and totally impractical in the real world.
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind
Sent: Tuesday, February 28, 2012 2:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject
-users] Asterisk RTCP
On 02/17/2012 12:09 AM, Sammy Govind wrote:
Hello,
Thanks for taking out tome for my query. Yes I do have an actual
problem. I've a monitoring tool to record the VoIP QoS (Asterisk servers
port mirrored to it). My end points(soft-phones) are sending RTCP
/2012 01:16 AM, Sammy Govind wrote:
Hello list,
I need to know about Asterisk's friendly nature with RTCP. I've phones
which support RTCP and they connect to the outer world via multiple
carriers. In one of my recent packet traces I've observed that the
caller initiated a call with rtcp string
Thanks for good advice, will definitely keep these in mind while doing
coding - starting from now :)
On Mon, Feb 13, 2012 at 12:30 PM, Steve Edwards
asterisk@sedwards.comwrote:
On Mon, 13 Feb 2012, Sammy Govind wrote:
Hi again,Just to update I fixed the issue. I read through your reply
,
Sammy
On Sun, Feb 12, 2012 at 10:40 AM, Sammy Govind govoi...@gmail.com wrote:
Hey Ron,
Thanks for taking out time for this weird issue. No this is the only code
thats running and I simply copy pasted it here. I'll go through the artivle
you mentioned and other advices you gave may hopefully
Hello all,
I'm struck with a very strange problem today. I've an AGI with some code
subroutine snippet as follows:
sub enable_sbc($) {
my $carrier = shift;
my $tmp = substr($carrier,1);
my $jkh = $tmp;
$server_port = $ast_agi-get_variable(SIPPEER($jkh,port));
I'd definitely go with AMI !
On Sat, Feb 11, 2012 at 6:39 PM, asterisk jobs asteriskcod...@gmail.comwrote:
Thanks for the input but using spool files or AMI or AGI is way different
from each other and that is what I want to get an input on. I do have hands
on with all methods like I noted but
for this and this same code was giving me 100% results. So that means that
the production AGI/perl code has something in it thats causing the issue !?
Regards,
Sammy
On Sat, Feb 11, 2012 at 11:20 PM, Ron Bergin r...@i.frys.com wrote:
Sammy Govind wrote:
Hello all,
I'm struck with a very strange problem today
such a
campaign using AMI? Maybe you can share of the code.
Most appreciated,
On Sat, Feb 11, 2012 at 10:15 AM, Sammy Govind govoi...@gmail.com wrote:
I'd definitely go with AMI !
On Sat, Feb 11, 2012 at 6:39 PM, asterisk jobs
asteriskcod...@gmail.comwrote:
Thanks for the input
Wow,
I bet even asterisk developers wouldn't believe so. What have they done !.
No, actually can you tell if server was processing media along with the
calls as well !?
I once tested without media and really I had some 1000+ CCs on asterisk
server on a regular dev machine with choppy audio on an
,
Zohair Raza
On Tue, Feb 7, 2012 at 11:38 AM, Sammy Govind govoi...@gmail.com wrote:
Hey Danny,
I've this thing exactly running and working as Zohair mentioned! i.e I do
not answer() the call rather put a progress() and soon after that playing
back the sound file from playback
: (Playback) Options:
(congestion,noanswer)
-- SIP/1000-0019 Playing 'congestion.slin' (language 'en')
-- SIP/1000-0019AGI Script agi.php completed, returning 0
Regards,
Zohair Raza
On Tue, Feb 7, 2012 at 11:38 AM, Sammy Govind govoi...@gmail.com wrote:
Hey Danny,
I've
Hello,
I've been managing multiple call centres, almost all of them having their
calls recorded one way or other. Even in PBX environments with MixMonitor
and call recordings I haven't came across the situation where I discovered
that I can't chanspy a call because its recorded !
Which version of
Oh Come on you are Using Asterisk 1.6.2.22. already. Atleast give it a
shot and if this still persists then look for other methods or fixes.
On Tue, Feb 7, 2012 at 5:44 PM, Jonas Kellens jonas.kell...@telenet.bewrote:
**
On 02/07/2012 01:07 PM, Sammy Govind wrote:
Hello,
I've been
Hey Danny,
I've this thing exactly running and working as Zohair mentioned! i.e I do
not answer() the call rather put a progress() and soon after that playing
back the sound file from playback with noanswer and then I get the file
streaming as 183-Session progress file.
I do understand that
Your Server Voipon isn't responding- See if internet is working fine, or
your Voipon sip trunk/peer is registered OK?
On Fri, Feb 3, 2012 at 5:53 PM, virendra bhati virbh...@gmail.com wrote:
Call is not routing from server to destination.
app8*CLI console dial 00918885268942
[Feb 3
You can use a combination of ChanSpy() and a local extension playing the
required file to caller/callee.
On Thu, Jan 26, 2012 at 2:11 PM, Eyal e...@mcr-m.com wrote:
Thanks
** **
But this is not what I am looking for, in this way I can start the sound
file from some point in the file
:D pretty much true !
On Tue, Jan 24, 2012 at 12:23 PM, Alex Balashov
abalas...@evaristesys.comwrote:
Is --[ UxBoD ]-- a first-person shooter gang--er, clan--name? Like, one
of those who rocket-jumps onto the platform and camps with the grenade
launcher, trying to stop the reds from
Yes I've personally experienced issue with nested macros and eventually
asterisk failing to process call any further. So I moved onto using GoSUBs
and everything worked perfectly. Since then I'm using GoSUBs happily.
On Wed, Jan 18, 2012 at 4:54 PM, Jonas Kellens jonas.kell...@telenet.bewrote:
Hi, why don't you try write two macros only and recursively call both of
them incrementing a counter each time you call the inner macro. Also
print(NOOP) system stats along with the counter. You'll soon see what
happens.
The para Matthew quoted is cent percent true. But if you don't need to
call
would have to copy the MoH to our
Asterisk (and change it on our side too, when it changes at the
SIP-server).
Kind regards,
John
2012/1/16 Sammy Govind govoi...@gmail.com
Hi,
yes, please see MusicOnHold() Application. You can call this app in your
dialplan. This however will use
Paste some SIP traces of the call while Unmonitored.
On Mon, Jan 16, 2012 at 4:58 PM, Arlen Nascimento
arlen.nascime...@gmail.com wrote:
It is a satellite connection, so ping is about 500ms. I know it is not ok
to keep a normal conversation, that is not the point.
On Sun, Jan 15, 2012 at
I'm only expecting NAT issues if not the latency issues. SIP traces of any
such calls will make more sense.
On Mon, Jan 16, 2012 at 6:05 PM, Arlen Nascimento
arlen.nascime...@gmail.com wrote:
the client is aware of the adverse environment and this is the only
solution for him
On Mon, Jan
Hi,
yes, please see MusicOnHold() Application. You can call this app in your
dialplan. This however will use the default music class and the
corresponding music files placed in the asterisk server. If you don't want
to stream music from Asterisk server side, try creating a new MusiconHold
Class
which version of Asterisk are you using !. AFAIK this issue has been in
asterisk for queue calls and I'm not sure if this has ever been resolved
fully and stabilized. Not binding to Local channel only, I've seen this on
SIP and IAX channels as well !
On Thu, Jan 12, 2012 at 12:56 AM, Vik Killa
[
asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind [
govoi...@gmail.com]
Sent: Tuesday, January 03, 2012 3:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call type in dial plan
Hi,
For such call you just need to select
Hi,
one reason for having that robotic voice could be improper codecs others
include low CPU processing power, memory not free etc. I once had the same
kind of issue with VAD(voice activity detection) turned ON from my service
providers equipment so my asterisk was performing poorly with VAD.
Hi,
The server or client application that is sending you sip packets is missing
the sip: string in from header. You should have it sorted out because if
that header goes to some external equipment the call may fail because of
this.
Regards,
Sammy
On Thu, Jan 5, 2012 at 12:44 AM, motty.cruz
Are you talking about having an SSH tunnel and route your SIP traffic
through it !!?
On Thu, Jan 5, 2012 at 4:20 AM, Kevin P. Fleming kpflem...@digium.comwrote:
On 01/03/2012 10:03 AM, Patrick Lists wrote:
On 03-01-12 16:24, Danny Nicholas wrote:
Hello List,
I work in an environment where
Hi,
For such call you just need to select the outbound codec before the dial()
app.
choose the audio-only codecs and thus no video codec strings will be
exchanged in that call.
--
Regards,
Sammy
On Tue, Jan 3, 2012 at 1:54 PM, Faraj Khasib fkha...@iconnecths.com wrote:
this is what my SIP
Easy, use Read() to capture the incoming DTMF from Server-B
Server-A Server-B
Initiate-Call - AnswerCall()
SendDTMF(5)-- Read()
Read()-SendDTMF(4)
SendDTMF(3)-- Read()
don't have right or permission to change programming
in server B.
otherwise your suggestion is best for channel base communication.
On Thu, Dec 29, 2011 at 2:33 PM, Sammy Govind govoi...@gmail.com wrote:
Easy, use Read() to capture the incoming DTMF from Server-B
Server
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
Hi,
as the Logs say clearly you need to create an extension in default context
named service
[default]
.
exten = service,1,NOOP(Incoming call from SIPp)
.
Regards,
Sammy
On Tue, Dec 27, 2011 at 5:48 PM, virendra bhati virbh...@gmail.com wrote:
Hi list,
I have installed SIPp into my
Hi,
Please see the Dial application documents from CLI, i.e core show
application dial. There is an option which will let you continue in the
DIal-plan after the Dial command on hangup.
Regards,
Sammy.
On Fri, Dec 23, 2011 at 5:54 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote:
Hello,
I'm
Hi,
STAT function can give you size of a file (
http://www.voip-info.org/wiki/view/Asterisk+func+stat) - Codecs do effect
the call file size, you can see the size difference in case of a gsm and a
wav recorded call.
Regards,
Sammy
On Wed, Dec 21, 2011 at 6:41 PM, silent sayz
Hi,
Not sure why you didnt get it, when I did thta command for originator
channel it showed me the CDR variables list which included
CDR Variables:
level 1: dnid=
level 1: clid=XXX
level 1: src=
level 1: dst=
level 1: dcontext=SIP-incoming
level 1: channel=
level 1:
oops, you got it.
On Wed, Dec 14, 2011 at 2:43 PM, Tony Mountifield t...@softins.co.ukwrote:
In article CAJUJwthT=
mpyxq+omt5hrextl1iqvd0kbs+jhtqlvsqscay...@mail.gmail.com,
Sammy Govind govoi...@gmail.com wrote:
Hi,
Not sure why you didnt get it, when I did thta command for originator
Hi,
I think you need to use the command sip show channel channel-id
Regards,
Sammy
On Wed, Dec 14, 2011 at 12:28 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote:
Hello,
asterisk version 1.6.2.7
I want to get the start time of all active calls from console, could you
please let me know the
Hi,
That depends on what else your asterisk is doing i.e if an AMI-based code
is running then AMI port needs to be open as well. It also depends what
other appliactions are running on asterisk-box which require port opening
i.e apache or mysql etc.
Regards,
Sammy
On Mon, Dec 12, 2011 at 3:21 PM,
Hi,
I'm only going to rephrase what James said, shorten the registration
expiration timer and retry timers. That way phones will retry registrations
lets say after 1 min so after 1 min all phones will failover to the
secondary SRV record.
Regards,
Sammy
On Mon, Dec 12, 2011 at 10:35 AM, Mike
Install DAHDI then !!?
On Thu, Dec 8, 2011 at 12:46 PM, Durgesh Mishra
durgesh.mis...@rancoretech.com wrote:
In make menuselect =application=XXX app_meetme . I am doing confrence
call using sip softphone.
I checked It Depends on: dahdi(E) .
How I can do app_meetme enable?
Thanks
Hello,
AFAIK Hints are used for looking out for a device state before actually
doing anything. Hints can be used from Dial-plan, AMI, or AGI. An example
can be to look for state of a SIP user.
Read these links for better understanding.
http://www.smartvox.co.uk/astfaq_subscribe_hints.htm
Hi again,
Asterisk could be aware of the registrations if the sipusers table is
shared with asterisk sip realtime, but then again the issue would remain
the same that asterisk want to authenticate the sip peer from
scratch..maybe try some Realtime configurations in sip.conf to avoid
Hi,
I dont think that 2 Queue commands would help, also wrapup time is for an
putting delay in an agent who just answered the call and hungup. I think
for this purpose you may need to open up the source code for queue and put
some delay in the relevant code.
Regards,
Sammy.
On Mon, Dec 5, 2011
Hi,
1- Are you sure Playback is capable of understanding/playing video files.
2- Make sure you've enabled videosupport in sip [general] and also allowed
h264,h263 in the sip users section trying to execute this playback app.
Regards,
Sammy.
On Sat, Dec 3, 2011 at 4:13 AM, asterisk jobs
Hi,
How are you calling this AGI in your dialplan !!?
Regards,
Sammy.
On Fri, Dec 2, 2011 at 3:18 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote:
Hello,
I tried to search the answer of my problem but unable to get resolution so
sending to you guys. I'm using asterisk 1.6.2.7 and writing
Can you also paste the Asterisk Console logs around the part where AGI is
dialing and after the dialing part ! make sure AGi debug is enabled as well.
On Fri, Dec 2, 2011 at 4:24 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote:
Hello,
in /etc/extension.conf
[privoip]
exten =
Hey,
Did you try google.com for this!
I've done this several times now. Video for one-to-one call works if H264
is supported at both end points. All you need to do is enable video in
sip.conf and set allow=h264 in the sip peers with video capability.
You may need to see if your asterisk has h264
Yes, Skype was a good thing. R.I.P
On Wed, Nov 16, 2011 at 5:01 PM, Abdul Basit basit.e...@gmail.com wrote:
Any has Skype For Asterisk (SFA) license.
http://www.digium.com/en/products/software/skypeforasterisk.php
PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for
I'd say try a2billing- thats abit of an overkill for just this
functionality but you'll get lot or options to play with there.
On Wed, Nov 16, 2011 at 7:02 PM, ad...@3a.hu wrote:
Hello Hans,
On 11-16-2011 14:46, Hans Goossen wrote:
I guess some billing solution can do the trick, but I
can I make call without registration to an registered SIP account? --
Yes, you can but first you need to set allowguest=yes in sip.conf (makes ur
server insecure)
I guess you can put in same user/sip account in all iphones and like (in
x-lite) don't let the phones register to server rather set
Hi,
Can I use 2 SIP Trunks from each remote offices to the central site
and permit 2 simultaneous calls across the SIP trunk that passes over
the smaller line, and permit 10 simultaneous calls across the larger
link?
Yes.
I also wish to have priorities, so that more important calls are
** **
*From:* Sammy Govind [mailto:govoi...@gmail.com]
*Sent:* Monday, November 14, 2011 10:36 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] How do extensions stay registered
** **
Continuing eherr here, behind the OPTIONS messages
Continuing eherr here, behind the OPTIONS messages(infact all SIP comm) you
definitely to look into SIP timers which tell how many time to resend a
packet if no response is received and for how long to wait before thinking
that the SIP packet got lost(network disconnected or end-point lost)
so,
Hey,
Though your requirements are unclear and below may not exactly fit your
specs unless you give some more usage details.
if your gateway requires no authentication, yes you can do this by writing
a dialplan extension like below
exten = calling-togw,1,NOOP(I'll be getting some variables from
Hi,
The end-point which isn't registered in asterisk will hit the default
context in asterisk. This is the one which you've defined in sip.conf
general section i.e
[general]
...
context=my-context
Also, if your calls are successful from any unregistered endpoint then I
think you've enable
Hello,
Reading about the application DumpChan() shows this:
[Synopsis]
Dump Info About The Calling Channel.
[Description]
Displays information on channel and listing of all channel variables. If
level is specified, output is only displayed when the verbose level is
currently set to that number
Hey Sunny,
I think your initial post on what you're looking for don't really tells
much. I think initially you were looking at a different architecture than
now i.e Kamailio+RTPproxy, this changes a lot of things.
If you dont want transcoding and thinking on using Kam+Rtpproxy then I
think
Hey,
How are you starting the recording? MixMonitor? or Monitor? or some option
in an application?
If you are using MixMonitor or anything alike then you should
StopMixMonitor when the call hits the h extension.
Paste your dialplan relevant to the recording scenario to suggest you
something
, Nov 4, 2011 at 10:05 AM, Sammy Govind govoi...@gmail.com wrote:
Hey,
How are you starting the recording? MixMonitor? or Monitor? or some
option in an application?
If you are using MixMonitor or anything alike then you should
StopMixMonitor when the call hits the h extension.
Paste your
There is only one way to find out connecta.agi in /var/lib/asterisk/agi-bin
directory by developing it yourself.
On Tue, Nov 1, 2011 at 6:57 PM, Thanasis thana...@asyr.hopto.org wrote:
on 11/01/2011 03:25 PM Danny Nicholas wrote the following:
One way to do this (there are probably more and
Type in asterisk CLIcore show application meetme
or google asterisk cmd meetme simple?
On Tue, Nov 1, 2011 at 10:33 PM, Thanasis thana...@asyr.hopto.org wrote:
on 11/01/2011 05:41 PM Yaroslav Panych wrote the following:
You need simple dialplan of four steps:
same
hmmm so IAX channel is playing with you guys.
1- Cant you guys use SIP, does this happen with SIP trunk as well !?
2- Which version of asterisk are there on both servers.
3- See the output of the command core show file versions in your both
asterisk servers. Mainly lookout for IAX channel
.*SIP/911|pbx.*SIP/911'
Interesting technique from Astresk Cookbook, Debugging dialplan with
Verbose()
http://ofps.oreilly.com/titles/9781449303822/DialplanFundamentals.html
2011/10/27 Sammy Govind govoi...@gmail.com
It was a challenge to read through all the interesting experience
Try turning on the Sip debug for the PSTN call as well as RTP debug. Paste
the output here.
The Dial server is connected to multiple 4-port Redfone devices for
handling PSTN incoming and outgoing calls. Outgoing calls always
originate from and incoming calls always terminate at the SIP
It was a challenge to read through all the interesting experience you've
shared over here. I don't know what others may be using for parsing the logs
beautifully and make them usable. What I would recommend you at the very
beginning ,since you mentioned using egrep, is figure out the Channel
I wrote my own shell scripts to collect core show calls value from
asterisk and then push the filtered value to an opensource monitoring tool.
That worked perfectly well.
#!/usr/bin/perl -w
use strict;
open(LINE, 'asterisk -rx core show channels|');
my ($chans, $calls, $line)=(0,0,undef);
while
OP may be able to use System through Dial plan but I'm thinking that since
tcpdump don't just give output within seconds or neither do it get
daemonized? so this system() call will hold the call to that priority. This
may even result in call failure. I think this system call should trigger a
shell
Set CDR(destination) or whichever field you need to get recorded in CDRs to
get your desired stats.
On Mon, Oct 24, 2011 at 2:19 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
As I am using the ${CALLERID(num)} to be part of the filename that I am
recording it, I am facing the
Yes, Macro will return to calling context BUT use GoSub instead and your
life will be easy. Forget using Macro whenever you need to get user input in
there.
On Mon, Oct 24, 2011 at 2:35 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
Is it possible to be part of the voicemail to play a
Try using variables between macros and contexts without doing anything. It
works fine for me in asterisk 1.6.13+. If not then use _ before variable
name.
On Mon, Oct 24, 2011 at 2:46 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
Is it possible to store a variable at context and using
Hi,
I've done some similar thing in one of my testing, using MixMonitor and
monitor at the same time. Everything worked perfectly well no issues even on
Vmware. Can you check if the CPU utilization is normal. Also which version
of asterisk you are using?
--
Regards,
Sammy
On Thu, Oct 20, 2011
2011 13:30:20 +0500
From: Sammy Govind govoi...@gmail.com
Subject: Re: [asterisk-users] Monitor does not work well (little cuts
in the audio file)
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID:
cajujwtg
Hi,
Call getting silenced in the middle definitely point to RTP but I think
the redialling part should be considered as well. I think that Phones are
loosing registrations or like Zeeshan mentioned could be getting blocked by
firewall - Asterisk server's firewall as well as any other firewall in
1- Are you sure your Asterisk Box is configured with an MTA / email utility
to send emails ?
2- Like Ishfaq suggested you should be getting into the voicemail
application after 10 seconds of Dial timeout. Are you even recording and
saving a voicemail?
3- To recieve an SMS to notify you of
Hey,
I don't think you are doing it right. The memebers/channels you need to spy
should be added in SPYGROUP and not the channel which is spying. i.e your
code maybe something like this.
exten = 4368,1,Answer()
exten = 4368,n,NoOp(${CHANNEL})
exten =
Hi,
I'd been thinking about such a huge conferencing system for about last few
months. Like Steve suggested, my concept is almost similar but instead of
making a central hub conference junction between multiple Conferences I was
thinking of making a peer2peer runtime connection between
Hey,
Can you enable sip trace for that particular sip extension. This sounds
weird that while other INVITES from the phone are reaching but the external
extensions are filtered. If there are no invites for external calls only
then more chances are that the phone is using some dial
Please paste the configurations in the #included files as well.
On Fri, Oct 7, 2011 at 7:07 PM, michael k mich...@inapp.com wrote:
Hi,
This is my /etc/asterisk/chan_dahdi.conf file.
[root@astrisks asterisk]# cat /etc/asterisk/chan_dahdi.conf
; Copied from DAHDI Module of FreePBX
If DAHDI is not really configured or chan_dahdi isn't loaded the the error
mesage would be can not create channel of type DAHDI but here its not the
case. Dadhi module is indeed loaded but the DAHDI device is not working
properly.
On Thu, Oct 6, 2011 at 8:49 PM, Gohar Ahmed gohar.ah...@vopium.com
How are you calling the beep.alaw from the dialplan?
paste the relevant dialplan here and corresponding CLI logs.
On Wed, Oct 5, 2011 at 11:58 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
I placed a beep.alaw file in de directory, but I get the same result.
Also I try to set
] *Namens *Sammy Govind
*Verzonden:* 05-10-2011 09:04
*Aan:* Asterisk Users Mailing List - Non-Commercial Discussion
*Onderwerp:* Re: [asterisk-users] Beep file with Record
** **
How are you calling the beep.alaw from the dialplan?
paste the relevant dialplan here and corresponding CLI
/ of but
without any success.
** **
*Van:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *Namens *Sammy Govind
*Verzonden:* 05-10-2011 09:26
*Aan:* Asterisk Users Mailing List - Non-Commercial Discussion
*Onderwerp:* Re: [asterisk-users] Beep file
)
On Wed, Oct 5, 2011 at 12:31 PM, Sammy Govind govoi...@gmail.com wrote:
hmmm...what i'm saying is this
*exten = s,n,Set(CHANNEL(language)=en))*
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
exten = s,n,Record(${A_serviceline_file}.wav,0,60
The alaw extension is bugging me.. can you locate the default beep.gsm
/beep.wav file in asterisk sounds directory !?
Also check the output of
*core show file formats*
*core show translation*
Also find out the codec of the established call.!
On Wed, Oct 5, 2011 at 12:50 PM, Jeroen Eeuwes
Give that moh1 directory permissions, I once had similar issue that same
files being placed in default moh directory were played but making a new
call and directory couldn't play anything. So I fixed that by granting
directory permissions.
On Wed, Oct 5, 2011 at 2:25 PM, salaheddine elharit
Can you please explain what you are trying to do? What I've perceived from
this thread is that you want to put call on hold (passively as in no
resources usage) and then on the base of some User's input from UI proceed
with the call accordingly !!
On Wed, Oct 5, 2011 at 3:33 PM, Yaroslav Panych
expert here guide you in a better direction.
On Wed, Oct 5, 2011 at 4:44 PM, Yaroslav Panych panyc...@gmail.com wrote:
Yes, something like that, but
hold-state should not answer channel. answer command will be given
explicitly. or call can be transfered to Dial command, etc.
2011/10/5 Sammy
89 matches
Mail list logo
