Excellent answer. Thank you very much.
Paul
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andreas Frackowiak
Sent: Saturday, May 15, 2004 1:32 AM
To: [EMAIL
Hello
I am looking for Czech (Czech Republic) country support to indications.conf
Have you ever seen it anywhere ?
We are a small country in middle Europe :)
thank you
--
Vit Bohacek
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[EMAIL PROTECTED]
Please take a look at http://www.snom.com/faq/FAQ-04-04-28-ut.pdf. It
describes the hand/headset policy! It was supposed to be an improvement...
CS
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger
Sent: Thursday,
8 kHz 16 bit/sample (linear) mono WAV files.
CS
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ernest W. Lessenger
Sent: Thursday, May 13, 2004 7:31 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] 2.05a firmware
Does
I have problem to change from handsfree mode to handset mode. When I
switch from handset to handsfree while waiting for connection I press
the green speakerphone button once. It is all well. Once it is connected
I don't want to give the called party too much echo and I want to switch
it back
While we're at the 2.05 firmware - the DTMF handling on the Codec
configuration page have disappeared. I assume this is because the phone now
got some kind of default behaviour based on the codec. Can you describe
that behaviour?
-Original Message-
From: [EMAIL PROTECTED]
Hi David,
I have problem to change from handsfree mode to handset mode. When I
switch from handset to handsfree while waiting for connection I press
the green speakerphone button once. It is all well. Once it is
connected I don't want to give the called party too much echo and I
want to switch
Hi all.
I was wondering about how to set different tones, in the Asterisk I use
indications.conf, in the Cisco ATA-186 I use the webinterface.
How do I set tones in the Grandstream, handytone, Cisco 7960 ?
The US tones does not apply to all countries. (Unfortunatley)
/Mike
hi,
do you have
nationalprefix=0
internationalprefix=00
in your zapata.conf?
best regards
Klaus
--
Klaus-Peter Junghanns
CEO, CTO
Junghanns.NET GmbH
Breite Strasse 13a - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax: (de) +49 30 79705391
iaxtel:
Perhaps stupid question but, is Asterisk a statefull or stateless proxy?
Ignace
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
On 16 May 2004, at 23:17, Aaron Clauson wrote:
maybe they just do it different here in Ireland.
They do it differently in Ireland. To get a functioning modem cable you
need to have a cable that takes the outside two wires and crosses them
to the inside two wires. We had hundreds of these made
may not correct but i tought * is not a proxy.
Ignace CARIA wrote:
Perhaps stupid question but, is Asterisk a statefull or stateless proxy?
Ignace
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[EMAIL PROTECTED]
G'day list,
Follow-up post to the one I sent last week regarding bad calls between SIP
and ISDN.
On Fri, 14 May 2004, Vic Cross wrote:
To me, it looks like a variation of the SIP RTP timestamp problem (yes, my
7960 is at 6.3 code), but the problem exists on the ATA-186 too and I
don't have
(Forwarded:)
- Original Message -
From: Usman Tahir [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: Christian Stredicke [EMAIL PROTECTED]
Sent: Monday, May 17, 2004 11:25 AM
Subject: Re: [Asterisk-Users] 2.05a firmware
Hi,
In firmware release 2.05c and later, the user has an option to
The gun issue highlights the absurd nature of Louderback's opinions.
-Original Message-
From: Steve Totaro [mailto:[EMAIL PROTECTED]
Sent: Friday, May 14, 2004 3:20 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] OMG THE SKY IS FALLING!! NOT!!!
Although I agree with eveything you
Yup
Juan J. Sierralta P. wrote:
On Sat, 2004-05-15 at 12:22, Michael Welter wrote:
I've gotten several Power alarm on module 1, resetting since I
installed a quad FXS TDM400 card. Dell 400sc.
Does your motherboard have the A-B-C-D LEDS above the keyboard/mouse
connectors?
I suppose
Follow-up post to the one I sent last week regarding bad calls between SIP
and ISDN.
On Fri, 14 May 2004, Vic Cross wrote:
To me, it looks like a variation of the SIP RTP timestamp problem (yes, my
7960 is at 6.3 code), but the problem exists on the ATA-186 too and I
don't have any
Hi All,
Could anyone tell me which is the recommended hardware to a system
running voicemail and conference, attending four E1 trunks and,
another, attending only one E1?
Can I use a PIII 850Mhz?
Thanks in advance.
Robert Almeida
As I understood, Asterisk has a lot of features but lacks native 3-way
calling and attended transfer. It would be great to have these features
available to a simple IAX phone.
I wonder how this could be implemented in Asterisk without asking for a
patch. It should be possible with parking,
Hi folks,
I'm trying to make an * PBX for a customer using 4 X100Ps
and 1 TDM400p(4FXS).
The problem I'm facing is to make one unique IRQ for each
PCI slot/board since shared IRQs create all kind of weird noises
and echos.
Anybody got any workaround for that?
Any recommended motherboard to
Ignace CARIA wrote:
Perhaps stupid question but, is Asterisk a statefull or stateless proxy?
Asterisk is a no-SIP-proxy-at-all :-)
Again: Asterisk is not a SIP proxy. A SIP proxy doesn't handle media streams and
doesn't terminate or originate calls. Asterisk does.
Asterisk is a stateful SIP UAS
On Mon, 17 May 2004, Rich Adamson wrote:
It sounds like you've nailed the problem with the signed int statement.
markster doesn't think so. Apparently this is normal ;)
However, I'd suggest you open a bug report on this (rather than using list
mail only) to get it some attention and
Those of us that use Cisco 7940/7960 sip phones know that we've been
impacted by two very different changes that have occurred over the last
couple of months.
First, when cisco created sip v6.x code, they implemented a new DSP (as
well as other software changes) that effectively drops any
This has been mentioned before on this list, but in order for md5.c to
compile successfully (OpenBSD 3.3), the following must change in md5.c:
#if defined( __FreeBSD__ ) || defined( __OpenBSD__ )
# include sys/endian.h
Change this to be:
#if defined( __FreeBSD__ ) ||
Duane wrote:
Grandstream v1.0.4.68 firmware
http://www.hellofone.com/downloads.html
Seems to have loaded ok on my BT100..
Hmmm well I need to kinda figure out how to get the custom ringtones to
ring on the phone... :-)
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Asterisk-Users mailing list
On Mon, 17 May 2004, Robert Almeida waxed:
Could anyone tell me which is the recommended hardware to a system
running voicemail and conference, attending four E1 trunks and,
another, attending only one E1?
Can I use a PIII 850Mhz?
Maybe for a single port E1 card, maybe. You'll
Asterisk is not a SIP Proxy, It's a soft PBX. But it is a SIP registrar, and
forwarding is stateful, i think. I could be wrong.
Regards, Girish
From: nicolas [EMAIL PROTECTED]
Subject: [Asterisk-Users] Re: Asterisk Proxy Type
Date: Mon, 17 May 2004 11:31:16 +0200
may not correct but i tought *
Hello all,
I've noticed several messages about the latest firmware on Snom's site,
2.05b, and today I see that another update is listed, 2.05c. However,
when I go to the download page (http://www.snom.com/support_dl_en.php),
the latest firmware version available for the Snom200 is 2.04g.
Are
Try this one. Took me a while too.
http://www.snom.com/download/share/snom200-2.05c-SIP.bin
-Original Message-
From: M3 Freak [mailto:[EMAIL PROTECTED]
Sent: Monday, May 17, 2004 11:42 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Snom200 Firmware: I only see 2.04g
Hello
Hi all,
I have just put a message from a few days with a problem with
CAPI hangup. I have noticed that line with 97% of hangs, is a line
connected with a ATA286 with a modem-fax. Could it be the problem?
Regards,
srsergio
___
Asterisk-Users
Christian,
That's the wonderful thing about VoIP phones... Just upload new firmware and
we can have the best of both worlds! (Thanks for making the change in
2.05c.) Great phones, by the way :)
--Ernest
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
You know what would be cool? A Show Variables command in the cli. It could
return something like this...
VariableScope Channel
=
CallerIDC ZAP/1-1
EPOCH G
EXTEN C ZAP/1-1
The silence last 60s (aprox)
On Sun, 2004-05-16 at 21:38, Juan J. Sierralta P. wrote:
On Sun, 2004-05-16 at 15:15, Jorge Verastegui wrote:
Hi
Please help!
I have one X101P and TDM400P in my asterisk Box
When i make a call from * to PSTN, everything goes Ok,
When the PSTN
Yes. I do.
Maybe though do I have a problem with my dialplan.
I have :
pridialplan = unknown
prilocaldialplan=national
This is for France. Would it explain ?
Thanks.
- Original Message -
From: Klaus-Peter Junghanns [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, May 17, 2004
Just a suggestion to anyone using speex:
Try running the 1.1.5 or svn code rather than 1.0.3.
As a quick example, here are the show translation outputs from * on a
2.8 GHz P4 with speex 1.0.3 (from debian sid's .deb) and from 1.1.5
(compiled with CFLAGS=-march=pentium4 and --enable-sse).
Note
Firstly, amazing software, props to all the developers.
I'm trying to compile the latest asterisk cvs checkout and keep getting
an error which I can't solve, any help would be much appreciated -
make[1]: Leaving directory `/usr/src/asterisk/stdtime'
if [ -d CVS ] ! [ -f .version ]; then echo
(As a side note, these were captured before Brian's ilbc Makefile
patch made it to the anon cvs tree; that optimization shaved 5ms
off the time to encode to iLBC on that box.)
Major improvements in speex... I'm impressed you did what I was going to
work on today :P I started on this quest
Hi all,
I use span_dsl (opencall.org) for faxing, libtiff 3.5.7-12 is installed.
But when trying to receive a fax, i get the following error:
[...]
Fast carrier up
Coarse carrier frequency 1699.97 (66)
Training error 7.516780
Training succeeded (constellation mismatch 9.892637)
Fast carrier
On Mon, 2004-05-17 at 17:35 +0100, Nicholas Ruddick wrote:
I'm trying to compile the latest asterisk cvs checkout and keep getting
an error which I can't solve, any help would be much appreciated -
res_crypto.c:25:25: openssl/ssl.h: No such file or directory
res_crypto.c:26:25:
Install openssl-devl
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Nicholas Ruddick
Sent: Monday, May 17, 2004 11:36 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Redhat 7.3 compiling problem
Firstly, amazing software,
I can't believe you are still on 7.3, but whatever.
On Mon, 2004-05-17 at 11:35, Nicholas Ruddick wrote:
res_crypto.c:25:25: openssl/ssl.h: No such file or directory
res_crypto.c:26:25: openssl/err.h: No such file or directory
Those two entries alone solve whats wrong. When you don't have .h
Looks like you need to install the openssl development packages and make
sure you are doing an 'export LANG=C'. Some of the stuff did not
compile on my redhat system (9.0) because it was set as
LANG=en_US.UTF-8. The first problem is indeed it cannot find your
openssl headers.
--
Matthew
http://asterisk.bkw.org/diff/translate.patch.txt
If you try that patch out it adds a nice feature...
show translation recalc [xx]
You can throw more than 1 sample thru it and recalculate your translation
matrix. It also allows you to see TRUE translation under a load or just
when ever you feel
Voiptalk provide an excellent service and great support.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian Potkin
Sent: 10 May 2004 23:51
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SIP in the UK
On Mon, May 10, 2004 at 08:58:23AM +0100,
hi,
dialing without answer is descriped in serveral docs.
But if i try to do so * sends an invite message ever and ever to my phone.
My phone (snom) sends a busy here back and the call is cancled.
When i answer before i dial all is ok.
Where is the problem ? make i everything wrong ?
nicolas
Steven Critchfield wrote:
I can't believe you are still on 7.3, but whatever.
Keeping it old school :-)
Those two entries alone solve whats wrong. When you don't have .h files,
you are missing the -dev or -devel packages for whatever it is they
belong to. In this case, you are missing the
I see this (but not using a recent asterisk version)
I had put it down to a software bug in the grandstream phones that I'm using -
are you sure its an asterisk bug or are you using grandstream also?
Steve
On Mon, 17 May 2004, John Vogel wrote:
I upgraded to the latest stable version of
Don't know what else to call this. Googling and some time on the IRC
channel haven't gotten me anywhere.
Here's the sitch, which is a bit complicated but is something my
customers are in fact encountering on an everyday basis:
1. Bob is on a Zap channel talking through the PSTN to Carol.
On Mon, 2004-05-17 at 09:57, Olle E. Johansson wrote:
Ignace CARIA wrote:
Perhaps stupid question but, is Asterisk a statefull or stateless proxy?
Asterisk is a no-SIP-proxy-at-all :-)
Again: Asterisk is not a SIP proxy. A SIP proxy doesn't handle media streams and
doesn't terminate or
If you are using ethereal to decode packet traces that include iax2
packets, you may have noticed that codecs such as ilbc were being
shown as unkown.
I've had a patch accepted into the ethereal cvs that corrects that,
updating packet-iax2.[ch] to match asterisk cvs HEAD.
I presume it will be in
During Astricon 2004, we'll have the first Asterisk developer's meeting.
The Asterisk developer's meeting is a one day meeting with discussions, brainstorms
and tests.
For each session, we need a white paper produced that outlines the topic to be
discussed.
If controversial, several whitepapers
Juan J. Sierralta P. wrote:
On Mon, 2004-05-17 at 09:57, Olle E. Johansson wrote:
Ignace CARIA wrote:
Perhaps stupid question but, is Asterisk a statefull or stateless proxy?
Asterisk is a no-SIP-proxy-at-all :-)
Again: Asterisk is not a SIP proxy. A SIP proxy doesn't handle media streams and
I was wondering if someone could shed some light on using SRV records in
conjunction with *. I have a domain in my sip.conf file that utilizes SRV
records to direct SIP traffic when using the regular domain name (i.e.
sip:[EMAIL PROTECTED]) in the SIP uri to a SNOM 4S proxy to which this
On Mon, 2004-05-17 at 12:10, Jorge Verastegui wrote:
The silence last 60s (aprox)
So maybe is the timeout used by your Telco (Entel?) Here at Chile we
use 30s to let called people to be able to hang and get the call on
another phone plugged to the same line.
So I think its better
Has anyone implemented the externnotify feature that is mentioned on the
wiki with the enhanced voicemail?
I have tried to invoke the command both in the general section as well
as a part of the user mailbox definition with no luck.
The explanation of the feature is as follows:
Externnotify
On Mon, May 17, 2004 at 05:11:40PM -0400, Kevin wrote:
Has anyone implemented the externnotify feature that is mentioned on the
wiki with the enhanced voicemail?
I have (I wrote it :-) ).
Externnotify
Want to run an external program whenever a caller leaves a voice mail
message for a
I'm having a weird problem with IAX2 in today's CVS HEAD. I have two boxes
with T100P cards connected via IAX2. Calls between them work fine, but when
I press a key at one end, it comes out the other end as a click, with no
tone. I've tested the DTMF on the T1 using SendDTMF with an outgoing
Thanks for the quick response. I am trying to implement a solution for
voicemail outcall notification for individual users. There is a
suggested solution posted in the wiki but it has some limitations thus I
was looking for an alternate solution.
-Original Message-
From: [EMAIL
Ever since I updated to CVS-head from 10 May, something weird has been
happening...
Every night at 1:10 AM (Eastern Australian time) my phone rings, there is no
callerid, and it results in a message in Voicemail which is just the
disconnect beeps (due to the inability of being able to detect
I have set up an extension so I can dial it and listen to my MusicOnHold from
any handset. This is what is in the extensions.conf:
exten = 997,1,MusicOnHold()
exten = 997,2,Hangup
After 180 seconds of playing, the call terminates. Why does this happen?
Simon
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Isamar Maia wrote:
| Hi folks,
|
| I'm trying to make an * PBX for a customer using 4 X100Ps
| and 1 TDM400p(4FXS).
| The problem I'm facing is to make one unique IRQ for each
| PCI slot/board since shared IRQs create all kind of weird noises
| and
when asterisk has more than 50 h323 calls it craps out on me. Can anyone
help?
May 17 10:45:35 WARNING[1769581]: chan_sip.c:1114 create_addr: No such
host: 19149191120-- Executing
Dial(H323/ip$66.238.200.224:32943/16164,
H323/[EMAIL PROTECTED]/1957408) in new stack-- Called
[EMAIL
I want to do this same thing, does anyone have an example of how to do it?
using a zap fxo and zap fxs card how can I set up caller announce? like
this.
1 call comes in and a prompt asks the called to identify themselves.
2 the system would then put the caller on hold and pick up the FXS
Thanks Chris, but I will use only voicemail and conference, I think that
is better 4 Pentium III boxes that one dual pentium box only. Do you
think that it can attend 30 channels?
regards
Robert Almeida
On Mon, 2004-05-17 at 17:23, Robert Almeida wrote:
Could anyone tell me which is the
Issue an Answer first!
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Simon Brown
Sent: Monday, May 17, 2004 4:38 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Music on hold
I have set up an extension so I can dial it
Hint use app_parkandannounce with a twist of app_record.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Gavin Hollinger
Sent: Monday, May 17, 2004 4:50 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] call announce?
I want
Hi all I am trying to compile Asterisk on RH9
I am have installed
pwlib_1.5.2
readline-4.3-5
readline-devel-4.3-5
openssl-devel-0.9.7a-2
openssl-0.9.7a-2
openh323_1.12.2
I read that it has to do with pwlib not being installed correctly so i made
sure it was in /usr/local correctly
When i try
Not sure it is an Asterisk bug - I am using GrandStreams. Will upgrade their
software and also try it on my Snom. Thanks.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen J.
Wilcox
Sent: Monday, May 17, 2004 12:45 PM
To: [EMAIL PROTECTED]
Where?
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of brian
Sent: Tuesday, 18 May 2004 8:04
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Music on hold
Issue an Answer first!
bkw
-Original Message-
From: [EMAIL PROTECTED]
Try 'export LANG=C' then 'make clean make'
--
Matthew Billings | Affordable WWW Internet Solutions
foreThought.net | for Small Business
[EMAIL PROTECTED] | 910 16th Street, #1220 (303)
228-0070 x821
--The Future is Now!--| Denver, CO 80202
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Isamar Maia wrote:
| Hi folks,
|
| I'm trying to make an * PBX for a customer using 4 X100Ps
| and 1 TDM400p(4FXS).
| The problem I'm facing is to make one unique IRQ for each
| PCI slot/board since shared IRQs create all kind of weird
At 06:23 PM 5/17/2004, you wrote:
gives the same error...
g++ -Wall -DPNXVER2=2 -DP_LINUX -mcpu=i686 -D_REENTRANT -DP_HAS_SEMAPHORES
-O3 -
DNDEBUG -DP_SSL -I../include -I/include -I/crypto
-DPNX_VERSION=\CVS-05/17/04-1
7:10:07\ -DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN -I/include/ptlib/unix
exten = 997,1,Answer
exten = 997,2,MusicOnHold()
exten = 997,3,Hangup
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Simon Brown
Sent: Monday, May 17, 2004 5:22 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Music on hold
John Vogel [EMAIL PROTECTED] wrote:
I upgraded to the latest stable version of 1.0 today and am still seeing
the *8 problem where the phone that was originally dialed keeps on
ringing even after another phone picks up.
Are other people also seeing this? Has somebody figured out how to make
Does parkandannounce create a variable with the parked extension number that
I could use in later linking the calls?
Hint use app_parkandannounce with a twist of app_record.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
Im a total newbie at this telephony stuff but I'm
putting together a low cost PBX for my small company and wanted a check on the
h/w Im planning on ordering and my system configuration. Any input is
appreciated. Take it offline and email me directly if appropriate ([EMAIL PROTECTED]).
Looks good to me.
You will want hunting or call forward busy on the
phone lines you order. Mine costs $1.15 per month
- Original Message -
From:
Mike Stupak
To: [EMAIL PROTECTED]
Sent: Monday, May 17, 2004 4:59 PM
Subject: [Asterisk-Users] total newbie
Question 5 will be harder.
What you need is DID ( Direct Inward
Dialing)
Not available in my area with regular phone
lines.
Perhaps it could be done with distinctive
ring?
Gavin
- Original Message -
From:
Mike Stupak
To: [EMAIL PROTECTED]
Sent: Monday,
DID is inbound only. DID will not work if you plan to use the same trunks for outbound calls. Do the Digium cards support DID? Normally DID lines require an external power supply which connects to the card. I don't remember seeing anything like that on the Digium analog cards.
Distinctive
Mike Stupak wrote:
Im a total newbie at this telephony stuff but I'm putting together a
low cost PBX for my small company and wanted a check on the h/w Im
planning on ordering and my system configuration. Any input is
appreciated. Take it offline and email me directly if appropriate
([EMAIL
When i make a call from Asterisk everything goes Ok,
I do have a problem: when a call from the PSTN originates, the extension
in Asterisk hangs up and I only hear silence in the PSTN for
approximately 60 seconds.
On Mon, 2004-05-17 at 16:56, Juan J. Sierralta P. wrote:
On Mon, 2004-05-17 at
Actually it encodes a second of data, which with a 20ms codec would be
50 frames. The timing shows better than expected results due to caching.
-Adam
brian wrote:
http://asterisk.bkw.org/diff/translate.patch.txt
If you try that patch out it adds a nice feature...
show translation recalc [xx]
You
Paste your extensions.conf
Check the answer command if you're running IVR of special services.
- Original Message -
From: Jorge Verastegui
To: [EMAIL PROTECTED]
Sent: Monday, May 17, 2004 8:46 PM
Subject: Re: [Asterisk-Users] X100P problem with PSTN from BOLIVIA
When i make a call from
Yes I realized my error in my wording but it was early :P It doesn't
improve alot but does give you some ways to get a better idea of translation
times if your box is loaded up with calls.
bkw
PS this patch was added to CVS-HEAD
- Original Message -
From: Adam Hart [EMAIL PROTECTED]
Hello,
Have done some research on the Wiki and via Google, but have not found
anything that describes what the tables and columns in the rate engine
database really mean. Does anyone have any documentation on those?
Thanks,
Darrin Johnson
Systems Engineer
IS Domain Inc.
Btw, Good work. 5ms is a huge different, espically in optimizing terms.
I've added a few flags and shaved off another ms
here's my flags: (only for p4/xeon)
-fPIC -O3 -march=pentium4 -funroll-loops -fomit-frame-pointer -msse
-msse2 -mfpmath=sse
keep up the good work,
Adam
brian k. west wrote:
Does call-waiting work for anyone that recieves 2 pstn calls on a X100P using a Sipura?
I have modified the dialplan in the Sipura such that the *0 is definately getting sent
to the Asterisk server now.
When the phone beeps and I flash hook I get tone, then dial *0# and the sip debug
shows
I toyed with -msse and -mmmx and others too but couldn't put any of those
in. :P
bkw
- Original Message -
From: Adam Hart [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, May 17, 2004 6:20 PM
Subject: Re: [Asterisk-Users] speex
Btw, Good work. 5ms is a huge different, espically
Follow-up post to the one I sent last week regarding bad calls
between SIP
and ISDN.
On Fri, 14 May 2004, Vic Cross wrote:
To me, it looks like a variation of the SIP RTP timestamp problem
(yes, my
7960 is at 6.3 code), but the problem exists on the ATA-186 too and
I
don't have any
Hi there,
::: i have an mgcp phone (iph-90) using the sample mgcp.conf for it
(the dlink section). i've tried both asterisk stable and development
release but i'm getting the following error when i lift the receiver:
. .. in stable branch:
-- MGCP mgcp_new(MGCP/aaln/[EMAIL PROTECTED]) created
Steve Creel wrote:
snip
I have heard complaints that once every couple weeks, when a user picks up
their analog phone (t1 span off of a TE410P into an Adtran 750 with 6 FXS
cards), they don't get dialtone, but instead, hear another conversation.
I'm under the impression that they can only hear and
I'm having a problem with outgoing dropped calls. They symptom is, when I
place a call from a sip extension to the outside, the call is connected
properly, but then abruptly disconnects anywhere from 10 to 60 seconds
later. This happens when the outgoing call is through a POTS line (TDM)
as well
do a 'sip debug' and make sure all looks good.
TL
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Bruce Komito
Sent: Monday, May 17, 2004 10:06 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Dropped calls
I'm having a problem with outgoing dropped
Hi
http://www.cisco.com/univercd/cc/td/doc/product/tel_pswt/vco_prod/
this is a useful link, (but not specific for Czech Republic )
I am looking this support for Bolivia (South America)
On Mon, 2004-05-17 at 03:20, Dudlik wrote:
Hello
I am looking for Czech (Czech Republic) country
MattB wrote:
Try 'export LANG=C' then 'make clean make'
Huh?
Jeremy McNamara
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I'm not seeing this - using stable CVS from 14-05-2004.
Phone types are Grandstream Budgetone 101 using firmware 1.0.4.68, and Cisco
7940 using SIP 6.2.
-Shaun
On Mon, 17 May 2004, John Vogel wrote:
I upgraded to the latest stable version of 1.0 today and am still seeing
the
*8 problem
Now it's always proposing out of band DTMF. If there should be a user agent
which does not support RFC2833 it will answer the SDP accordingly and then
the phone automatically falls back to inband DTMF.
The setting was previously necessary because some equipment could not deal
with this
Hi all I am trying to compile Asterisk on RH9
I am have installed
pwlib_1.5.2
readline-4.3-5
readline-devel-4.3-5
openssl-devel-0.9.7a-2
openssl-0.9.7a-2
openh323_1.12.2
I read that it has to do with pwlib not being installed correctly so i made
sure it was in /usr/local correctly
When i try
Hi,
The incoming caller id on the X101P always comes up scrambled except
when there is no name, just a number. Usually a cellphone would do this,
and the number is perfect.
I was reading posts about using ztmonitor to capture the spill and
listening to it. The resulting file is alway 0 bytes...
brian == brian k west [EMAIL PROTECTED] writes:
brian I toyed with -msse and -mmmx and others too but couldn't put
brian any of those in. :P
The options -msse, -msse2, -mmmx et al are all implied by the
relevant -march options. uname only reports i686, so you have to use
some other construct
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