Hi,
We noticed a very sharp drop in voice quality when using digium g729a
codec. The problem seems to happen if the A channel (caller's channel)
is a landline/mobile number contacted using the same outgoing provider
(as a local channel). It sounds like listening to a mono speaker on
low volume.
Ken,
Thank you for posting the details. The method worked perfectly.
I was about to give up on connecting via SSH to manually provisioned
Cisco phones.
Thank you,
Vladimir
On 1/15/2012 8:52 PM, Ken Alker wrote:
Flavio,
Thank you for pointing this out! I was using the reference
Hello sir,
There is an echo problem in sip voip call. I think it is because of delay in
audio.
Let me try to explain you my system setup.
I have test asterisk on two different system.
System : 1
OS :Ubuntu(10.04)Lucid
System Type :x64-based pc
Processor
Hi!
Many thanks for this hint. I will try this! :-)
A quick question: when doing this with MusicOnHold(): will the SIP
server be aware that the call is placed onHold (i.e. will Asterisk
send the mentioned re-INVITE)?
The point is - if possible - we want the caller to hear the OnHold
Music from
Hi,
I've recently upgraded a system from 1.8 to asterisk 10 and also
updated spandsp while doing so.
I wondered what is the safest and easiest way to check from command
line which libraries a running Asterisk system is currently using
(just like dahdi show version, for instance).
Though I'm
ldd
2012/1/16 Olivier oza_4...@yahoo.fr
Hi,
I've recently upgraded a system from 1.8 to asterisk 10 and also
updated spandsp while doing so.
I wondered what is the safest and easiest way to check from command
line which libraries a running Asterisk system is currently using
(just like
Hi Olivier,
I suppose you give strace a try.
It's a powerful debugging utility, you should be able to find everything
you are looking for.
best regards,
Ruben
Am 16.01.2012 11:14, schrieb Olivier:
Hi,
I've recently upgraded a system from 1.8 to asterisk 10 and also
updated spandsp while
Hi to all,
I've managed to get the XMPP PubSub method to work on my set-up! Just
carefully follow these instructions on the wiki:
https://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+XMPP+PubSub
Maybe this IRC log would also help you troubleshoot:
Hey,
I have never worried about looking at the SIP re-invites or anything when
we engage MoH() application in asterisk. You can do a quick test on your
test machine for this.
Regards,
Sammy
On Mon, Jan 16, 2012 at 2:57 PM, Johannes Zweng john999...@zweng.at wrote:
Hi!
Many thanks for this
2012/1/16, Anton Kvashenkin anton.juga...@gmail.com:
ldd
Thanks for replying.
I got this:
# ldd /usr/sbin/asterisk
linux-gate.so.1 = (0xb7886000)
libssl.so.0.9.8 = /usr/lib/i686/cmov/libssl.so.0.9.8 (0xb7834000)
libcrypto.so.0.9.8 = /usr/lib/i686/cmov/libcrypto.so.0.9.8
On Monday 16 January 2012, Olivier wrote:
Hi,
I've recently upgraded a system from 1.8 to asterisk 10 and also
updated spandsp while doing so.
I wondered what is the safest and easiest way to check from command
line which libraries a running Asterisk system is currently using
(just like
Hey all!
I'm not sure if this went out the first time I sent it so I apologize now
if it's a duplicate.
I've been banging my head against the wall for a while (almost 18 hours
today alone) with this one... I migrated our incomming T1's from the Option
11 to our Asterisk box this morning. We have
On 01/16/2012 04:48 AM, Louis Carreiro wrote:
I've been banging my head against the wall for a while (almost 18
hours today alone) with this one... I migrated our incomming T1's from
the Option 11 to our Asterisk box this morning. We have 1 local T1 and
2 long distance T1's. The local T1
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 10:34 PM, Flavio Miranda flaviormira...@hotmail.com
wrote:
Hi Arlen,
A reasonable time to Voip calls is about 250 ms. What about the Ping
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
Dale,
That's funny! That is almost exactly what I'm trying to do. Thanks for
the quick response! I'm on the way into the office now and I'll give
the configuration a shot. I hope the config really helps. Maybe with
our two migrations happening at the same time we maybe able to help
each other
basically CLI shows
SIP/X called SIP/Y
I answer the call on Y but X keeps ringing and then both hangup.
On Mon, Jan 16, 2012 at 8:01 AM, Sammy Govind govoi...@gmail.com wrote:
Paste some SIP traces of the call while Unmonitored.
On Mon, Jan 16, 2012 at 4:58 PM, Arlen Nascimento
On Mon, Jan 16, 2012 at 11:14:48AM +0100, Olivier wrote:
Hi,
I've recently upgraded a system from 1.8 to asterisk 10 and also
updated spandsp while doing so.
I wondered what is the safest and easiest way to check from command
line which libraries a running Asterisk system is currently using
Hi,
Where to find meaning of /n in Local/6613@from-queue/n ?
Regards
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Ok, I will try this and let you know!
Kind regards,
John
2012/1/16 Sammy Govind govoi...@gmail.com:
Hey,
I have never worried about looking at the SIP re-invites or anything when we
engage MoH() application in asterisk. You can do a quick test on your test
machine for this.
Regards,
http://www.voip-info.org/wiki/view/Asterisk+local+channels
Regards
- Original Message -
From: Olivier oza_4...@yahoo.fr
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, January 16, 2012 7:41 AM
Subject: [asterisk-users] Where
2012/1/16, bakko asannu...@gmail.com:
http://www.voip-info.org/wiki/view/Asterisk+local+channels
I don't know why but I was thinking of some sort Dial app magic and
didn't look after Local channels options.
Thanks for correcting me.
Regards
- Original Message -
From: Olivier
Unless you are doing test with SIP under adverse environmet, that is not the
point, but, if you intend to have Communication, you should worry about this
detail.
Basic infra-estructure is the first thing to think in any new project.
Good luck!
Att,
Flavio Roberto Miranda
the client is aware of the adverse environment and this is the only
solution for him
On Mon, Jan 16, 2012 at 9:00 AM, Flavio Miranda
flaviormira...@hotmail.comwrote:
Unless you are doing test with SIP under adverse environmet, that is not
the point, but, if you intend to have Communication,
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
Anybody? I've read this might be a deadlock
On Thu, Jan 12, 2012 at 8:09 AM, Vik Killa vipki...@gmail.com wrote:
Asterisk 1.6.1.22
On Thu, Jan 12, 2012 at 2:08 AM, Sammy Govind govoi...@gmail.com wrote:
which version of Asterisk are you using !. AFAIK this issue has been in
asterisk for
In addition: I tried adding Playback(hello) to the 123 extension, before
the SayDigits. Then everything is being played perfectly.
Also when I park a call to 700, I cannot hear the playback of the parking
lot. I do see this in the logs though, so I can pickup the call then, but
it should be
You aren't opening the line in the 123 call. In the 200 call, the
Answer() opens the output audio channel. In the 123 call you are plunging
into the SayDigits() function without opening the channel. Some functions
will generate their own Answer() if not present, others will not.
From:
Ok, got it. Indeed, starting with Answer() helped.
But I still don't understand why the parking feature isn't working then. I
used the sample config. Transfer the call to 700, playback of the lot is
being executed, but I hear nothing. Probably the same problem, but how do I
change this?
On Mon,
Hi list,
how we can configure between call add the IVR.
My scenarios is
A get the incomming call from C.In between them I need to one side IVR
play for C, C enter the some DTMF inputs and A should be on hold.
after finish C input will complete again they want talk each other .This
is the
Post your dialplan snippet you use to park the call.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roland
Sent: Monday, January 16, 2012 9:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
This symptom usually means you are doing an attended transfer instead of a
blind transfer.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roland
Sent: Monday, January 16, 2012 10:57 AM
To: Asterisk Users
A should transfer C to a local channel that plays the IVR then returns the
call to A.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of mahesh katta
Sent: Monday, January 16, 2012 9:56 AM
To: Asterisk Users Mailing List - Non-Commercial
Hi,
A calls B and B has it's phone forwarded to C. So the call rings at C.
Is there any way to inform A about that forwarding? Best way would be
to update the called name so A has B forwarded to C in his display.
Any chance to get this?
I tried Set(REDIRECTING(to-name)=...). This sends a SIP/2.0
I am just starting with Asterisk .. I think you are right, I am doing an
attended transfer, although I don't exactly understand what that means. I
still need to know in what lot I can pickup my call again right?
Ok, my config .. (i will leave out the commented stuff, because there's lot
of
I was tried it but its not going.. with same
Best Regards,
Mahesh Katta
On Mon, Jan 16, 2012 at 9:32 PM, Danny Nicholas da...@debsinc.com wrote:
A should transfer C to a local channel that plays the IVR then returns the
call to A.
** **
*From:* asterisk-users-boun...@lists.digium.com
I would do it something like this
[ivrandreturn]
Exten = s,1,playback(message)
Exten = s,n,waitexten(5)
Exten = 1,1,noop(stuff for press 1)
Exten = 1,n,dial(SIP/A)
Exten = 2,1,noop(stuff for press 2)
Exten = 2,n,dial(SIP/A)
In real life SIP/A would be something like SIP/${ARG1} where
Yes, a 'call' refers to two channels bridged.
Jim, please help me to undertand the numbers. I have two g729 licenses, my
SIP provider uses only g729 and my softphones support g729 too,
asterisk.conf is set in its default value (sln).
When a call (2 channels) is being made and succesfully
Best Regards,
ahesh Katta
On Mon, Jan 16, 2012 at 9:57 PM, Danny Nicholas da...@debsinc.com wrote:
I would do it something like this
[ivrandreturn]
Exten = s,1,playback(message)
Exten = s,n,waitexten(5)
Exten = 1,1,noop(stuff for press 1)
Exten =
Where to find meaning of /n in Local/6613@from-queue/n ?
See https://wiki.asterisk.org/wiki/display/AST/Local+Channel+Modifiers
Richard
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New to
Hi all,
please help me.
how we can configure between call add the IVR.
My scenarios is
A get the incomming call from C.In between them I need to one side IVR
play for C, C enter the some DTMF inputs and A should be on hold.
after finish C input will complete again they want talk each other .This
Is it possible to make Asterisk jump into action and play a sound file as soon
as a handset is lifted, instead of providing a dialling tone and waiting for
the user to dial an extension?
--
AJS
Answers come *after* questions.
--
On Mon, Jan 16, 2012 at 05:52:10PM +, A J Stiles wrote:
Is it possible to make Asterisk jump into action and play a sound file as
soon
as a handset is lifted, instead of providing a dialling tone and waiting for
the user to dial an extension?
With analog phones (chan_dahdi) -
Hello,
I can do simple, yum install asterisk18-* and it installs Asterisk and
Dahdi-tools/Dahdi-Linux on my OpenVZ container. Everything runs well and
smooth.
However, if I want to compile dahdi-linux on the same openvz then I get the
error, *You do not appear to have the source for the
On Mon, Jan 16, 2012 at 01:41:30PM -0500, asterisk jobs wrote:
1- Based on above error and Google search I have concluded that dahdi-linux
module should be installed on mother node. So, I am puzzled. How does
Digium yum repository achive this without acessing the mother node?
The repo files are
On 16-01-12 19:47, Russ Meyerriecks wrote:
[snip]
2- Do I even need Dahdi, if the server doesn't connect to PSTN at all and
it's all SIP? If yes, what do I need it for?
Dahdi is a set of drivers for telephony hardware. You won't need it for pure
sip Asterisk implementations.
Unless things
I've never done it myself yet but I think I would look after COLP
function (1.8 and above).
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On 01/16/2012 12:55 PM, Olivier wrote:
I've never done it myself yet but I think I would look after COLP
function (1.8 and above).
Asterisk 1.8 and later will do this automatically; if the phone can
display the redirection information, it will get displayed (not all
phones can do so).
--
On 01/16/2012 12:52 PM, Patrick Lists wrote:
On 16-01-12 19:47, Russ Meyerriecks wrote:
[snip]
2- Do I even need Dahdi, if the server doesn't connect to PSTN at all
and
it's all SIP? If yes, what do I need it for?
Dahdi is a set of drivers for telephony hardware. You won't need it
for pure
sip
2012-01-16 19:41, asterisk jobs skrev:
Hello,
I can do simple, yum install asterisk18-* and it installs Asterisk
and Dahdi-tools/Dahdi-Linux on my OpenVZ container. Everything runs
well and smooth.
However, if I want to compile dahdi-linux on the same openvz then I
get the error, /*You do
Thanks for all the input guys.
I am using Asterisk 1.8 for this purpose.
1- So, I do I still need Dahdi? And yes conference will be used.
2- Can you please detail on compiled already code? My mother node for
OpenVz is probably different from what Digium uses to compile the source.
How does this
On 01/16/2012 12:55 PM, Olivier wrote:
I've never done it myself yet but I think I would look after COLP
function (1.8 and above).
Asterisk 1.8 and later will do this automatically; if the phone can
display the redirection information, it will get displayed (not all
phones can do so).
See http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE pay special
attention to the sendrpid note.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gunnar Schaller
Sent: Monday, January 16, 2012
Hello Eric,
See http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE pay
special attention to the sendrpid note.
That does not work. CONNECTEDLINE is for answered calls.
A calls B. B has a forward to C in Asterisk dialplan. A want's to
notice the forwarding _before_ C answers. Cause
Hi,
Freepbx includes a fax_process.pl which convert TIF files into PDF
files before sending by email.
I'm used to use sSMTP with Asterisk.
I'm certain ssmtp is correctly configured in my (Debian Squeeze) setup
as I'm correctly receiving voicemails in email box.
Is it possible to tell
Are both A and B extensions of the same Asterisk system or is A an
inbound caller ?
2012/1/16, Gunnar Schaller li...@nowin.de:
Hello Eric,
See http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE pay
special attention to the sendrpid note.
That does not work. CONNECTEDLINE is for
On Mon, Jan 16, 2012 at 5:48 AM, Louis Carreiro carreir...@gmail.com wrote:
Hey all!
I'm not sure if this went out the first time I sent it so I apologize now if
it's a duplicate.
I've been banging my head against the wall for a while (almost 18 hours
today alone) with this one... I
Any one is help ?
Best Regards,
Mahesh Katta
On Mon, Jan 16, 2012 at 10:41 PM, mahesh katta maheshka...@flexydial.comwrote:
Hi all,
please help me.
how we can configure between call add the IVR.
My scenarios is
A get the incomming call from C.In between them I need to one side IVR
play
Are both A and B extensions of the same Asterisk system or is A an
inbound caller ?
Both are snom phones at the same Asterisk (1.8.8).
Regards,
Gunnar
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