11.1
We have a situation where one of our incomings POTS lines will not
answer. There are 2 lines configured by the
I got the following warning during the build. Is it anything to worry
about?
WARNING: could not find
/home/mclaborn/asterisk/dahdi-2.6.y-snapshot-20130528-linux-20a479b/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_64.o.cmd
for
/home/mclaborn/asterisk/dahdi-2.6.y-snapshot-20130528-linux
On 28/05/2013 1:23 PM, Tim Nelson wrote:
- Original Message -
What are you trying to accomplish?
What is the USB 'sound card' attached to?
Your description is too cryptic for someone to propose a solution.
The target use is to record mic level audio from various devices (could be an
Sorry for the blank message. Fingers pressed send while brain was disenaged.
Would Audacity be a better choice?
http://wiki.audacityteam.org/wiki/Multichannel_Recording
Ron
On 28/05/2013 1:23 PM, Tim Nelson wrote:
- Original Message -
What are you trying to accomplish?
What is the
It seems that initial audio for SIP channels does not get transmitted
for a period of varying length, typically about 1 second. This also
applies to bridged SIP calls as well to one-legged calls where only
Playback() gets called.
The Definitive Asterisk Guide uses constructs like silence/1 or
- Original Message -
Sorry for the blank message. Fingers pressed send while brain was
disenaged.
Would Audacity be a better choice?
http://wiki.audacityteam.org/wiki/Multichannel_Recording
It would absolutely be a better solution. However, the recording is to be
automated on a
On Tue, May 28, 2013 at 12:44:47PM -0500, Mitch Claborn wrote:
I got the following warning during the build. Is it anything to
worry about?
WARNING: could not find
/home/mclaborn/asterisk/dahdi-2.6.y-snapshot-20130528-linux-20a479b/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_64.o.cmd
You are still being a bit evasive but should I understand that you want
to run a headless machine with open microphones that records what ever
it hears?
What do you want to do with each sound bite?
How long does the silence have to be before you close the recording and
dispose of it (save,
I'll take a stab, since you said no GUI and also USB based mic.
Raspberry Pi project? I'm interested in this vein as well, especially
after the recent post about voice recognition. I was thinking that
Raspberry Pi's with mics could live around my house and all have
dedicated always-open
So what is the fix for it?
It was working all fine, all of a sudden it stopped working.
Regards
Jitesh Gala
Director
Fantasia Business Park, Nano Wing, S-10,
Sector-30A, Vashi, Navi Mumbai - 400705
Cell No: +91 9769144905
Skype ID: jitesh_gala
www.hubrisbpo.com
-Original Message-
Wednesday AM I hope
Connected by DROID on Verizon Wireless
-Original message-
From: Shaun Ruffell sruff...@digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tue, May 28, 2013 15:53:45 GMT+00:00
Subject: Re: [asterisk-users]
Claborn wrote:
I got the following warning during the build. Is it anything to
worry about?
WARNING: could not find
/home/mclaborn/asterisk/dahdi-2.6.y-snapshot-20130528-linux-20a479b/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_64.o.cmd
for
/home/mclaborn/asterisk/dahdi-2.6.y-snapshot
Kamlesh,
Please provide SIP traces of both call legs for a failed call.
Your last message only included a SIP trace of the call leg from the SIP
softphone to the Asterisk server. There was no SIP trace for the call leg from
the Asterisk server to the ITSP and, as shown below, that is probably
Let me try with dtmfmode as auto...
On 28 May 2013 19:32, Asghar Mohammad asghar...@gmail.com wrote:
work around was block dtmf.
set wrong type of dtmf in incoming trunk.
On Tue, May 28, 2013 at 11:15 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
So any resolution for this?
I
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