.call files are through the manager.
An simple app exists, and should make it online very soon on
www.astertest.com (just cleaning up the code to make it a bit more user
friendly atm).
At 20:26 21/10/2004, you wrote:
Is there a way to load test IAX? I know I can setup long duration calls
via
Randy Bush wrote:
i come from an automated ip backbone world where we generated
configs automatically from sql data tied to the back office and
sales systems. i want to have a shipping person take a new spa3k
out of the box, plug it into an ether, hit the 'Confirm' button on
the customer order
Hello,
During asterisk bootup, I've been having a fun time with a random delay
which can be quite long, from what seems to be the codec_ilbc.so file.
I notice in verbose mode the cost is rather high, and was hoping someone
will have some insight on what's going on here.
Prior to a harddrive
I have seen similar things in the past, but only during startup.
When started, do a show translation and look again, if that value is ok,
you can ignore the one on startup.
Zoa.
At 12:06 23/10/2004, you wrote:
Hello,
During asterisk bootup, I've been having a fun time with a random delay
which
joachim wrote:
I have seen similar things in the past, but only during startup.
When started, do a show translation and look again, if that value is
ok, you can ignore the one on startup.
*CLI show uptime
System uptime: 27 minutes, 2 seconds
*CLI show translation
Translation times between
If you look very hard, you can find two versions on trabas on the web, an
old one, not working and a new one not complete and not installing. (the
SQL files are incomplete for example)
If you combine both, and you are extremely patient you might be able to get
it to actually display something
Could you give us more information on:
Distro, kernel version, compiler, makefile flags, version of asterisk, and
hardware on your machine, + loaded modules ?
GSM to LPC10 is also way tooo slow.
-
*CLI show uptime
System uptime: 27 minutes, 2 seconds
*CLI show translation
Miroslav Nachev wrote:
Dear Olle,
I can say that Emil Ivov has very good knowledge on IPv6 too. You
can use it.
Great - the more IPv6 experts that can help us with coding advice,
code review and patches - the better!
/O
___
Asterisk-Users
Chad,
I need a more complete SIP debug than just one packet to try to look into this
issue. If the device registers, both a REGISTER transaction and a subsequent
call with the ACK - THank you!
/O
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
joachim wrote:
Could you give us more information on:
Distro, kernel version, compiler, makefile flags, version of asterisk,
and hardware on your machine, + loaded modules ?
GSM to LPC10 is also way tooo slow.
Sure. Running Fedora Core 2 with latest updates (kernel 2.6.8-1.521) on
a Celeron
Trevor Peirce wrote:
Sure. Running Fedora Core 2 with latest updates (kernel 2.6.8-1.521) on
a Celeron 1.70 GHz chip. Half a gig DDR ram, one generic X100P card
with it's very own IRQ.
Asterisk is the latest CVS. It's about time for bed.. spent too many
hours trying to figure out other things
Could you tell us what RAID card you are using + what drivers you are using
for it.
Could you try to run it without the raid card ?
Zoa.
At 12:35 23/10/2004, you wrote:
Trevor Peirce wrote:
Sure. Running Fedora Core 2 with latest updates (kernel 2.6.8-1.521) on
a Celeron 1.70 GHz chip. Half
On Friday, October 22, 2004 2:40 PM
Stewart Nelson wrote:
I presently have a small VoIP network using H.323 and gnugk,
and would like to upgrade it to an Asterisk-based system,
primarily to take advantage of low cost unlimited calling
plans offered by SIP providers such as Vonage.
FYI
Hi list,
I have the following setup : a first asterisk is connected to the legacy
Alcatel PaBX to connect to a remote site with a second asterisk server.
PSTN
|
Legacy phones == Alcatel Omnipcx == Asterisk1
|
On Sat, 23 Oct 2004 02:16:44 -0700, Trevor Peirce [EMAIL PROTECTED] wrote:
G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC
G723 - - - - - - - - - - -
GSM - - 2 2 4 2 1 1238 - - 529695
Is there a FREE third party module for webmin ?.
How much bandwidth do I have to reserver in order to get a good call quality
?.
Let's say I have 20 people calling each other.
Is 1MB of bindwidth Ok or can I reserve even less ?.
To your experience what is the minimum compression to get
Thanks a lot, I tried, the string for the busy tone (from Voxzilla) is :
[EMAIL PROTECTED];10(.5/.5/1)
I also tried with [EMAIL PROTECTED];4(.5/.5/1) for a shorter detection delay (my pstn
provider
doesn't play the tone for 10 seconds). It still doesn't work.
The sipura support told me before
As of version 4.59a, no, it does not support NAT. Rumor had it that Uniden
was going to release new firmware for the phone in October, but it's not
there as of right now, it has not been posted on their web site.
Lyle
- Original Message -
From: Me [EMAIL PROTECTED]
To: Asterisk Users
Hi all
Problem with gnophone:
I can not make a call. (just hangs)
Im am a novice to Asterisk but quite experienced Linux user. I am having
some problems with the gnophone. I have tried to isert my user/password
but nothing have changed.
I have tested the michrophone and it is working. The sound
Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote:
On Sat, 23 Oct 2004 02:16:44 -0700, Trevor Peirce [EMAIL PROTECTED] wrote:
G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC
G723 - - - - - - - - - - -
GSM
On Sat, 23 Oct 2004 15:06:15 +0200, Yves-Marie CRABBE [EMAIL PROTECTED] wrote:
Thanks a lot, I tried, the string for the busy tone (from Voxzilla) is :
[EMAIL PROTECTED];10(.5/.5/1)
I also tried with [EMAIL PROTECTED];4(.5/.5/1) for a shorter detection delay (my pstn
provider
doesn't play the
I'm using IAXComm on the Mac to connect to my Asterisk system and it
all seems to work well when I'm connected to my wired network. When I
use wireless instead, IAXComm never registers with Asterisk and when I
call, ASterisk seems to think it's connected but no sound comes back.
My Asterisk
Olle,
No...Thank you! You are the perfect guy to look at this problem as well
since ultimately I need to switch to chan_sip2 given the outboundproxy
functionality.
My testing shows that not only stable has this issue but so does head.
That said, the problem could carry over to chan_sip2.
Still looking for some feedback...
We are trying to configure our * box to receive RDNIS using ISDN PRI
circuits from a Lucent 5ESS so that when a call gets forwarded to the vmail
system (using call forward no answer) we get the original dialed digits to
identify the mailbox owner.
The local
On Sat, 23 Oct 2004 17:10:40 +0200, Neal Nelson [EMAIL PROTECTED] wrote:
I'm using IAXComm on the Mac to connect to my Asterisk system and it
all seems to work well when I'm connected to my wired network. When I
use wireless instead, IAXComm never registers with Asterisk and when I
call,
I have succesfully integrated some phonejacks with Zaptels. I am able to
transfer calls from my tdm board to my phonejack (from quicknet) using the
hangup button (pressing it once). But I am unable to do this the other way
around with the quicknet board.
This works :
Phonejack --Digium
Just an FYI
show translation recalc 10
Give that a go.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Benjamin on Asterisk Mailing Lists
Sent: Saturday, October 23, 2004 4:56 AM
To: Asterisk Users Mailing List - Non-Commercial
I have some ASM200 and ASM400, these are analog gatewyas,
The ASM 200 - 2 fxs 2 fxo ports (only two simulatenous calls)
The ASM 400 - 4 fxs 4 fxo Ports ( only 4 simulatenous calls )
My intention is to integrate them with Asterisk, so that I can use their
FXS channels as internal extensions in
Kevin Walsh wrote:
Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote:
On Sat, 23 Oct 2004 02:16:44 -0700, Trevor Peirce [EMAIL PROTECTED] wrote:
G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC
G723 - - - - - - - - - -
Does anyone have any experience with running Asterisk on dedicated
servers from any of the cheap hosting providers, like 11?
I'd like to get my asterisk/mail/web server out of my house. There
isn't a whole lot of traffic involved, but I'd rather not end up with
someplace that *utterly*
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Scott Laird
Sent: Saturday, October 23, 2004 12:37 PM
To: Asterisk Users Mailing List
Subject: [Asterisk-Users] Cheap hosted servers and Asterisk
Does anyone have any experience with
is there a doublehash patch for 1.0.1?
o old one to res_parking.c does not apply as there is no longer
res_parking.c
o wiki search is useless
o google only finds the problems applying old patch to 0.7
thanks
randy
___
Asterisk-Users mailing
I would like to let asterisk send an URL to a PC based softphone or a PC based
message client. This would allow for many great applications such as automatic
client data lookup. Or for technical client support. It is a must for many
types of customer support centers.
I understand that the send
Scott,
I use an AMD 2400 hosted in The Planet (www.theplanet.com) to host my
asterisk box currently. They don't directly offer AMDs but a provider
that colocates there does. $60/mnth. SeverMatrix.com is the low end
dedicated biz of The Planet directly. It is only 60ms from my home in
NJ even in
I bought one of these phones and I am trying to set it up.
So far, I have figured out how to get to the web interface but I can't seem
to figure out how to set some of the most important things like the Proxy
address etc..
The manual is useless for things like this as well as their website. The
Hitete wrote:
Is there a FREE third party module for webmin ?.
How much bandwidth do I have to reserver in order to get a good call quality
?.
Let's say I have 20 people calling each other.
Is 1MB of bindwidth Ok or can I reserve even less ?.
To your experience what is the minimum compression
http://bugs.digium.com/bug_view_page.php?bug_id=0002460
Give that a whirl
Bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Randy Bush
Sent: Saturday, October 23, 2004 11:46 AM
To: splatters
Subject: [Asterisk-Users] doublehash
We just got setup with Broadvoice
yesterday for LD. This isnt something I REALLY need (No local numbers
avail so we just got a Houston number),
but Im just curious. I can make outbound calls to Broadvoice
and they work great, but I cant do inbound. I have bvs voicemail turned off so all I
Eric Wieling wrote:
Kevin Walsh wrote:
Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED]
wrote:
Can you set up a test call where Asterisk will transcode from ulaw to
ILBC and see what it does to your CPU load?
How should I go about creating such a test call?
Also, try recalculating the
and the patch take19.txt in bug 0002010 does not apply cleanly
to the freebsd port of 1.0.1
randy
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On Sat, 2004-23-10 at 08:14 -0500, Lyle Giese wrote:
As of version 4.59a, no, it does not support NAT. Rumor had it that Uniden
was going to release new firmware for the phone in October, but it's not
there as of right now, it has not been posted on their web site.
Yes - the new firmware is
http://bugs.digium.com/bug_view_page.php?bug_id=0002460
This patch includes the double key hangup patch too which lets you define
what you want.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Randy Bush
Sent: Saturday, October
On Friday 22 October 2004 02:05 pm, Neil Cherry wrote:
David Ishmael wrote:
I think my Netgear router will try to lease the same DHCP address to a
device based on MAC automatically each time the device queries for an
address (but I'm not 100% sure about that, never really watched it). So
On Wednesday 20 October 2004 04:47 pm, Matt Hess wrote:
Remember, you pay for what you get.. especially with Dell networking
equipment. I have heard about several groups who tried the dell switches
only to give up on them because the dell switches just didn't perform.
Yes, price-wise they look
On Wednesday 20 October 2004 04:08 am, Jay Wilton wrote:
Hello,
The Smc 8508T goes for about $95, jumbo frame support,
lifetime warranty but no QOS. The Netgear GS608 is $ 100,
no jumbo frames, 1 year warranty, QOS, gig latency 10U max.
The 3com switch reviews that I read were not happy.
On Thursday 21 October 2004 09:16 am, Matt Hess wrote:
There was a thread on NANOG a while back about dell switches and the
opinion at the time seemed almost in complete agreement - dell switches
stink for everything but pure ipv4 shuffle packets.. unmanaged without
any features.
They are not
Has any one integrated to a Geotel with Asterisk?
Thanks.
Greg
Advanta
___
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To UNSUBSCRIBE or update options visit:
Ok lets get this out of the way... WTF is Geotel?
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Greg Smith
Sent: Saturday, October 23, 2004 4:19 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Geotel integration with
Hitete wrote:
Is there a FREE third party module for webmin ?.
How much bandwidth do I have to reserver in order to get a good call quality
?.
Let's say I have 20 people calling each other.
Is 1MB of bindwidth Ok or can I reserve even less ?.
To your experience what is
I just signed up for the BroadVoice service a few hours ago, but for
the life of me I can't get any incoming voice. The incoming
connection is fine as it rings my extension from outside, but I can't
hear anyone talking. Outgoing voice is working fine though.
I've been looking through the
So in keeping with the topic, the GS phones work well with the Asterisk
system? Should I get a GS phone or is there another phone that I should
consider? Since this is for my home rather than a company, I just want
something that will work with little fuss. ;)
-Original Message-
From:
GS is fine for that
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Ishmael
Sent: Saturday, October 23, 2004 5:49 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Re: cannot call Grandstream
So in
Thomas Hutton wrote:
Mr Kielhofner, you answer nothing, while adding to the noise you
complain about.
Googling for information on the webmin module leads to nothing. The
webmin module on the digium FTP site is worthless. Can somebody talk
Jamie Cameron into writing one?
Most people will tell
On Sat, 23 Oct 2004, Terry Evans wrote:
I just signed up for the BroadVoice service a few hours ago, but for
the life of me I can't get any incoming voice. The incoming
connection is fine as it rings my extension from outside, but I can't
hear anyone talking. Outgoing voice is working fine
Hi guys I know this has been asked on the list before, but my
hard drive crashed and I lost all of the past posts, I need to know what
motherboard works ok for asterisk, I have no problems with the Dual and
Quad Xeon processor boards I have used. Now I plan on building a Pentium
4 3.0 with
On Sat, 23 Oct 2004, Tim Jackson wrote:
We just got setup with Broadvoice yesterday for LD. This isn't something
I REALLY need (No local numbers avail so we just got a Houston number),
but I'm just curious. I can make outbound calls to Broadvoice and they
work great, but I can't do inbound. I
Because I don't want to clog the list with more never-ending
discussions that seem to be so popular lately, I probably won't reply
to this thread any longer. I feel that I have gotten my point across.
Yes... You have...
And good points you made...
There are people on this list who
Look for support by whatever operating system you plan on running.
Henry Devito wrote:
Hi guys I know this has been asked on the list before, but my hard
drive crashed and I lost all of the past posts, I need to know what
motherboard works ok for asterisk, I have no problems with the Dual
and
[EMAIL PROTECTED] wrote:
is there a doublehash patch for 1.0.1?
o old one to res_parking.c does not apply as there is no longer
res_parking.c o wiki search is useless
o google only finds the problems applying old patch to 0.7
I've attached an old-school, no frills, double-hash patch
--On Saturday, October 23, 2004 19:56 -0400 Stan Brinkerhoff
[EMAIL PROTECTED] wrote:
Look for support by whatever operating system you plan on running.
I second thatpretty much any P4 based hardware should be perfectly fine
for asterisk. I'd tend to lean towards SCSI drives though, but
Stan Brinkerhoff wrote:
Look for support by whatever operating system you plan on running.
Henry Devito wrote:
Hi guys I know this has been asked on the list before, but my hard
drive crashed and I lost all of the past posts, I need to know what
motherboard works ok for asterisk, I have no
Any chance you can pass me the Beta Version or let me know how to get it
myself?
I love this phone except for this problem, either way I guess I will keep it
and wait for the new firmware since it's a nice phone overall.
Thanks,
Todd
--
Start Your Own ISP!
http://www.YourOwnISP.com
-
--On Saturday, October 23, 2004 19:39 -0500 Kristian Kielhofner
[EMAIL PROTECTED] wrote:
Hey,
This is probably a good time to ask if there is any planned support for
a g729 binary for YDL and G3/G4, etc. I would love to start playing with
apple hardware, YDL, and asterisk. But I need
Hello,
On Sat, 23 Oct 2004 08:54:27 -0700, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
I would like to let asterisk send an URL to a PC based softphone or a PC based
message client. This would allow for many great applications such as automatic
client data lookup. Or for technical client
Just tried the patch you made with the latest CVS and it patches fine
although it does not work. Now when I hit # it does not send the DTMF
to the other side at all. Although hitting ## does get the transfer.
Now # doesn't do ANYTHING :)
On Sat, 23 Oct 2004 19:00:38 -0500, Barton Hodges
Michael Loftis wrote:
--On Saturday, October 23, 2004 19:39 -0500 Kristian Kielhofner
[EMAIL PROTECTED] wrote:
Hey,
This is probably a good time to ask if there is any planned
support for
a g729 binary for YDL and G3/G4, etc. I would love to start playing with
apple hardware, YDL, and
[EMAIL PROTECTED] wrote:
Just tried the patch you made with the latest CVS and it patches
fine
although it does not work. Now when I hit # it does not send the
DTMF
to the other side at all. Although hitting ## does get the
transfer.
Now # doesn't do ANYTHING :)
I'm not sure why that is,
I'll be running the Red Hat Enterprise. I thought I saw people posting
certain motherboards had issues with sound, I know I saw where others said
to stay away from the VIA chipset.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stan
Brinkerhoff
Sent:
Trevor Peirce wrote:
I have noticed that when * is first loading, CPU usage goes to 100%
for exactly the same duration that it takes that ilbc codec to load.
Upon closer inspection, it seems that every time a caller is hears MOH,
a new mpg123 is spawned. Right now top is showing 8 mpg123's
Upon closer inspection, it seems that every time a caller is hears MOH,
a new mpg123 is spawned. Right now top is showing 8 mpg123's running,
and between then and * CPU utilisation is maxed out. About 30% user and
65% system.
IMPOSSIBLE... What mpg123 version are you running?
bkw
Brian West wrote:
Upon closer inspection, it seems that every time a caller is hears MOH,
a new mpg123 is spawned. Right now top is showing 8 mpg123's running,
and between then and * CPU utilisation is maxed out. About 30% user and
65% system.
IMPOSSIBLE... What mpg123 version are you
REMOVE THAT POS and install mpg123 0.59r, compile from src.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Trevor Peirce
Sent: Saturday, October 23, 2004 8:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Brian West wrote:
REMOVE THAT POS and install mpg123 0.59r, compile from src.
Done and done. FYI you may want to update
http://voip-info.org/wiki-Asterisk+mpg123+redhat as that's where I got
inspired to download the RPM.
I just stopped asterisk and killed off all the mpg123 processes... ran
Trevor Peirce wrote:
Trevor Peirce wrote:
I have noticed that when * is first loading, CPU usage goes to 100%
for exactly the same duration that it takes that ilbc codec to load.
Upon closer inspection, it seems that every time a caller is hears
MOH, a new mpg123 is spawned. Right now
Or for that matter, is there a planned G729 binary for Mac OSX ?___
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Wo trevor, Format and start over? Don't go crazy, just remove the files
created by make install.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Trevor
Peirce
Sent: Saturday, October 23, 2004 10:09 PM
To: Asterisk Users Mailing List - Non-Commercial
I am installing a new * server using Fedora Core 2 but I ran into a
problem after I installed the X100P. When FC2 boots it runs KUDZU to detect
new hardware and it detected the card and insists on loading the module
crc_ccitt before the zaptel module. Because of this I cannot load the wcfxo
Also quietmp3nb: and you'll only have one process per music class.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Trevor Peirce
Sent: Saturday, October 23, 2004 9:09 PM
To: Asterisk Users Mailing List - Non-Commercial
Done and done. FYI you may want to update
http://voip-info.org/wiki-Asterisk+mpg123+redhat as that's where I got
inspired to download the RPM.
Repeat after me... RPM is bad source is good.
I have put a nice warning on that page. Its already been proven to use
0.59r and you'll notice the
--On Saturday, October 23, 2004 21:35 -0500 Brian West [EMAIL PROTECTED]
wrote:
Done and done. FYI you may want to update
http://voip-info.org/wiki-Asterisk+mpg123+redhat as that's where I got
inspired to download the RPM.
Repeat after me... RPM is bad source is good.
I have put a nice
Stay away from boards with VIA chipsets, those are problematic in my
experience. I have had some good results with the D865PERL boards from
Intel, along with several other Intel boards. Those seem to be of high
quality. They may not have the very best performers, but the PCI bus
is implemented
Chad,
I noticed you wrote this earlier (see
below). I have the same problem with the chan_sccp module with a Cisco
7910 phone. I have traced down the * crash to a reference to an undefined
variable. Adding the speeddial entries would fix the issue, but I am VERY
unclear on the format.
Stay away from boards with Intel chipsets. Those are problematic in my
experience. The FX, LX, 820, 840 and various others have been extremely
flaky, and caused no end of problems. :-)
VIA used to be bad, but seem to get steadily better. Intel are just
erratic. I think most makers have made
On Sun, 24 Oct 2004 01:45:19 +0200, Stewart Nelson [EMAIL PROTECTED] wrote:
I looked at NuFone.net and some others, but it appears that
IAX is not right for my system.
I'd say this is only because you don't know enough about IAX yet ;-)
I live near Reno, NV, and
have a second home in Paris.
I wanted to use Outlook 2000 to dial my Contacts using Asterisk. So I
installed AstTapi on my Windows XP machine. When I try to dial a contact,
the call originates just fine. My SIP phone rings, and when I pick up,
Asterisk makes the call to the dialed number correctly.
However, Outlook displays
It also sounds like some type of NAT issue to me, but I can't figure
out what's going wrong. I changed the RTP ports back to 1-2
and set the router up to forward those, but still no incoming voice.
Kevin suggest I try the two inbound sections in the sip.conf, but I
had already tried them
I'm having the same issue, and I'm not behind NAT.
Maybe this is a BV issue?
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Terry
Evans
Sent: Saturday, October 23, 2004 4:33 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk and
Caveat: I've only got about three weeks of experience working with Asterisk
so it's possible I've completely overlooked a more obvious solution
to this issue. Snide comments are welcomed if this is the case.
One of the more puzzling frustrations I've faced in working with
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