--- On Mon, 11/10/08, Igor Goncharovsky [EMAIL PROTECTED] wrote:
From: Igor Goncharovsky [EMAIL PROTECTED]
Subject: Re: [asterisk-users] changing the size of voice packets
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Date:
sean darcy wrote:
[set-callerid-name]
exten = 0,1,NoOp( no CALLERID num set)
exten = 02025462677,1,Set(CALLERID(name) = Fred )
Try adding the following to your subroutine:
SetCallerPres(allowed)
I would also suggest that you put the name/number groups in either the
What is the AMI command to see how many PRI channels are being used / available?
Thanks
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
I returned both cards and got a Sangoma. We'll see...
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Hi All,
I am checking srtp support in asterisk 1.6,
Let me know any patches available or changes needed for srtp support in
asterisk 1.6.
Thanks in advance!
Thanks,
Max Alex
Voip Developer
___
-- Bandwidth and Colocation Provided by
On Thu, Nov 06, 2008 at 11:46:48AM +0100, Louis-David Mitterrand wrote:
Hi,
After upgrading our server from asterisk 1.2.16 to 1.4.21.2 we
experience crashes at random intervals with:
[Nov 6 11:03:28] WARNING[12230] app_dial.c: Unable to forward voice frame
read(0, unfinished ...
On Sunday 09 November 2008 20:14, Eric ManxPower Wieling wrote:
The best (and maybe only way) is to set your client and your service
provider to only do G.723.
Really, thats not the way it should work.
How I can find out the codec of an incomming call?
Is there any way to use ${SIP_CODEC} to
hi folks,
i have a issue with my setup.
i recently install xen in my server, and create several
domu, but when i start a domU or i am working in it,
my fxs in my tdm400p have noise. when i am calling
from iax or sip extension trough the fxo there not noise, it
just with the fxs port and domU
Dear,
is any way to change , the size of voice packets?
I want to increase the quality by decreasing the size of each packets, because
of bandwidth failure.
thanks in advance
Mani
___
-- Bandwidth and Colocation Provided by
Hi,
I want to be able to bridge two sip channels using direct RTP
between my endpoints (Audio IP : not local) but without
using reinvites. So I set up my asterisk sip endpoints as follows:
[test1]
type=friend
host=dynamic
username=test1
dtmfmode=info
context=test_rtp
allow=all
Wilton Helm wrote:
Good points. I got an access point instead of a router specifically
so I could locate it in the best position. IMO Wi-Fi routers are dumb
by definition because where you want a router is probably NOT anywhere
close to the best point for the Wi-Fi part. This unit has a
On Monday 10 November 2008 16:52, Eric ManxPower Wieling wrote:
Thomas Winter wrote:
On Sunday 09 November 2008 20:14, Eric ManxPower Wieling wrote:
The best (and maybe only way) is to set your client and your service
provider to only do G.723.
Really, thats not the way it should work.
You may want the original link I started out with:
http://www.newthink.net/2007/05/18/smarthost-authentication-with-sendmail/
I deviated a bit though - I put the AuthInfo stuff into its own authinfo
file though, which also required me to add this line to sendmail.mc:
FEATURE(`authinfo', `hash -o
You can sometimes find the older Cisco Aironet boxes that run at 900Mhz.
That frequency is AWESOME in rural areas. Mountains will still block
it, but trees and water does not.
Drew Gibson wrote:
Wilton Helm wrote:
Good points. I got an access point instead of a router specifically
so I
Godson Gera wrote:
On Mon, Nov 10, 2008 at 7:41 PM, David Budny [EMAIL PROTECTED]wrote:
What is the AMI command to see how many PRI channels are being used /
available?
Thanks
There is no direct command in AMI which will give you used channels number.
But you can easily keep
On Mon, 10 Nov 2008 11:05:46 -0500, Drew Gibson wrote:
Wilton Helm wrote:
Good points. I got an access point instead of a router specifically
so I could locate it in the best position. IMO Wi-Fi routers are dumb
by definition because where you want a router is probably NOT anywhere
close
It is easier than that, I have found that the link event watching is
buggy and not very reliable. Here is what I do in my PSTN context:
exten = _X.,1,Wait(1)
exten = _X.,2,NoOp(CIDName: ${CALLERIDNAME} - CIDNum ${CALLERIDNUM} )
exten = _X.,3,Set(GROUP()=ZAP)
exten =
On Mon, Nov 10, 2008 at 8:13 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi,
I want to be able to bridge two sip channels using direct RTP
between my endpoints (Audio IP : not local) but without
using reinvites. So I set up my asterisk sip endpoints as follows:
[test1]
type=friend
Kristian Kielhofner wrote:
What version of Asterisk is this? Last I heard (from Olle) this
option was very experimental and should not be used on production
systems.
Oh.
Well.
That throws a wrench into the gears of a few uses of Asterisk as a
dedicated signaling-only B2BUA I was planning
Kristian Kielhofner wrote:
What version of Asterisk is this? Last I heard (from Olle) this
option was very experimental and should not be used on production
systems.
He even helpfully documented it that way in the sip.conf.sample file,
along with a list of (known) cases where it will fail,
Hi!
On Mon, Nov 10, 2008 at 5:16 PM, Pezhman Lali [EMAIL PROTECTED]wrote:
Dear,
is any way to change , the size of voice packets?
I want to increase the quality by decreasing the size of each packets,
because of bandwidth failure.
You can specify size of voice packets in allow line of
Get a core dump.
On Mon, Nov 10, 2008 at 12:26 PM, Jerry Geis [EMAIL PROTECTED] wrote:
I am using zaptel 1.4.12.1 and asterisk 1.4.18 - I also tried 1.4.21
lspci shows
00:00.0 Host bridge: VIA Technologies, Inc. VT8623 [Apollo CLE266]
00:01.0 PCI bridge: VIA Technologies, Inc. VT8633
Folks,
I'm tracing this error and looks like newer 1.4 (1.4.22) does not behave as
I stated (no DNS for channel domainname) and it must have been solved in the
way from my versions to the newer.
I'll update to the newer versions and confirm it.
Apologies for the flames :p
Samuel.
2008/11/10
I am using zaptel 1.4.12.1 and asterisk 1.4.18 - I also tried 1.4.21
lspci shows
00:00.0 Host bridge: VIA Technologies, Inc. VT8623 [Apollo CLE266]
00:01.0 PCI bridge: VIA Technologies, Inc. VT8633 [Apollo Pro266 AGP]
00:10.0 USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1
I have an asterisk system where the asterisk daemon dies typically at least
once per day. It is running in the wrapper safe_asterisk, which automatically
starts the daemon back up. But we find this unacceptable because when the
daemon dies, we usually have active calls drop, and sometimes we
naturally wooded does not bode well for WiFi
True, and it's even worse for the 5.6 GHz stuff that most of DECT is using
these days! The marketing departments have everyone convinced that bigger
frequency numbers are better. For most real-world environments the exact
opposite is true. The
It not only happens on INVITE or BYE but also happens when a 18x is received
(180 Ringing or 183 Session Progess at least) so it's not directly related
where the Dial command is executed in the dialplan, isn't it? It looks more
as a SIP channel internal stuff...
It looks like there is some
HI Shaun and Robb,
Thanks for the assistance.
I was able to force the codecs and had avaya talk in the right way. Also
addressed the DTMF issues.
I seem to be having issues with asterisk and avaya not detecting Hang ups.
i am using the Analog phones connected to the POTS ports on the IP Office.
We've found the Trixbox 2.6 series to be quite unstable which is why we're
still running 2.4 for those installations that require Trixbox.
Having someone tell you your daemon crashing and restarting is normal doesn't
seem very bright. Wouldn't the proper path be to find out why it is crashing?
Thomas Winter wrote:
On Sunday 09 November 2008 20:14, Eric ManxPower Wieling wrote:
The best (and maybe only way) is to set your client and your service
provider to only do G.723.
Really, thats not the way it should work.
How I can find out the codec of an incomming call?
Is there
Get a core dump.
Steve this is the stack trace still working on the dump.
Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread -1229517920 (LWP 9875)]
__ast_read (chan=0x8294a68, dropaudio=0) at channel.c:2016
2016f =
Igor Goncharovsky wrote:
You can specify size of voice packets in allow line of sip.conf peer
configuration.
ex.: allow=alaw:30,g729:50
If you are interfacing with any commercial VoIP equipment/media gateway
equipment, do be aware that packetisation durations for G.711u and G.729
that are
Does anyone here know anything about GEN-GEN analogue circuits, also
known as Manual Ring-Down (MRD)? Apparently they are widely used in
Hoot'n'Holler systems for financial dealer-boards.
I have been asked to try and interface to such circuits, and have been
having great difficulty locating any
Jerry Geis wrote:
Get a core dump.
Steve this is the stack trace still working on the dump.
Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread -1229517920 (LWP 9875)]
__ast_read (chan=0x8294a68, dropaudio=0) at channel.c:2016
2016f =
On Nov 10, 2008, at 3:30 AM, Igor Goncharovsky wrote:
Hi!
On Mon, Nov 10, 2008 at 5:16 PM, Pezhman Lali
[EMAIL PROTECTED] wrote:
Dear,
is any way to change , the size of voice packets?
I want to increase the quality by decreasing the size of each
packets, because of bandwidth failure.
-- Forwarded message --
From: Jerry Geis [EMAIL PROTECTED]
Date: Mon, Nov 10, 2008 at 1:36 PM
Subject: Re: [asterisk-users] console/dsp asterisk seg fault
To: asterisk-users@lists.digium.com
Jerry Geis wrote:
Get a core dump.
Steve this is the stack trace still working on
If the packetisation durations are different between endpoints, the SDP
offer/answer should fail with a 488 Not Acceptable Here. Right?
On Mon, November 10, 2008 2:17 pm, John Todd wrote:
On Nov 10, 2008, at 3:30 AM, Igor Goncharovsky wrote:
Hi!
On Mon, Nov 10, 2008 at 5:16 PM, Pezhman
On Mon, Nov 10, 2008 at 7:41 PM, David Budny [EMAIL PROTECTED]wrote:
What is the AMI command to see how many PRI channels are being used /
available?
Thanks
There is no direct command in AMI which will give you used channels number.
But you can easily keep track of the active zap
On Mon, 2008-11-10 at 10:10 -0700, Douglas Mortensen wrote:
I have an asterisk system where the asterisk daemon dies typically at
least once per day. It is running in the wrapper safe_asterisk, which
automatically starts the daemon back up. But we find this unacceptable
because when the daemon
On Mon, Nov 10, 2008 at 2:40 PM, Kristian Kielhofner
[EMAIL PROTECTED] wrote:
Depends?
What is the status of maxptime in Asterisk?
... or the remote end, for that matter...
--
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
On Mon, Nov 10, 2008 at 2:24 PM, Alex Balashov
[EMAIL PROTECTED] wrote:
If the packetisation durations are different between endpoints, the SDP
offer/answer should fail with a 488 Not Acceptable Here. Right?
Depends?
What is the status of maxptime in Asterisk?
--
Kristian Kielhofner
On Mon, 2008-11-10 at 10:50 -0700, Steve Murphy wrote:
On Mon, 2008-11-10 at 10:10 -0700, Douglas Mortensen wrote:
I have an asterisk system where the asterisk daemon dies typically at
least once per day. It is running in the wrapper safe_asterisk, which
automatically starts the daemon back
I have c-client installed on a 64bit system running Gentoo. I am trying
to run configure so I can test the IMAP voicemail functionality. But
asterisk-1.4.22 # ./configure --with-imap=/usr/include/imap
just gives me the following error.
checking for gnutls_bye in -lgnutls... no
checking for UW
Yes, but my conf is quite straightforward, isn't it?
No NAT etc...
I just want to know what is the combination of directives that I have to use
in order to achieve my goal.
Is there going to be any support in the future for this feature?
Because from the little I' ve seen in the mailing lists
Well, I should say irreconcilably different, not different.
On Mon, November 10, 2008 2:40 pm, Kristian Kielhofner wrote:
On Mon, Nov 10, 2008 at 2:24 PM, Alex Balashov
[EMAIL PROTECTED] wrote:
If the packetisation durations are different between endpoints, the SDP
offer/answer should fail
c james wrote:
I have c-client installed on a 64bit system running Gentoo. I am trying
to run configure so I can test the IMAP voicemail functionality. But
asterisk-1.4.22 # ./configure --with-imap=/usr/include/imap
just gives me the following error.
checking for gnutls_bye in
On 11/11/2008 1:34 a.m., samuel wrote:
Folks,
I'm tracing this error and looks like newer 1.4 (1.4.22) does not behave as
I stated (no DNS for channel domainname) and it must have been solved in the
way from my versions to the newer.
I'll update to the newer versions and confirm it.
Alex Balashov wrote:
If the packetisation durations are different between endpoints, the SDP
offer/answer should fail with a 488 Not Acceptable Here. Right?
Only if the 'ptime' or 'maxptime' values offered are not legal for the
codec involved; if they are supported by the codec, then 'ptime'
Kevin P. Fleming wrote:
Alex Balashov wrote:
If the packetisation durations are different between endpoints, the SDP
offer/answer should fail with a 488 Not Acceptable Here. Right?
Only if the 'ptime' or 'maxptime' values offered are not legal for the
codec involved; if they are supported
On Mon, Nov 10, 2008 at 4:08 PM, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Alex Balashov wrote:
If the packetisation durations are different between endpoints, the SDP
offer/answer should fail with a 488 Not Acceptable Here. Right?
Only if the 'ptime' or 'maxptime' values offered are not
Mark Michelson wrote:
c james wrote:
I have c-client installed on a 64bit system running Gentoo. I am trying
to run configure so I can test the IMAP voicemail functionality. But
asterisk-1.4.22 # ./configure --with-imap=/usr/include/imap
just gives me the following error.
checking for
On Monday 10 November 2008 05:56:18 Tony Mountifield wrote:
Does anyone here know anything about GEN-GEN analogue circuits, also
known as Manual Ring-Down (MRD)? Apparently they are widely used in
Hoot'n'Holler systems for financial dealer-boards.
I have been asked to try and interface to
Kristian Kielhofner wrote:
Are you confirming our understanding of the spec or Asterisk's
implementation of the spec?
Well the former, and I hope the latter too, since it should match the
former :-)
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - The Genuine Asterisk
On Mon, Nov 10, 2008 at 4:52 PM, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Kristian Kielhofner wrote:
Are you confirming our understanding of the spec or Asterisk's
implementation of the spec?
Well the former, and I hope the latter too, since it should match the
former :-)
--
Kevin P.
c james wrote:
Mark Michelson wrote:
c james wrote:
I have c-client installed on a 64bit system running Gentoo. I am trying
to run configure so I can test the IMAP voicemail functionality. But
asterisk-1.4.22 # ./configure --with-imap=/usr/include/imap
just gives me the following error.
On Tue, Nov 11, 2008 at 4:56 AM, Krishna Sumanth Chava
[EMAIL PROTECTED] wrote:
HI Shaun and Robb,
Thanks for the assistance.
I was able to force the codecs and had avaya talk in the right way. Also
addressed the DTMF issues.
Glad to hear it.
I seem to be having issues with asterisk and
I've been having trouble with making outbound calls to my
TELCO from a TDM400 card (FXS KS signalling) after upgrading
from 1.6-beta9 to 1.6.0. The problem is completely intermittent.
When it fails, I get this message:
[Nov 9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of
What cpu/memmory configurations have people had good luck with for a
small office asterisk server... (polycom's/poe linksys)
Dell's got their $200 SC440 going again until the 12th... and I'm
thinking it might be just the ticket...
___
-- Bandwidth and
Who you calling? Is it a remote non PSTN phone number? Or a PSTN number?
On Mon, Nov 10, 2008 at 7:21 AM, Doug Lytle [EMAIL PROTECTED] wrote:
sean darcy wrote:
[set-callerid-name]
exten = 0,1,NoOp( no CALLERID num set)
exten = 02025462677,1,Set(CALLERID(name) = Fred )
hi guys:
i have a question: when i dialed a number via sip channel to pstn,i want
to listen to my music instead of silence while the sip message was transfering.
I have tried the DIAL(,,A(.gsm)),but it always play the sound after the
channels have been connect!
what can i do
On Mon, 10 Nov 2008, nb wrote:
What cpu/memmory configurations have people had good luck with for a
small office asterisk server... (polycom's/poe linksys)
Dell's got their $200 SC440 going again until the 12th... and I'm
thinking it might be just the ticket...
You want to google 'asterisk
hii guys:
i get the message from the asterisk:
Started music on hold, class 'default', on Local/[EMAIL
PROTECTED],1
[2008-11-11 14:32:41] WARNING[1781]: format_wav.c:156 check_header: Unexpected
freqency 11025
[2008-11-11 14:32:41] WARNING[1781]: file.c:322 fn_wrapper:
The reason is your audio file is too high quality.
Asterisk can only play back audio file of 4000Hz.
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ??
Sent: Tuesday, 11 November 2008 5:35 PM
To: asterisk-users
Subject: [asterisk-users] music
On Mon, Nov 10, 2008 at 6:56 AM, Tony Mountifield
[EMAIL PROTECTED] wrote:
Does anyone here know anything about GEN-GEN analogue circuits, also
known as Manual Ring-Down (MRD)? Apparently they are widely used in
Hoot'n'Holler systems for financial dealer-boards.
I have been asked to try and
I am confused now. I called Polycom early October and was told to
submit a ticket for the latest firmware.
I submitted a ticket and was told to fuck off.
Dave posted a link and I can download firmware 3.1.1 from there.
The Firmware Table shows 3.1.0RevB as the newest firmware.
3.1.1. says
65 matches
Mail list logo
