I am looking to upgrade my asterisk server from using a single analog
X100P card to a T100P card. The PRI is already in the process of being
ordered, and I am wondering if there are any gotchas that I should be
aware of.
Also, is there any reason, other than the number of ports per PCI card,
Hi,
I have been tried to configure IAXphone (http://www.sokol-associates.com/IaxPhone.htm),
but I have the following problems:
- with auth=plaintext (and register indicator green):
chan_iax2.c:4341 socket_read rejected connect attempt from myip
- with auth=md5,plaintext,rsa :
chan_iax2.c:3168
As you point out, the TE405P and TE410P support 4 spans instead of just one.
They also support both E1 and T1, selected individually by span.
I believe the TE405P and TE410P are able to be a bus-masters instead of
slaves like the 100/400 series. Although this is touted as resulting in
The configuration
that I have is TE410P with 2 E1 trunks. Ports 3 and 4 on the board (channels
above 62) I do not use.
Every few minutes I
get this message:
Feb 19 11:52:03
WARNING[1209277232]: chan_zap.c:5949 zt_pri_error: PRI: Read on 86 failed:
Unknown error 500PRI got event: 8Feb 19
Hi everyone,
I have X100P connected to internal line of my Tenovis PBX.
I try to use * as simple DISA box.
When I have a call on this line, after user selects extension,
I want * to flash a line (to put on hold on Tenovis) and then
call selected extension.
Problem is that I cannot make Flash
Hello,
I configured one simple IVR
###
exten = 15001,1,Goto(ivrmenu,s,1)
;IVR
[ivrmenu]
exten = s,1,Ringing
exten = s,2,DigitTimeout,30
exten = s,3,Background(welcome.wav)
exten = 1,1,Dial(SIP/[EMAIL PROTECTED])
exten = 2,1,Dial(SIP/[EMAIL PROTECTED])
Hi,
-Original Message-
I configured one simple IVR
###
exten = 15001,1,Goto(ivrmenu,s,1)
;IVR
[ivrmenu]
exten = s,1,Ringing
exten = s,2,DigitTimeout,30
exten = s,3,Background(welcome.wav)
exten = 1,1,Dial(SIP/[EMAIL PROTECTED])
exten =
I had this with X-lite (on windows though) where I could hear it on one
end but not on the other. On the end where I couldn't hear audio I did
this
Advanced System Settings -- Audio Settings -- Silence Settings --
Transmit Silence -- Change this to Yes
That worked for us. Give that a shot.
Hi,
Does anyone operate Asterisk with zaphfc in NT-Mode successfully ??
I have the problem, that I could not contact my ISDN-Phone on such a
channel. It rings, but If I pick up the phone, I only get a Hangup in
the console
Thanks for any clues on it...
Here my setup:
3 HFC Cards.
first
Thanks - that gave me the basis for a couple of google searches.
Near the top of the script I put in
$SIG{HUP} = \exitGracefully;
and I added a subroutine that looks like this:
sub exitGracefully {
exit(0);
}
It now kills itself off without needing to be killed and take * with it.
Hi Ernst,
use this:
exten = 74341423,1,Dial(Zap/g2/74341423,r)
instead of:
exten = 74341423,1,Dial(Zap/5/74341423,r)
--
best regards
Klaus
--
Klaus-Peter Junghanns
CEO, CTO
Junghanns.NET GmbH
Breite Straße 13 - 12167 Berlin - Germany
fon: (de) +49 30 79705390
fon: (uk) +44 870 1244692
fax:
On Monday 16 February 2004 08:51 pm, Jamin W. Collins wrote:
Do any of you know of a cost effect device that could be connected to
an Asterisk station port to provide room monitoring? I'm looking to
replace the wireless baby monitor we currently have, since there is
too much interference
Do you see the green mic 'audio meter' on X-Lite moving up and down
with your voice? I found X-Lite's audio-in to work only intermittently
for me with my internal mic, and not at all with my iSight mic.
Eventually I opted to use SJ phone, which is working properly.
My Cisco 7960 is working well with * using SCCP, but I want to change it
to SIP.
Can anyone here help me on how/where I can buy a SIP image? I contacted a
few Cisco partners in the US and some replied will not sell 1 copy/can't
handle a small contract and others ignored me.
Thanks, Hermann
- Original Message -
From: Greg Hill
To: [EMAIL PROTECTED]
Sent: Wednesday, February 18, 2004 9:24 PM
Subject: Re: [Asterisk-Users] Room Monitor
On Wed, 18 Feb 2004, Jamin W. Collins wrote:
On Tue, Feb 17, 2004 at 10:04:02PM -0800, David Liu wrote:
Well use a Polycom IP 500 and
Buy SmartNet support for the phone. That grants you access to software images through
their website. Try Insight. 1-800-INSIGHT. They sell all quantities.
-sb
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Hermann Wecke
Sent: Thursday, February 19,
Title: H-324M/SIP Gateway
Hello,
Does any one knows about any H-324 to SIP gateway available in the marked? I heard about Radvision and Ericsson but I only could find a short description about the gateways in the companie's web pages.
Thank you very much.
Best regards,
João
On Thu, Feb 19, 2004 at 10:01:37AM -0500, Jim Flagg said:
- Original Message -
From: Greg Hill
To: [EMAIL PROTECTED]
Sent: Wednesday, February 18, 2004 9:24 PM
Subject: Re: [Asterisk-Users] Room Monitor
On Wed, 18 Feb 2004, Jamin W. Collins wrote:
On Tue, Feb 17, 2004 at
I have Vonage for two of my home phone lines.
Using ngrep, I'm seeing my [EMAIL PROTECTED] pop up
quite a bit.
Does anyone know if I can use this as an outside line, and if
I can, how?
Thanks!
Kristofer
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Hi Ryan..
No, I don't. I can see just the red one going up and down with my
voice. Strange is that on the installation test for mic and speaker,
everything was right. Anyways, I will try the SJPhone I see how it
would be...
Thank you
best regards
Osvaldo
On Feb 19, 2004, at 11:44 AM, Ryan
But nevertheless my mobile is still showing the number I'm dialing from.
Our provider is song networks which is a Danish Telco provider. If anymore
debug info is needed let me know
Hello again
Just wanted to say that on another location with exact same setup but
another telco provider,
I find it !!!
I have a bad syntax in extension.conf
Sorry
- Original Message -
From:
Rattana BIV
To: [EMAIL PROTECTED]
Sent: Thursday, February 19, 2004 2:27
PM
Subject: [Asterisk-Users] Problem with
call to IAX
Hi,
I've got a problem with Call to
Hi,
I had the exact same problem, and it was caused by my crappy ADSL
connection. I had great download and upload speeds too, but inspecting it
closer, there was a great deal of lost packets. The problem went away when I
changed my ADSL provider.
- Original Message -
From: yair hakak
Hey Ryan,
I got the same problem with the SJPhone. Is there something different
on the sip configuration for softphone? I'm using:
allow=gsm
[1234]
type=friend
insecure=yes
nat=yes
username=1234
callerid=Osvaldo
secret=xx
host=dynamic
canreinvite=no
qualify=200
context=default
Osvaldo
On
Am Do, den 19.02.2004 schrieb Ernst Lehmann um 15:26:
Am Do, den 19.02.2004 schrieb Klaus-Peter Junghanns um 15:00:
Hi Ernst,
Hi Klaus-Peter,
use this:
exten = 74341423,1,Dial(Zap/g2/74341423,r)
instead of:
exten = 74341423,1,Dial(Zap/5/74341423,r)
I found the bug in my
After 2 weeks on bug #981 (Dropped Channels during dual redirect), I
just posted 2 patches and almost have it fixed. The only problem is,
the patch has a side effect of leaving some zombies. That is, the
zombie channel that is created during the masquerade process doesn't get
hungup. If I
Hi,
If I have two sip t.38 enabled gateways connected to asterisk, will I be
able to send a fax from one to the other with the mediastream passing
through asterisk ?
Thanks,
Dave
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[EMAIL PROTECTED]
You can use AGI, the example below uses asterisk-perl:
---
#!/usr/bin/perl -w
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-setcallback(\mycallback);
$number = $AGI-get_data(input-number, 1, 8);
$AGI-say_number($number);
exit 0;
sub mycallback {
Title: FW: H-324M/SIP Gateway
Sorry, in my last email body é mean H-324M to SIP gateway instead of H-324 to SIP.
Thanks
João
-Original Message-
From: Joao Sampaio
Sent: quinta-feira, 19 de Fevereiro de 2004 15:12
To: '[EMAIL PROTECTED]'
Subject: H-324M/SIP Gateway
Hello,
my config is pretty bare-bones:
note that I am only allowing ulaw (not gsm)
allow=ulaw
[1000]
type=friend
username=1000
secret=password
host=dynamic
context=from-sip
mailbox=1000
On 19-Feb-04, at 9:06 AM, Osvaldo Mundim wrote:
Hey Ryan,
I got the same problem with the SJPhone. Is there
I used an external gateway - a Mediatrix 1204. It's nice because the voice
streams are offloaded to it, reducing any load on the server. But it has a
nightmare setup and interface, and it's kind of expensive.
On Thursday 19 February 2004 00:27, Chad Brown wrote:
What is my best bet If I want
I could be wrong but I think I remember seeing mention of recommendation
about the number per server although I don't remember the number.
- Original Message -
From: Christian Hecimovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, February 19, 2004 12:26 PM
Subject: Re:
Hello there,
I have a system up that I can receive calls to now (can navigate the demo
script successfully)
However, when I try to run one of the EAGI tests I get the following.
Any ideas?
-- Accepting AUTHENTICATED call from 66.234.228.132, requested format = 4,
actual format = 2
--
Hi,
I'm a new Asterisk user who is having trouble getting his IP 500 to
register with Asterisk (I've already gotten things up and running with
X-Lite softphones just fine).
I have an X-Lite softphone on extension 2000, and the polycom on
extension 2002. I can dial '2002' from X-Lite and my
this is what i got from digium re digital interfaces. so i guess, you can
use this as basis of your design.
We only recommend 2 TE410P cards and 3 TDM Cards in a machine. You would
probably run into IRQ sharing if you use any more cards or the cpu wouldn't
be able to handle the amount of calls
try putting something in for these values instead of leaving them blank in
the phone1.cfg file:
reg reg.1.displayName=2002 reg.1.address=2002 reg.1.label=2002
let us know if that clears things up.
MATT---
-Original Message-
From: James Treleaven [mailto:[EMAIL PROTECTED]
Sent:
Also since you are using Asterisk::AGI you can register a callback that
gets called when most of the AGI commands return error/hangup.
James
On Thu, 19 Feb 2004, Tim Petlock wrote:
Thanks - that gave me the basis for a couple of google searches.
Near the top of the script I put in
$SIG{HUP}
I'm trying to compile gastman on Mandrake 9.2 using gcc 3.3.1 and I get
the following error:
gui.c: In function `gui_init':
gui.c:944: warning: passing arg 2 of pointer to function from
incompatible pointer type
gui.c:944: warning: passing arg 4 of pointer to function makes pointer
from
On Thu, Feb 19, 2004 at 10:28:09AM -0500, Walt Reed wrote:
Hmm. Is it just me, or does this sound like a sledgehammer for a
thumbtack kind of application?
Radioshack has cheap intercoms that work fairly well. They have 900Mhz
wireless and FM over powerline versions. Most cheap baby
Hi all
Having a very much bit of an oddity with some phones connected to a
Rhino 24 port FXO Channel Bank off a TE410P... Though I can find similar
references to this throughout the asterisk-users list I can't find a
solution.
If you pick up a phone, then replace the receiver, the phone will
I sounds like it may not be your Asterisk configuration which is at issue
here. If the phone has a dirty hookswitch, the on-hook transition can be
interpreted by the channel bank as both on-hook and loop-start. Combined
with the channel being active (from the channel bank's point of view - as
the
Title: mgcp endpoint question
Hi,
I'm testing an mgcp phone with * 0.72. While * accepts the endpoint gateway enclosed in [ ], the phone baulks when presented with the gateway domain not enclosed in [ ], see below. Is there anything I can add to my mgcp.conf to force the inclusion of the [
Hi all
Another oddity for you...
24 port FXO Rhino Channel Bank connected to a TE410P card on Span 1, 6
channel Q.931 PRI connected to Span 2 (Telewest not BT so using Nortel
Equipment), Caller ID will not appear on the client phones (ADSI
PT350s).
The CLI says:
-- Executing Dial(Zap/25-1,
Hi all,
I was wondering if someone would mind giving me an example
of how to configure a fax machine with asterisk. We would ideally like to have
2 fax extensions configured, one is a modem for faxing the other is an actual
fax machine. Im not sure if I should configure them as standard
On Thursday 19 February 2004 12:17, Tony Buser wrote:
I'm trying to compile gastman on Mandrake 9.2 using gcc 3.3.1 and I
get the following error:
gui.c: In function `gui_init':
gui.c:944: warning: passing arg 2 of pointer to function from
incompatible pointer type
gui.c:944: warning:
Hello Users,
I am attempting to create a sip connection in the following network:
Sipgate.de -- Internet -- Gate -- Asterisk PBX -- Some Extension
Gate, the gateway and nat'ing firewall has sip udp (5060) traffic and
rtm udp (8000 to 8020) traffic forwarded to the asterisk pbx machine.
Both
Never mind about the zombies. I fixed 'em real good...
I didn't think I would find the solution so quickly. Thanks anyway.
- Matt
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To
Matt Lawson wrote:
Never mind about the zombies. I fixed 'em real good...
I didn't think I would find the solution so quickly. Thanks anyway.
- Matt
Could you post your solution for posterity?
-
Andrew Thompson
http://aktzero.com/
___
[EMAIL PROTECTED] wrote:
Is it possible with * to take make a SIP phone go off hook and dial
remotely? I assume this is a function of the phone. The scenario is to have
a script initiate a call from a given phone as if the user had dialed it
from the phone. This can be done with a softphone, but
On Thu, 19 Feb 2004, Jamin W. Collins wrote:
Actually the baby monitors tend to be in the 47Mhz band, but yes they
still suck. There are newer models in the 900Mhz and 2.4Ghz range.
However, reviews of the 900Mhz models are almost unanimous in declaring
them to be worse than the 47Mhz
I don't think so... KewlStart is just LoopStart with disconnect?
supervision... either should work...
The debounce settings in the Asterisk configuration affects how Asterisk
handles hookswitch transitions on it's FXO/FXS interfaces. Since you are
using a channel bank connected to a T1 interface,
$8/yr from Cisco.
-Heison
On Thu, Feb 19, 2004 at 01:59:13PM -0700, Chris Hirsch wrote:
Bisker, Scott (7805) wrote:
Buy SmartNet support for the phone. That grants you access to software
images through their website. Try Insight. 1-800-INSIGHT. They sell all
quantities.
Given
Is it possible to have the system outdial and take surveys. either by
receiving DTMF or voice?
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To UNSUBSCRIBE or update options visit:
On Thu, 19 Feb 2004, Chris Hirsch wrote:
Bisker, Scott (7805) wrote:
Buy SmartNet support for the phone. That grants you access to software
images through their website. Try Insight. 1-800-INSIGHT. They sell
all quantities.
Given than I'm interested in getting a Cisco phone off something
Yes, look at /usr/src/asterisk/sample.call, put it in /var/spool/asterisk/outgoing and
have it connected to an extension that does the survey.
-Heison
On Thu, Feb 19, 2004 at 02:15:28PM -0700, PBXtech wrote:
Is it possible to have the system outdial and take surveys. either by
receiving DTMF
Is it possible to have the system outdial and take surveys. either by
receiving DTMF or voice?
Yup. Just have the system use the outgoing queue (see sample.call) and
have it call an AGI script upon answering.
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[EMAIL
Is there a way to record phone conversations. I am using Asterisk with a
IP phone and the digium hardware to make ouside calls.we need to have
all outside calls
One S100U USB FXS Interface (including the USB cable)
One X100P PCI FXO Interface
the system is working quite well. but we have got a
Title: Message
Do you
need to have a zap interface for it to work?
B.
J.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bisker,
Scott (7805)Sent: Wednesday, February 18, 2004 15:34To:
[EMAIL PROTECTED]Subject: RE:
anybody came up with something on this? :)
i am also planning to replace our SER with *. it has most of SER's features
and is extensible as the SER. but how many concurrent calls it can handle? I
have no answer. Plus, there is still problem on voicemail on recorded
message using G729-format.
hi.
Assuming calls will be using G711, a little of g729 (max-15), 1000 SIP
(multi-vendor: Cisco, GS, MOtorola, Dlink,etc), of which 80% clients are
behind NAT, and server of I-P4 2GHz, 80GB HD and 4GRAM, will that work?
depends on how many concurrent calls you have.
you can have 10k users, but
hi.
mind that current CVS already has mixing support,
if soxmix is installed into the sistem,
so loligo.com exten file could be simpler.
matteo.
I would recommend checking out this link.
http://www.loligo.com/asterisk/current/extensions.conf
Darren Wiebe
[EMAIL PROTECTED]
kemal asad
I am interested in subscribing to a service that will let me
dial the PSTN in Ireland and am
interested in what the community thinks about who has the best services
available. I would prefer to purchase time in blocks of minutes or pay as I go
in lieu of having a monthly fee to contend
Hi Ernst,
On Thu, 2004-02-19 at 15:26, Ernst Lehmann wrote:
use this:
exten = 74341423,1,Dial(Zap/g2/74341423,r)
snip
Interesting fact is, that the ISDN-Phone on the NT line rings still, if
the calling phone has dropped the call..
The same thing here.
wkr,
--
Envida
Ernest W. Lessenger wrote:
Any news on when 800 numbers will be available?
Thanks,
--Ernest
For small quantities of numbers, just point your existing 800 at your
DID.
It probably won't deliver the 800 as the number dialed, but you could
order a new did for each 800 you want to
Is it possible to have the system outdial and take surveys. either by
receiving DTMF or voice?
Yup. Just have the system use the outgoing queue (see sample.call) and
have it call an AGI script upon answering.
If you want CDR data, be sure you connect to an extension that starts the
AGI.
For those of you buying smart net support contracts, just because you
can download the software does not mean you are licensed for it. If you
have a cisco phone with a sccp image on it, it is most likley licensed
with that image. You would need to buy a license for SIP from cisco in
order to
if I need to buy 150 phones can I order them with the SIP image?
Ryan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Matthew Enger
Sent: Thursday, February 19, 2004 3:19 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco 7960 SIP image
On Thu, 2004-02-19 at 19:00, Tim Petlock wrote:
I need to do something like this because I've timed calls with a
stopwatch and can't figure out why the records going into the CDR table
are 20 seconds longer (or more) than the actual call time. I understand
that the actual call time includes
Michael Graves wrote:
Well my Pingtel Xpressa arrived today. It's configuration is nowhere
near as clear as the SNOM 200. Can someone here provide some guidance
on getting the Xpressa talking to *?
Michael
Michael,
In my experience, the Xpressa most times registers with * but never
Hi,
I am call Japan via Voicepulse. My IAX Connection to Voicepulse was
sucessfull. But when I put a call (dial), I get an error message:
Feb 19 22:12:23 WARNING[81926]: chan_iax2.c:1128 attempt_transmit: Max
retries exceeded to host 66.234.228.132 on IAX2[voicepulse]/4 (type = 6,
subclass =
For starters voicepulse is down again at the moment.
matt
Daniel Bichara wrote:
Hi,
I am call Japan via Voicepulse. My IAX Connection to Voicepulse was
sucessfull. But when I put a call (dial), I get an error message:
Feb 19 22:12:23 WARNING[81926]: chan_iax2.c:1128 attempt_transmit: Max
Have played with several languages in Asterisk - by using the
SetLanguage() command in the extensions.conf.
The scenario is that I do not want every caller to use the same
language, although they do use the same extension.
E.g. H323 users can have a different language than SIP user. Read in
I am currently looking at using the app_sql_postgres.c stuff, but it
almost immediately occurred to me that there is no way of guaranteeing
that the whole of the extension code will be executed. As a result I
also can't be sure that the postgresql connections will actually be closed.
I realise
Getting a toll-free number can be done through any major carrier. Andrew, you might
as well contact me with your rates so I can compare because I am always looking to
save money. Contact me at [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On
Hi Michael,
Michael Welter wrote:
I live at 8000' in the Rockies. We have lots of woodpeckers--they
especially love to drill 4 holes in the north side of my house.
They also like to drill on the arial telephone cables. Water then
gets into the cable and causes a partial grounding on the
They seem to be a mickey-mouse organization. They don't have a phone
number, they don't answer their support emails, and they don't seem to
provide working service. I've been trying to troubleshoot an inbound line
provisioned with them but they aren't answering their emails. outbound
calls seem
I usually use
[EMAIL PROTECTED]
they do eventually get back to you.
We operate a call centre and have offered them an inbound package, but
it seems they are not interested.
Matt
P.S. Our DID line hasn't been working for around a month nowin the
process of signing up with other
Hi,
I am trying to connect through call2ua with no success.
It seems to be authentication problem.
Anyone could inform to me how is the Dial/H323 parameter to authenticate
with them and dial?
Thanks,
Isamar
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My VP has been up all day without any problems.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of matt
Sent: Thursday, February 19, 2004 8:16 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IAX Connection - Voicepulse
I usually use
[EMAIL
At 05:15 PM 2/19/2004, you wrote:
I usually use
[EMAIL PROTECTED]
they do eventually get back to you.
We operate a call centre and have offered them an inbound package, but
it seems they are not interested.
Matt
P.S. Our DID line hasn't been working for around a month nowin the
process of
FYI - bug 1043 has been fixed on Feb 18:
From my log, below, you will see that ast_rtp_bridge is not comparing
the codecs properly. Asterisk is currently comparing the integers, and
not the bits of the codec.
In the below example codec0 = 260. That means Codec0 allows both 256
(g729) and 4
What other companies have you found? We've used NuFone, but aren't too
impressed by their payment and CDR interface (i.e. email the
salesperson). Otherwise they seem to be stable and knowledgeable.
=
Just signing
Matthew B Marlowe wrote:
My VP has been up all day without any problems.
Strange...our's wenty down an hour ago and I went onto #asterisk to see
if anyone else was having problems and their service is down also...
are you using gw5.voicepulse.com?
Matt
On Fri, 20 Feb 2004, matt wrote:
What other companies have you found? We've used NuFone, but aren't too
impressed by their payment and CDR interface (i.e. email the
salesperson). Otherwise they seem to be stable and knowledgeable.
Ours is back up again now...in the hour it was down we had all staff on
extended lunch break and I signed up with two new providers.
I wonder why you get special treatment?
:-)
Matt
Matthew B Marlowe wrote:
Yes, I am.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hi!
Just signing up now to NikoTel...
They responded within 5 minutes of my mail...normally I'd discount this
as I am asking for a service, but most of the companies have even had
trouble replying to me to set up an account.
They also react fast on trouble tickets - so far my experience
Wonder why mine is different..also why does mine say unmonitored?
Yours:
pbx*CLI iax2 show peers
Name/Username Host Mask Port Status
isprime-peer-ma 66.230.128.53 (S) 255.255.255.255 4569 OK (7ms)
voicepulse 66.234.228.132 (S) 255.255.255.255 4569 OK (7ms)
nufone 66.225.202.72 (S)
Add qualify=yes in iax.conf
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of matt
Sent: Thursday, February 19, 2004 9:03 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IAX Connection - Voicepulse
Wonder why mine is different..also why does mine
Get a spectrum analizer.
Software will do it. Record the humming connetion to a file
and then run it through software that plots a power spectrum.
THere is plent of good open source software. Even some audio
file ditors have this feature. You should be able to see the
hum as periodic peaks
Hi Michael,
Michael Welter wrote:
the codecs I use? Filter-out everything between, say, 55 and 65Hz?
Notching may not be that effective, as it will not deal with the
harmonics. The analogue to digital converter should already be
filtering below 300Hz, so you probably have quite a lot of
On Thu, 19 Feb 2004, Nick Bachmann wrote:
300Hz is pretty high to filter out... it's still well within the rage of
voices. To compare, 300Hz is about a diatonic concert D.
POTS has been filtering out 300HZ and 3000HZ for years.
--
Joel
___
I've released a beta version of the new firefly, to
address the crashing issue with incoming calls from Asterisk. (the problem was I
assumed the caller id would be populated). Also, firefly will now reject calls
if there's no common codecs.
I'd recommended anyone using firefly with asterisk
(Philipp von Klitzing) wrote:
FYI - bug 1043 has been fixed on Feb 18:
From my log, below, you will see that ast_rtp_bridge is not comparing
the codecs properly. Asterisk is currently comparing the integers, and
not the bits of the codec.
In the below example codec0 = 260. That means Codec0
Hello List,
Just thought I would post an update, I got asterisk to register with
sipgate.de.
I was wrong, it was my firewall (maybe).
Here is the way a normal nat under openbsd pf works:
udp 192.168.1.100:5060 - 24.102.192.227:(random port) - 217.10.79.9:5060
but add this line to pf.conf
Dear Matt,
I have never used shady dial but would be happy to try and get it
working over IAX for you. Here is how I see this working; during you
initial R D phase I would put you on one of my development servers
with a PRI T1 that I use for testing. Your cost during the RD phase
will be
I have the same problem, most carriers out there deal with both g723.1 or
g729. During passing through via Asterisk, carrier customers will send us
calls broadcasting both codecs with one having priority over the other, the
way it is supposed to work is that asterisk will try to negotiate the top
I'm wondering whether people know if there could be a problem
with * receiving retransmissions of INFO/DTMF requests.
I'm trying to play DTMF via INFO to *. If it takes a 200 reply too
long to come back, the request is retransmitted. Whenever this
happens, the IVR down in PSTN reports that the
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