Hello Tom,
On 26-Aug-2005 7:50, Tom wrote:
Most of the time i can find answers to my questions on the wiki, google,
or searching the list now i am stuck .
I have a small * box at my house running 1.0.9 stable and a devlite kit.
Every thing is awesome VM, IVR, Echo canceling, and Meetme are
Title: Message
Has anyone here
managed to get UK Caller ID (BT) working
using a TDM400P card?.
I've gotthe latest drivers from CVS, but
can't find clarification if UK caller ID is supported and if so what the
settings should be.
Cheers
Graham
I finally figured out that echo directives and channel specific stuff needs
to go between group and channel otherwise it didn't work or just gave weird
results. I still have a problem with fax detection in terms of it turning
off echo canceling. I have tried both, incoming, and everything in
Try different entry in this parameter.
In Italy mobiles start with 3, while public services with 1 and normal user
numbers with 0.
Using pridialplan=none, every number different from 0 was resulting in
termination code 1, normally used for number never seen on the network.
Ciao
Mauro
Hi,
I do have two questions regarding call queues:
1) How can I reach that waiting calls are also removed on removing the
last agent listening to the queue. All I found is the switch to
prevent new calls enter the queue after the last agent left.
2) Currently my queue does ring the agent
Mauro Zanin wrote:
Try different entry in this parameter.
In Italy mobiles start with 3, while public services with 1 and normal user
numbers with 0.
Using pridialplan=none, every number different from 0 was resulting in
termination code 1, normally used for number never seen on the network.
[EMAIL PROTECTED] wrote:
I just setup a Dell 1800, not a 2850, which is working
awesome.. only had to
disable USB, which realistically no-one on a phone system
would care about
anyways.
Oh, really? Only if you're running a 2.6 kernel or using
a zaptel card you don't need it.
--
Andreas
I often get a Maximum retries error while making outgoing calls. Why
does this happend? Most of the time a reload solves the problem, but not
all the time? What to do?
Aug 26 09:52:46 WARNING[6613]: chan_sip.c:694 retrans_pkt: Maximum
retries exceeded on call [EMAIL PROTECTED]
for seqno 1
Hello,
I have the following problem when I send fax to asterisk:
-- Executing RxFAX(IAX2/[EMAIL PROTECTED]:4569/2,
/var/spool/asterisk/fax/_1125039307.1.tif) in new stack
2005-08-26 06:55:09 NOTICE[30852]: channel.c:1317 ast_read: Dropping
incompatible voice frame on IAX2/[EMAIL
Hi,
I intend to use asterisk realtime. I have test it with sip.conf and extension.conf. It works fine.
Anyone already use it in practice. I am not sure about its stability for I got the code from the cvs head, not the stable version.
Any advice and help will appreciated !
Best Regards,
Gary Li
The next stable release, to be released
any day now, will include realtime as far as I know.
It works well. It would be nice to have a “stable”
release with it included so you do not pick up other bugs from CVS Head.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hi,
I'm very interested to understand that Warning too..does it happen
every 30 minutes??
g
Arne Morten Johansen wrote:
I often get a Maximum retries error while making outgoing calls. Why
does this happend? Most of the time a reload solves the problem, but not
all the time? What to
Mauro Zanin wrote:
Try different entry in this parameter.
In Italy mobiles start with 3, while public services with 1 and normal user
numbers with 0.
Using pridialplan=none, every number different from 0 was resulting in
termination code 1, normally used for number never seen on the network.
Eric Wieling aka ManxPower wrote:
What about pridialplan=unknown ?
As noted before, it's not reaching the end number that's the problem
as what pridialplan would help with sending the correct numbers out.
Dialing to a non forwarded phone works, but if you forward that phone
to a mobile phone
This is an interesting document about VoIP and Echo.
http://www.cisco.com/univercd/cc/td/doc/cisintwk/intsolns/voipsol/ea_isd.htm
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There is no static interval. But i found out that it was my IP-Phone Service
Provider that was having serviceproblems today.
-Opprinnelig melding-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne av Giorgio Incantalupo
Sendt: 26. august 2005 11:33
Til: Asterisk Users Mailing
Hmmm. I am often surprised when I don't get a response to a post that I
think would interest at least _one_ person in the community. This one
surprised me a little more, since I offered some code ;-).
This morning, I just got a bounce notice that it was undelivered, which
might explain it,
Hello,
I am using asterisk as voicemail for my sip proxy.
When a user (1234)dials , the call is forwarded to
asterisk. However I receive the following error:
--Executing VoiceMailMain(SIP/1234-9afc, 1234) in
new stack
--Playing 'vm-password' (language 'en')
[WARNING]: app_voicemail.c:3359
I've the following setup :
sip phone - ser (auth and routing) - asterisk with capi isdn
when I call a pstn number everything works fine, but I can't hear
anything till the called answer.
this is the output from a test call :
-- Executing Playtones(SIP/2.7.184.61-08152880, dial) in new
We are in the process of an Asterisk call center deployment using IAX2
G711 ulaw softphones. Outbound sound quality is terrible.
This week we rebuilt the entire LAN with Cisco 2950-EI switches and have
employed QoS on the switches and router. Still sounds terrible.
What we are now finding
Hi,
The only thing i know is that you need a netbootserver
using five special files. So, if possible, ask Siemens
for the optipoint 600 netboot upgrade procedure. AFAIK
it is a known problem...
hope it helps...
--- Anthony Cox [EMAIL PROTECTED] wrote:
Not strictly a problem with Asterisk but
We just had * crash on us - no calls could be made / received. We had to
kill -9 the * process.
Checking the error logs, I came across these two lines, with the times
matching the crash:
Aug 26 13:48:00 WARNING[19282] pbx.c: Local/[EMAIL PROTECTED],2 already
has PBX structure??
Aug 26
Hi,
I have 2 CDs that would like to make a backup of , I am having a hard time doing. I have tried NERO ver.6 but it does not work it always report unrecoverable sector.
Does anyone knows of a copy tools to use to copy the CD
Any help will be very nice and appreciated. Thank you all...
Hi all
I'm trying to figure out what values are valid for the insecure option in a
realtime configuration table. The table field is 4 chars long and the actual
valid values for this is longer. Can I modify the field length or has this
changed? Below is where I looked, if I'm not looking in
Alessio Focardi ha scritto:
Hello Lars,
Have you got kernel sources installed ?
I think that are mandatory for Zaphfc.
Regards
Not only, you have to have the kernel config save file too.
Remember to make dep too.
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I cannot reach voip-info - is it just me or is the site not available ?
Julian
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Ellafi Fituri wrote:
*/Hi,/*
*/I have 2 CDs that would like to make a backup of , I am having
a hard time doing. I have tried NERO ver.6 but it does not work
it always report unrecoverable sector./*
*/Does anyone knows of a copy tools to use to copy the CD/*
*/Any
On Fri, Aug 26, 2005 at 06:42:33AM -0700, Ellafi Fituri wrote:
Hi,
I have 2 CDs that would like to make a backup of , I am having a hard
time doing. I have tried NERO ver.6 but it does not work it always
report unrecoverable sector.
Nero? what is it? I don't have it in my apt
Hi,
I'm not the OP, but I had a similar problem, in my case fxotune ran
successfully for just one out of 3x FXO modules, but the coefficients were
all 0's. My kernel is 2.6.11 on CentOS 4.1.
So I'm curious if 2.6 kernel (instead of 2.4) has any input in this whole
echo issue, not just
At present, I would recommend [EMAIL PROTECTED] It comes with some
monitoring tools as well as AMP.
Darren Wiebe
[EMAIL PROTECTED]
Zeeshan Zakaria wrote:
Hi all,
What are the best and free tools for remotely adding, removing users
on Asterisk server and also for monitoring the status of
Hi,
is there anybody who knows what this warning means??
WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type
TIA
Giorgio
--
GIORGIO INCANTALUPO
Tel. +39 02 9350 4780 (104)
FGA Software
20017 Rho - Via
On Fri, 26 Aug 2005, Simone Cittadini wrote:
I've the following setup :
sip phone - ser (auth and routing) - asterisk with capi isdn
when I call a pstn number everything works fine, but I can't hear anything
till the called answer.
If you want tones from isdn before the connection is
T.38 isn't a trivial enhancement, and I think that the community should
consider itself extremely fortunate if someone actually gets T.38
implemented (including DSPs) for as little as $5500 being the motivation.
True, but Steve Underwood does already has a lot of the DSP stuff done
already
I've been trying for 18 hours ... ;)
Julian
Giorgio Incantalupo wrote:
Hi,
sometimes it is not available.
Be patient, wait 10 minutes and try again.
Giorgio
Julian Lyndon-Smith wrote:
I cannot reach voip-info - is it just me or is the site not available ?
Julian
True, but Steve Underwood does already has a lot of the DSP stuff done
already with spandsp, doesn't he?
I sincerely apologize for that first sentence... wow. :-)
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I haven't had similiar experience, but in several threads about sound
quality people have talked about Network cards being the culprit. In
particular, a few people have commented all sorts of problems on
onboard NIC's, since they tend to be of lesser quality than
stand-alone NICS.
On 8/26/05,
Hi,
Im
experiencing a problem with playing back my voicemail. (Failed
to write frame). It has been indicated in the archives that this is problem
can be solved by updating asterisk from the cvs. I
did make update in the /usr/src//asterisk
directory to resolve this. However I got a
I have had no trouble reaching it.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: Friday, August 26, 2005 10:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] voip-info -
I am trying to configuring/running Asterisk::LDAP perl module getting
from http://projects.alkaloid.net/ but no luck i have successfully
installed this module but when i include its scheme file which is
asterisk.scheme in the LDAP include list and try to start the LDAP
Server service its gives the
I am trying to configuring/running Asterisk::LDAP perl module getting
from http://projects.alkaloid.net/ but no luck i have successfully
installed this module but when i include its scheme file which is
asterisk.scheme in the LDAP include list and try to start the LDAP
Server service its gives the
Anyone have a good
tool(s) to use for simulating a bunch of calls? Benchmarking or stress
testing?
I only need SIP
protocol, and do appreciate any replies...I realize I could google it, but I am
looking for opinions as well.
Sherwood
McGowan
A little off topic but for packet 8 users out there using
asterisk behind a x100p check out the new firmware
http://www.dslreports.com/r0/download/872826~6e5c593b26b72aef4bf68f6710eed5b8/sip1315unl.zip
Allows you to assign your own codecs, currently using g711
90kbs and sounds
ARTCP (not yet released) will be doing exactly this, along with Zabbix for
monitoring (custom UserParameters will be included in ARTCP) for *REALTIME.
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Darren Wiebe
-Sent: Friday, August 26, 2005 10:23
Giorgio Incantalupo wrote:
Hi,
is there anybody who knows what this warning means??
WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type
type = friend
type = user
type = peer
TIA
Giorgio
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Ok, dont flame me, I know this is a question with an
obvious answer to some, but I am not one of them.
Installed FC3, but this time I decide to update since my
ISOs are a bit old, so typical yum update
Downloaded the FC3 SRPM for my kernel 2.6.12
Installed the SRPM package
Ran
Hi All:
We have a very old Aspect ACD in our call center, I am doing research to
replace it by using Asterisk, my boss has some kind of questions about
capability and reliability of Asterisk, does anyone have done this kind of work
with good result? I need some examples to convince him.
Short answer: Yes. It's just data.
Long answer:
In your LAN:
Usually depends on the nature of the other data on your LAN. If you LAN
has a ton of traffic you will have to use something like QoS tagging to
ensure that your voice traffic is prioritized. Any decent switch supports
this tagging and
This could be a duplicate post, sent it originally 4 hours
ago, it never showed up!
I know this is a question with an obvious answer to some,
but I am not one of them.
Installed FC3, but this time I decide to update since my
ISOs are a bit old, so typical yum update
Downloaded the
I've been thinking about how one would accomplish the same thing.
I've got a CTI enabled GUI that tells the agent that they're logged in
with the call centers that I've deployed thus far, but it's not quite
the same as the agent just being able to look at the phone as well and
know that they're
Matt Schulte wrote:
1) You have to do a factory reset, or wipe out the line config.
2) By default it dials ext 8500 I believe.
3) You *should* be able to change _name, I can't remember the effect
that has since you already have authname in.
Matt
-Original Message-
From:
Greetings all
Grandstream released a new firmware and it seems like the
speaker phone problem has been fixed. However we updated to firmware 1.0.1.12 to fix the echo problem but found other problems were
now created. The worst of these new problems is that the whole phone starts degrading, the
On Aug 24, 2005, at 11:21 AM, Doug Lytle wrote:[EMAIL PROTECTED] wrote: Hello,But does not go into how to mark a user. voip-info archives, and google didn't lead me to any clue, anddigging to app_meetme.c wasn't fruitful.Anyone have an example on how they marked a user in their dialplan? Create
Billy wrote:
`insecure` varchar(4) default NULL,
This can be changed. I just read the chan_sip.c code and the following
values are acceptable:
very
yes
true
basically anything with true/false value
port
invite
port,invite
Everywhere it is RFC2833 including in SIP phone, Asterisk's sip.conf.
DTMF work only from the phone that is hooked with asterisk box.
Thanks,
-Original Message-
From: [EMAIL PROTECTED]
Sent: Wed, 24 Aug 2005 12:04:04 -0400
To: asterisk-users@lists.digium.com
Subject: Re:
Hello everyone, I had an asterisk box which was
working great bt now for some reason I cannot dial out
on any of my outside lines. I am using a TDM card
with 4 FXO ports.
System:
Debian Sarge, 2.6.8-2-386
Compaq Proliant ML370 G2 Server
Polycom IP500 Phones
dtmfmode=rfc2833
I initially set up
Anyone have a good
tool(s) to use for simulating a bunch of calls? Benchmarking or stress
testing?
I only need SIP
protocol, and do appreciate any replies...I realize I could google it, but I am
looking for opinions as well.
Sherwood
McGowan
Sherwood McGowan wrote:
Anyone have a good tool(s) to use for simulating a bunch of calls?
Benchmarking or stress testing?
I only need SIP protocol, and do appreciate any replies...I realize I
could google it, but I am looking for opinions as well.
There is SIPp:
Greetings Everyone!
The version 1.0 of the PhoneCALL Administrative Manual has been released.
It is more of an outline of the features and interface, and we'll be
adding lots of more detailed information in the manual over the next few
days/weeks.
Of course, we'd love to get your input on
HooDaHek, the caller ID and instant messaging notification service for
Asterisk boxen, is now updated to version 0.4.
Information/download here:
http://www.nathanpralle.com/software/hoodahek.html
Changes:
- Changed the AIM bot to use Net::OSCAR instead of Net::AIMTOC since AOL
managed to
Stijn Jonker wrote:
Hello Tom,
On 26-Aug-2005 7:50, Tom wrote:
Most of the time i can find answers to my questions on the wiki, google,
or searching the list now i am stuck .
I have a small * box at my house running 1.0.9 stable and a devlite kit.
Every thing is awesome VM, IVR, Echo
I have had similar experience with an Intel NIC that had DELL's name on
it vs a 3COM 3C905b.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Logan
Sent: Friday, August 26, 2005 10:42 AM
To: Asterisk Users Mailing List - Non-Commercial
Hi Pat,
I would check the DTMF settings on your phone - I had a similar problem
until I switched to RFC from Inband.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Feed: http://www.dnv.org/rss.asp
On Aug 26, 2005, at 4:56 AM, pat
Hi -
I'm running CVS-HEAD from 2005-08-11 20:17:17 UTC, and I'm trying to
set up some redundancy on IAX connections between locations. I have
two IAX peers set up that work correctly by themselves: ast551-out
and ast551-out-backup:
[ast551-out]
type=peer
secret=secret
username=ast551
I am looking at
alternatives to the Digium TDM04B. The only one I can find is the
Voicetronix openline4 but I cannot find a lot of information on
it.
Does anyone have any
experience with it on Asterisk that they can compare to a Digium TDM04B. I
am particularly interested in the built
So bottom line please.
Have we decided that it is STILL correct to set RX/TX gain for 14800 with
ztmonitor quantitative using a telco 1004hz 0dbm test phone number? If not,
what should we set it to with ztmonitor.
-Original Message-
From: Rich Adamson [mailto:[EMAIL PROTECTED]
Sent:
What was the issue with zaptel and 2.6.12?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lee Archer
Sent: Thursday, August 25, 2005
1:22 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
fedora core 3 kernel source -
I would like to know if any body is using the Polycom Soundstation IP
4000 SIP conference phone with Asterisk. I am thinking of purchasing
one.
Kurt
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I have one and it is absolutely awesome. Works great and the quality of
Polycom conference phones is excellent regardless of protocol.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kurt x
Sent: Friday, August 26, 2005 9:50 AM
To: Asterisk
Hello List!!!
Does anyone know where AstTAPI stores it's configuration information?
We have a domain and our users are 'domain users'. When they change the TAPI
registration information in AstTAPI it does not stick. Certainly, I don't
have to give every admin permission on their box for this?
Hi,
I noticed the .WAV file for voicemails is what gets e-mailed to people
when someone leaves a voicemail. I also noticed today that I can not
play the .WAV files on my macintosh or linux machines. I *can* play
the .WAV files on my Windows machines. I can play the .wav files on
either
Hi,
On several occasions one or more of our grandstream Handy tone 486 ATA
would crash. If for some reason that ATA is not rebooted immediately,
asterisk would not disconnect the call, even though the party on the other
end of the call have already hung up the call. The call would continue
Hi Kurt -
I would like to know if any body is using the Polycom Soundstation IP
4000 SIP conference phone with Asterisk. I am thinking of purchasing
one.
Yes, we have one, and we have the add-on pod mics for it, too. The
setup works well, for the most part. The mics aren't quite as
We are using Plantronics H51N headset top with DA55 USB adapter which
has DSP built-in. Terrible means garbled, unintelligible,
underwater-sounding.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philipp
von Klitzing
Sent: Friday, August 26, 2005 11:23
I am about to build a Dual Opteron Asterisk box as our soon to be
production server.
Is Core 4 supported or should I stay with Core 3?
There was a recent post about an issue with the latest Core 3 Kernel and
zaptel. I had the same experience, but just rolled back to the previous
version of the
Try format=wav|gsm instead of format=wav49|gsm in your voicemail.conf
Be advised, however, that the attached files will be considerably
larger - we made this change to increase the volume of attached
messages, and can live with the increased file size.
I use a Mac, and the files play just
Hey, all. I have the following, and
ignorepat = 9
; Testing - access to telco1/FXO
; XXX
exten = _9.,1,Dial(SIP/outboundfxo/${EXTEN:1},20)
exten = _9.,2,Hangup
Unfortunately, once I hit 9 on a connected phone, I do *not* get a dial
tone back.
Can someone suggest what I might have done wrong?
Hello,
In my quest to figure out the source of the random echo on our shiny new
asterisk install, I have been using ztmonitor on the TDM400p channels
for the good part of today.
I have 2 TDM400p cards with 8 FXO modules and 6 pots lines connected to
them (last 2 channels are unused but configured
Take it from someone who owns 25 of them. Stay away from FC anything.
Use CentOS 4 its better more stable and has true multi-treading as FC
doesn't thread anything..
..o---o.
Brian Fertig
NOC/Network Engineer
Planet Telecom, Inc.
Tampa, FL
Mason Loring Bliss wrote:
Hey, all. I have the following, and
ignorepat = 9
; Testing - access to telco1/FXO
; XXX
exten = _9.,1,Dial(SIP/outboundfxo/${EXTEN:1},20)
exten = _9.,2,Hangup
Unfortunately, once I hit 9 on a connected phone, I do *not* get a dial
tone back.
ignorepat does not
VaibhaV Sharma wrote:
Hello,
In my quest to figure out the source of the random echo on our shiny new
asterisk install, I have been using ztmonitor on the TDM400p channels
for the good part of today.
I have 2 TDM400p cards with 8 FXO modules and 6 pots lines connected to
them (last 2 channels
On 8/25/05, Lance Grover [EMAIL PROTECTED] wrote:
I have now tried the lattest zaptel drivers for the 4 port tdm card
(wctdm) and it still cuts off after the 10 - 15 seconds.
Any Ideas?
I now found the issue, in the extensions.conf file I had the variable
TRUNK=Zap/g2c to check the channel
Thanks for the suggestions everyone!
The thing is, when I run tcpdump, this phone never really requests anything
as far as I can see.
The IP-address serving is not handled by the Asterisk box, which is on its
dedicated IP-address, but by a consumer type SMC Barricade 4-port router.
There's no
tourn of AGC , and mybe use GSM.
for usb device that use iax2 prtocol there are this one that have nice
http://www.gedameurope.com/us/002servizi_e_prodotti%5Bus%5D.htm
this usb device doe not need external sound card.
Philipp von Klitzing wrote:
Hi!
We are in the process of an Asterisk
Anybody using voicetronix cards? The 12 ports for example? What has been
your experience and how many cards can be put into one server?
Do they have the same IRQ problems as Digium ones?
AK
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With the deadline coming up for sending notices to customers, I found
it curious that out of 4-5 different providers I use, to date only one
of them has contacted me. The rest don't even have anything on their
website that I could find. Junction Networks was the only one that
actually sent me a
An extension of 30 days has been granted. Just like the HDTV broadcast
requirement deadlines the FCC cooked up I'd predict there will be a few
more extensions before the fight is over.
http://tinyurl.com/a8tj8 -- Reuters article
Am I missing something here? Is the FCC going to be extending
On Fri, 2005-08-26 at 12:37 -0500, Eric Wieling aka ManxPower wrote:
VaibhaV Sharma wrote:
Hello,
I have 2 TDM400p cards with 8 FXO modules and 6 pots lines connected to
them (last 2 channels are unused but configured in zaptel). Even when
the lines are onhook, the Tx values settle down
Hello Asterisk friends,
Does somebody know few french phone numbers to do telco 1004Hz 0dBm signal
tests phone ?
Thanks in advance.
Best Regards,
Francois BERGERET,
Happy French Asterisk user :-)
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Title: PCI 2.3
Hello All,
Anyone know if this is backwards compatible with 2.2?
Here is the spec from the Mobo I am looking at.
Five 32-bit v2.3 Master PCI bus slots (support 3.3V/5V PCI bus interface).
Thanks!
Wley
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I just upgraded my asterisk install from january 2005 CVS HEAD to current CVS
HEAD :
* zaptel
* libpri
* asterisk
My asterisk have one TE405P with one span (the clock source) pluged into a
telco PRI E1, a second span is a PRI E1 to another PBX and the third one is a
T1
Anyone played with the possibility of modifying how a voice
sounds on asterisk?
Eg make an outbound call from asterisk but by pressing *1
your voice goes into a higher pitch etc?
Just a thought,
Cheers,
Dean
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I'm not the OP, but I had a similar problem, in my case fxotune ran
successfully for just one out of 3x FXO modules, but the coefficients were
all 0's. My kernel is 2.6.11 on CentOS 4.1.
So I'm curious if 2.6 kernel (instead of 2.4) has any input in this whole
echo issue, not just
Bottom line... ztmonitor can be used to 'assist' in setting some starting
values, but the further your asterisk box is from the central office, the
more likely the gain values will have to be adjusted lower then what you
want, and may very well appear off-scale with ztmonitor.
Given the curent
If I fire up asterisk it connects to my MSSQL server via ODBC just
fine. However, if I issue a reload it unloads the ODBC.. then loads
it again and I get an error... and keep getting it until I
On a fresh start:
Aug 26 15:43:14 WARNING[13818] cdr_odbc.c: cdr_odbc: table not
specified. Assuming
Andres,
Thanks for the suggestion. I did try it but it is not moving to the
next priority after the Dial command. I also do know for a fact that
it is not actually being answered. On the console I just get:
-- Called g1/123456789
On 8/26/05, Andres [EMAIL PROTECTED] wrote:
Eric Bishop wrote:
Already am using this option.
On 8/25/05, Jens von Bülow [EMAIL PROTECTED] wrote:
Hi Eric,
Don't you need to use out-of-band PRI signaling...
From /etc/asterisk/zapata.conf
snip
; PRI Out of band indications.
; Enable this to report Busy and Congestion on a PRI using out-of-band
;
ignorepat only works for analong phones connected to FXS modules.
Steve Maroney
On Fri, 26 Aug 2005, Mason Loring Bliss wrote:
Hey, all. I have the following, and
ignorepat = 9
; Testing - access to telco1/FXO
; XXX
exten = _9.,1,Dial(SIP/outboundfxo/${EXTEN:1},20)
exten =
I' m using Asterisk 1.09 on an virtual pc (VMWare 4.5) for testing. I can
make calls from a Softphone to softphone, Hardphone to Softphone and so on.
I can hear both RTP Streams. But when I call prompst on Asterisk I can hear
nothing. RTP Stream goning from Phone to Asterisk but not the other way.
We just replaced our old system at work with Asterisk. We use a TDM
card with 3 FXO ports. In tuning my gains, I discovered that the
on-hook rx levels do in fact waver a little bit. Line 1 is right around
145, Line 2 is around 235, Line 3 is around 380, and Line 4, which
doesn't go to a POTS
Bjorn,
Save yourself the trouble with the console cable. Take a spare PC,
install [EMAIL PROTECTED], setup the included DHCP server, install the
appropriate files in the /tftpboot directory, and plug the PC and the
phone into the same switch.
Voila!
Tom
On Aug 26, 2005, at 12:48 PM,
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