A more proper way to fix this:
; this takes care of them pesky telemarketers
exten = s/,1,goto(nocid,1)
exten = nocid,1,Answer
exten = nocid,2,Macro(record-on|${MYHOMEPHONE}|ANONYMOUS)
exten = nocid,3,Zapateller(answer|nocallerid)
exten = nocid,4,PrivacyManager
exten = nocid,5,Goto(home,3)
exten
On Thu, 2003-12-04 at 18:43, Miguel Cavazos wrote:
Hello guys, i have been on this mailing list for some weeks now, and i
was wondering if someone here has installed linux on the XBOX and use it
as a dedicated server. Its a 200 USD computer and could make it perfect
to asterisk, its little and
i have this very same problem but i have a different context
; Answering incoming calls
[incoming]
include = asterisk
exten = s,1,Wait,15
exten = s,2,Answer
exten = s,3,Wait,1
exten = s,4,Background(${SOUNDS}/casa)
exten = 1,1,Dial(SIP/101)
exten = 2,1,Dial(SIP/152)
here my problem is they can
On Fri, 2003-12-05 at 07:13, Steven Critchfield wrote:
During Phreaknic, Mark was showing off a Xbox running asterisk with 4
S100U interfaces connected to the game ports on the front. It was
interesting. In the end, I don't think it is cost effective as a real PC
since you can also build a PC
On Thu, Dec 04, 2003 at 10:35:13PM -0800, Andrew Gillham wrote:
Well as far as I can tell, the only version I have on the box is 2.4.22-1.
I certainly only have 'kernel-headers-2.4.22-1' installed and 'linux'
symlinked
to that directory in /usr/src.
i have not gotten the zaptel drivers to
On Thu, Dec 04, 2003 at 10:34:02PM -, Linus Surguy wrote:
I don't want to criticize your idea, but you do have to consider certain
points. Starting from (as has already been mentioned) the bandwidth of
DS3
is far too much to reasonably shove down the PCI bus without data loss /
On Thu, Dec 04, 2003 at 02:53:43PM -0600, Steven Critchfield wrote:
On Thu, 2003-12-04 at 14:06, John Todd wrote:
Obviously, there are no DS3 TDM cards that are currently compatible
with Zap channels. (or are there?)
Does anyone know of an inexpensive DS3 card that could perhaps be
I am uncertain of PCI bus speed limits - too many conflicting reports
are wedged into my head.
However, the intent here is to dump calls out via VoIP and not simply
switch between channels elsewhere on the DS3, so overcoming that
limitation needs to be addressed (if it exists at all, as a
Hi,
I have a FAX machine connected to a TDM400 card FXS port.
When I receive a fax call through X100 and transfer it to that extension,
the FAX machine display REC, but nothing happen (no fax received).
There is something special to be done for this configuration?
Thanks,
Dan
No, but I wouldn't expect it to work because of the echo issues with the
X100P cards.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Dan
Sent: Friday, December 05, 2003 4:27 AM
To: Asterisk Users
Subject: [Asterisk-Users] FAX
I actually have a small echo problem on my x100p that I haven't gotten
around to fixing yet, however my FAX works fine, even at 14.4k.
Unless you have a huge echo problem, it should be negligable.
-Pat
- Original Message -
From: Andrew Joakimsen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Hi all,
A new version (0.9.6) of DIAX is available for download at:
http://www.laser.com/dante or
http://www.geocities.com/tdanro
There are no new functions, but some bugs fixed:
What's new in version 0.9.6:
- add Default_user locales as new language. The program language can be
automatically
Hi,
- Original Message -
From: Patrick Cantwell [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 11:44 AM
Subject: Re: [Asterisk-Users] FAX connected to a TDM400 card port
I actually have a small echo problem on my x100p that I haven't gotten
around to fixing
Well, do a small script (perhaps in php) that will place 2 call files in
/var/spool/asterisk/outgoing
http://www.voip-info.org/wiki-Asterisk+auto-dial+out
In the call file, you can specify the context/extensions/... ,
For example, you can define the same extension that the Meetme one in
your
Miguel Cavazos wrote:
On Fri, 2003-12-05 at 07:13, Steven Critchfield wrote:
During Phreaknic, Mark was showing off a Xbox running asterisk with 4
S100U interfaces connected to the game ports on the front. It was
interesting. In the end, I don't think it is cost effective as a real
PC since
Skuse, Phil wrote:
Thanks for the reply.
After a lot of digging in the oh323 code, I've discovered that if the
callerid is a valid E164 (ie. entirely composed of digits 0123456789*#) then
the callerid is put into the Calling Party Number field, otherwise the
callerid gets put into the Display
Hi there,
feel free to drop me a line if anyone out there is interested in a
simplistic CDR import/analysis Db for Lotus Notes (import via myODBC).
However, if you have more than, say, 20.000 CDR records per day then this
approch is certainly not for you. :-)
Cheers, Philipp
On Wed, 2003-12-03 at 18:09, Darren McIntosh wrote:
hostname 192.168.65.200
[192.168.65.200]
host = 192.168.65.200
I seem to recall a similar issue with a different IAD. Try changing the
hostname and endpoint name to something else (like cisco2430)
Ill try that and report.
Hi,
I use Iconnect and is working fine.
www.deltathree.com
Best regards,
Chris HARIGA
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, December 05, 2003 1:40 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users]
On Friday 05 December 2003 04:27 am, Dan wrote:
Hi,
I have a FAX machine connected to a TDM400 card FXS port.
When I receive a fax call through X100 and transfer it to that extension,
the FAX machine display REC, but nothing happen (no fax received).
There is something special to be done
Thanks for the debug procedural steps. That's a lot of stuff to do. I think
I will attempt to do what you suggest on one of my servers and just try some
streamlining and reduction of functionality in the other server and see if
that fixes it. Since it can take upto 2 weeks for a server to freeze
If this is going to be more of a mainstay installation, I would highly
recommend that you get a T100P card and channelbank. They work like
champs and I've had virtually no complaints from any of those installs
I've done.
Thats actually what I'm using already. I had some issues with the X100P
Hi,
- Original Message -
From: marrandy [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 2:50 PM
Subject: Re: [Asterisk-Users] FAX connected to a TDM400 card port
You transfer it manually ???
Why ?
Because it is an operator who decide when and if to start de
Hello
I have couple of Grandstream phone and some of them
after a day or two
just stops receiving calls, you can still make a
call from that phone but you
cannot receive calls until you restart the
phone.
Is it a wrong configuration of phone or Asterisk
?
Thanks for any advices.
bart
- Original Message -
From: Carl Youngblood [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 1:35 AM
Subject: [Asterisk-Users] Internet-to-phone gateway?
Okay, I'm an asterisk newbie, so forgive me if this is really obvious,
but I'm wondering if there are services
I have a similar setup and have found that faxing, sending or receiving
works at best 50% of the time. I finally hooked the fax machine to the
second RJ-11 jack of the x100p and set it to listen for fax tones and to
grab the line from asterisk if it hears a fax. (Actually I may have a Y
adapter
On Fri, 2003-12-05 at 11:17, Senad Jordanovic wrote:
I like you idea. Very Cool :)
Is RAM upgradable on xbox?
Thanks
no its not, BUT its very optimize the xbox hardware should work REALLY
REALLY GOOD i dont know how good it will be on long uptimes
Miguel Cavazos
Do you type reload at the cli a few times a day?
If so try not reloading Asterisk and I'll bet Asterisk
stop blocking. If you don't normally reload the box
you will need to trouble shot normally.
mattf wrote:
Hello,
I have had several instances over the last month of Asterisk freezing,
Hi,
I havea setup using Xlite SIP phone and X100P card and I am using the Dec 1 CVS. I have put the incomming limit in both zapata.conf for the FXO card and in the sip.conf for the user. Incomming limit works fine on FXO, but on Xlite SIP phone the second line starts blinking and i can hear the
Hi,
- Original Message -
From: John Vozza [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 3:27 PM
Subject: Re: [Asterisk-Users] FAX connected to a TDM400 card port
I have a similar setup and have found that faxing, sending or receiving
works at best 50% of the
We have an installation with 9 inbound voice channels (one is the
fax) and 768K data. It is a Hybrid PRI. It terminates into a
T100P. It is working great! The cost was better than the
POTS plus data.
Can I ask what Telephone/Internet service provider you are
getting this
|
|Hello
|
|I have couple of Grandstream phone and some of them after a day or two
|just stops receiving calls, you can still make a call from that phone but you
|cannot receive calls until you restart the phone.
|Is it a wrong configuration of phone or Asterisk ?
|Thanks for any advices.
|
|bart
I have the same problem but when the phone is up for about one day.
I have also conected Cisco phones to asterisk and I do not have such a
probelm with them.
Is it something to do with configuration of grandstream phones?
Bart
- Original Message -
From: Eris Riswanto [EMAIL PROTECTED]
I do extensions reload maybe once a day, does that have the same effect?
Can AGI perl scripts cause the locking as I have described it?
Anyone have any experience with that?
Thanks,
MATT---
-Original Message-
From: James Sizemore [mailto:[EMAIL PROTECTED]
Sent: Friday, December 05,
Nick Bachmann wrote:
Hello
I have couple of Grandstream phone and some of them after a day or
two just stops receiving calls, you can still make a call from that
phone but you cannot receive calls until you restart the phone. Is
it a wrong configuration of phone or Asterisk ? Thanks for any
Ed Rubright wrote:
The company I work for has deployed an Avaya IP phone system. They have
deployed the Avaya 4602 and 4620 IP telephones. They might be sending
me one of these phones for use in my home office.
I recently (April) left Avaya. I was in the IP router/GigE switch area
mostly,
Jonathan Moore wrote:
Quoting Nick Bachmann [EMAIL PROTECTED]:
Having a DSS (the blinking lights for each extension, short for
Digital Station Selector) is a feature that I wish Asterisk had. A
week or so ago there was discussion about a new Windows-based Asterisk
application (Asterisk
Please try my Asterisk search engine at:
http://search.voip-forum.com
for searching the list.
Any comments to me off list.
I'm indexing lists.digium.com voip-info.org and iptel.org
/Olle
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Friday 05 December 2003 13:45, Azher Amin wrote:
Hi,
I have a setup using Xlite SIP phone and X100P card and I am using the Dec
1 CVS. I have put the incomming limit in both zapata.conf for the FXO card
and in the sip.conf for the user. Incomming limit works fine on FXO, but
on Xlite SIP
This may not be the preferred mode of operation, but I read in an earlier
post that this is caused by a bug in the * sip stack which causes the
phone to lose registration during a qualify action from *. If you turn off
qualify the phone doesn't do this anymore and becomes quite stable.
On Fri, 5
I'm looking for a solution that allows me to queue up many outbound
voice messages that will be sent to our customers, not a solution for
making personal phone calls. Sorry for not being specific enough.
On Dec 5, 2003, at 5:19 AM, Chris HARIGA wrote:
Hi,
I use Iconnect and is working fine.
Please could you tell me how to turn off qualify ?
- Original Message -
From: Jonathan Moore [EMAIL PROTECTED]
To: ASTERISK USERS [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 11:54 AM
Subject: Re: [Asterisk-Users] grandstream budgeTone phones or Asterisk ??
This may not be the
I already lower it to one minute a long time ago but it did not fix my
problem.
- Original Message -
From: Nick Bachmann [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 11:27 AM
Subject: Re: [Asterisk-Users] grandstream budgeTone phones or Asterisk ??
Hello
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steven Critchfield
Sent: Thursday, December 04, 2003 4:57 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Operating environment for *
3) How do you handle crashes (murphy -will- visit you
Symptom: Phone after about 15mins
will stop functioning
Problem: DHCP lease renewed but default route
dropped
Fix: Assign a static ip and problem
is resolved. Upgrade to new firmware once it is
released
It turn's out thatthese phones have a few
issue in 1.0.3.81 firmware. Thephone may
Picture this though - a rack of XBOX's running as SIP/IAX/IAX2 application
servers and some sweet one RU Dell's as POTS servers ... hmm
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of costas
Sent: Saturday, 6 December 2003 2:18 AM
To: [EMAIL
At 07:17 AM 12/5/2003, you wrote:
I guess for the XBox you would need some external gateway. Audicodes or
Mediatrix come to mind but they start at $500.
A year ago, I installed Linux on Playstation 2. I had to purchase it with
the hardware for about $200. (40GB, keyboard and some network
On Fri, Dec 05, 2003 at 10:42:02AM -0500, Glenn Dalgliesh wrote:
Symptom: Phone after about 15mins will stop functioning
Problem: DHCP lease renewed but default route dropped
Fix: Assign a static ip and problem is resolved. Upgrade to new firmware once it is
released
It turn's out that
Can you imagine trying to sell this to a customer though? The customer
see's you walk in the door w/ an X-Box or PS2 and they say That's our phone
system ?!? Not that I would think someone would do it, just a thought that
entered my mind.
Joe
-Original Message-
From: [EMAIL
I am also looking for a NAT-Friendly tftp server too. Let me know if you
find one please.
Thanks,
Andres
- Original Message -
From: Nicolas Bougues [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 11:18 AM
Subject: Re: [Asterisk-Users] GrandStream Budgetone Phone
My understanding is that the ascend gear only speaks IPDC and not MGCP,
so not sure it would even work with asterisk.
On Thu, 2003-12-04 at 15:09, Steve Dolloff wrote:
I would be seriously wary of putting a DS3's worth of voice traffic on a
TNT. I don't believe they are rated to handle that
I have been mulling over what it would take to get drivers done for
ImageStream's products. They have a component architecture that
is supposed
to reduce development time/cost. The component stuff is open
source. The
part of the driver that you have to write can be open source or
On Fri, 5 Dec 2003, Nicolas Bougues wrote:
On a slightly different topic : does somebody know of a NAT-friendly
(as Grandstream means it) tftpd server ? It seems theirs replies from
port 69, which is the only thing their phones will accept.
[ If anybody wants it, I can send the 1.0.4.17
I would love to try it out too!
- Original Message -
From: Senad Jordanovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 1:38 PM
Subject: RE: [Asterisk-Users] GrandStream Budgetone Phone DHCP General
Observations
Nicolas Bougues wrote:
On Fri, Dec 05,
FYI you can't get back to the old firmware in some cases apparently.
- Original Message -
From: Bartosz Jozwiak [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 11:53 AM
Subject: Re: [Asterisk-Users] GrandStream Budgetone Phone DHCP General
Observations
I
At 10:42 AM -0600 12/5/03, Andy Hester wrote:
I have been mulling over what it would take to get drivers done for
ImageStream's products. They have a component architecture that
is supposed
to reduce development time/cost. The component stuff is open
source. The
part of the driver that
On Thu, Dec 04, 2003 at 08:15:21PM -0500, Jim Flagg wrote:
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 04, 2003 7:54 PM
Subject: Re: [Asterisk-Users] Channelbank Recomendation and GS102 question
We have an
Hi
I wouldn't mind the 1.0.4.17 firmware.
Dave
mailto:[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Glenn Dalgliesh
Sent: 05 December 2003 17:13
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] GrandStream Budgetone Phone DHCP
Hi!
Do you type reload at the cli a few times a day?
If so try not reloading Asterisk and I'll bet Asterisk
stop blocking.
Recently I started running reload every 30 minues (to solve a IAX qualify
problem). Since then I do see problems that weren't there before, i.e.
when I issue reload
about the hardware the xbox has HD 8 GB, ethernet port and 4 usbs a
nvidia video card and all you need is the usb adapters and some hours
reading on how to break in without a mod chip
using an xbox its a cool idea because its nice and small and cheap!
using the S100U was a great idea for the
Hi,
I cannot get two Polycom 600 phones to bridge natively. My sip.conf has
canreinvite=yes for both phones. They connect, and I can talk as usual, but
sniffing shows the RTP stream is routed through Asterisk.
The exact spot where the attempt to natively bridge fails is in rtp.c, line
1281
I'm glad other people are seeing the same problem I've been seeing and
posted about a day or so ago. My * is running on rh9 with most recent
kernel with up2date.
Does someone figure this is a threading issue? Does it need to be debugged
using the method presented on the list yesterday?
Jeremy,
I talked to Imagestream this morning about the possibilites. Their lead
engineer said that there would be no way to do voice over their
DS-3 cards
using software processing because it would take too much
processing power.
It would be possible to do some custom design for their boards that
-Original Message-
From: TC [mailto:[EMAIL PROTECTED]
Sent: Thursday, December 04, 2003 11:58 PM
I have had several instances over the last month of Asterisk freezing,
Does anyone have any suggestions? or ideas as to what may be causing it?
Sounds like some type deadlock
Take a look here
Well, we know that we would be able to handle a partial DS3... assuming such a
thing is possible. Wouldn't people prefer a partial DS3 for say... 12T1's to
no way to do that many?
Why not just try to get the card working, then testing would show exactly how
much data could be handled...
Hi!
I'm glad other people are seeing the same problem I've been seeing and
posted about a day or so ago. My * is running on rh9 with most recent
kernel with up2date.
In my case: RH 7.2, kernel 2.4.20, reload through cron job each 30 min.
Reason:
using software processing because it would take too much
processing power.
He might mean the processing power of the controller on the card,
not the PC it's sitting in.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
On Fri, Dec 05, 2003 at 11:58:44AM -0600, Andy Hester wrote:
The guy did leave open the possibility that he could be wrong, and said that
he'd be glad to answer any further questions or if we had some other way of
doing it. If you or some of the others think that this should be possible
I have exactly that configuration, and it's working fine for me. I have the
following config which may or may not be relevant to fax working...
echocancelwhenbridged = no
callprogress = no
I've heard that there have been echo problems when ring and tip are reversed...
but mine havn't ever
Greg Boehnlein wrote:
On Thu, 4 Dec 2003, Bob Knight wrote:
Steve Dolloff wrote:
I would be seriously wary of putting a DS3's worth of voice traffic on a
TNT. I don't believe they are rated to handle that much voice. The
APX1000 would be a much better platform, but I don't know if you
Miguel Cavazos wrote:
about the hardware the xbox has HD 8 GB, ethernet port and 4 usbs a
nvidia video card and all you need is the usb adapters and some hours
reading on how to break in without a mod chip
using an xbox its a cool idea because its nice and small and cheap!
using the
There are a number of vendors for DS3 cards -- The issue when I last looked at these
cards ( 1yr
ago) was that many of them were not fully channelized -- i.e. the cards did not
support breaking the
data stream into indiv 64Kbps DS0s. At the time I was talking with the vendors, all
of them were
On Fri, 2003-12-05 at 04:43, Areski wrote:
Well, do a small script (perhaps in php) that will place 2 call files in
/var/spool/asterisk/outgoing
http://www.voip-info.org/wiki-Asterisk+auto-dial+out
In the call file, you can specify the context/extensions/... ,
For example, you can define
I have no reason to disbelieve this report, but I will offer some
minor scepticism at this reply. A well-equipped PC can currently
handle 8 T1 channels, and it seems that only the IRQ issue is causing
more channels to not be viable in the current TE410P environment. It
would seem
On Thursday 04 December 2003 14:06, John Todd wrote:
Obviously, there are no DS3 TDM cards that are currently compatible
with Zap channels. (or are there?)
This isn't so much a technological limit as much as a detail of
implementation, but the current Zaptel drivers have a limit of 252
On Fri, 2003-12-05 at 08:34, Jim Flagg wrote:
Original Message -
From: Ariel Batista [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 8:52 AM
Subject: Re: [Asterisk-Users] Channelbank Recomendation and GS102 question
We have an installation with 9
On Fri, 2003-12-05 at 10:18, Walker Haddock wrote:
On Thu, Dec 04, 2003 at 08:15:21PM -0500, Jim Flagg wrote:
- Original Message -
From: Walker Haddock [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 04, 2003 7:54 PM
Subject: Re: [Asterisk-Users] Channelbank
Imagestream is really not the company to look to for this kind of
solution. They are not really interested in selling anything other than
their complete routers from what i can tell.
Sangome will have a DS3 card out shortly I believe. It should have the
capability to work down to the DS0 level
I just got 2 IpDialog phones for use with my
Asterisk system. I have been able to get the phones to just dial local
extensions but it is not able to register with my system correctly. I
would like to know if someone has set these phones up before and how they did
it! Is there any examples
Has anyone out there had the freezing problem(where they have to kill
asterisk with kill -9) on any linux distro other than RedHat?
What other distros do people out there use with their production Asterisk
systems?
MATT---
-Original Message-
From: Philipp von Klitzing
[mailto:[EMAIL
On Friday 05 December 2003 14:44, mattf wrote:
Has anyone out there had the freezing problem(where they have to
kill asterisk with kill -9) on any linux distro other than RedHat?
What other distros do people out there use with their production
Asterisk systems?
We have no problem with
See below!
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 3:55 PM
Subject: Re: [Asterisk-Users] Asterisk freezing HELP
On Friday 05 December 2003 14:44, mattf wrote:
Has anyone out there had the freezing
For those that have had X100P echo problems, it seems that somewhere just
prior to Asterisk CVS-12/04/03-14:24:40 it has been fixed to where its hardly
detectable for just a couple of seconds at the start of a call. (I use to
have it rather bad for the first 15 seconds or so.)
I'm using
No problems with freezing: using Redhat 9 in production systems (2) (but
IVR only traffic in these), and Fedora in house for testing
-Scott
Scott M. Stingel
Emerging Voice Technology Inc.
Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
URL:www.evtmedia.com
This is a little off topic, but this is one of the greatest
concentrations of potential VoicePulse customers I know of. We signed
up with an account last night just to do some testing. We tested the
outbound functionality of calling through VoicePulse from * and it
worked just fine. We then
Running multiple boxes with RH7.3 and RH9.
They all freeze every few days and always when processing an inbound SIP
call. Sometimes it will freeze 5 times in a row when processng a SIP call
from the same user.
We have been unable to reproduce this so no bug has been opened.
The only suspicious
I guess all we'll need now is an * sticker to stick over the X, to make
a *box
[EMAIL PROTECTED]:~$ sudo whois pbxbox.org
NOTICE: Access to .ORG WHOIS information is provided to assist persons in
determining the contents of a domain name registration record in the PIR
registry database. The
Just for giggles did you use ztmonitor to adjust your rx/txgain?
bkw
On Fri, 5 Dec 2003, Rich Adamson wrote:
For those that have had X100P echo problems, it seems that somewhere just
prior to Asterisk CVS-12/04/03-14:24:40 it has been fixed to where its hardly
detectable for just a couple
Thanks for that
One question how do I stop * from picking up that line
But still allow it to dial
Regards Mick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony Wood
Sent: Thursday, 4 December 2003 11:25 AM
To: [EMAIL PROTECTED]
Subject:
The only suspicious thing is that they always seem to freeze when the SIP
channel is right around this line:
if (pkt-owner-owner) {
/* XXX Potential deadlocK?? XXX */
ast_queue_hangup(pkt-owner-owner, 0);
On Friday 05 December 2003 14:44, mattf wrote:
Has anyone out there had the freezing problem(where they have to
kill asterisk with kill -9) on any linux distro other than RedHat?
What other distros do people out there use with their production
Asterisk systems?
We have no problem
Nope, I'm going to try it for grins, but the levels are very acceptable
right now with 7960's primarily.
Just for giggles did you use ztmonitor to adjust your rx/txgain?
bkw
On Fri, 5 Dec 2003, Rich Adamson wrote:
For those that have had X100P echo problems,
Don't set it to answer the zap channel.
Joe
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, December 05, 2003 4:48 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Fax
Thanks for that
One question how do I
I know that Time Warner Telecom (www.twtelecom.com) also provides a
mixed data/voice burstable PRI T1 with their VersiPak service. I have
installed a number of both channelized/channel banked
VersiPaks and one
PRI versipak (though it was not with Asterisk, it's connected to an
InterTel
See below.
- Original Message -
From: Ariel Batista
To: Asterisk User List
Sent: Friday, December 05, 2003 3:14 PM
Subject: [Asterisk-Users] Help with setup IpDialog Sip Phones.
I got the phone to work with NetxUSA's help(Sean). Good people there. Works
great. I normaly don't reply to
Another scary thing I noticed this week is that the newer the CVS the more
frequently it happens:
CVS from 2003-10-01 once every few days
CVS from 2003-11-03 2 times a day to once every two days
CVS from 2003-12-03 1 to 4 times a day
All on otherwise identical systems. Has anything been added to
I sure will try. (I am working on it right now)
Thanks,
Andres.
- Original Message -
From: TC [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 05, 2003 4:47 PM
Subject: Re: [Asterisk-Users] Asterisk freezing HELP
The only suspicious thing is that they always seem
I have done this, but I haven't put the server in place yet... It
appears to run absolutely fabulous, with the exception that OSS/dsp is
noisy as all get-out. Alsa drivers tend to fix this problem, though.
Other than that, I can pass at least 5-10 calls through with no problem
whatsoever.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Senad Jordanovic
Sent: Friday, December 05, 2003 5:18 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] XBOX as and * Dedicated Server
Miguel Cavazos wrote:
On Fri, 2003-12-05
We're using an X100P card to connect to the SBC PSTN and have a SIP
client (X-Lite) configured. I'm using a headset (Plantronics 20) to
connect to my computer. When I first make a call to someone over the
PSTN I can really hear myself echo bad in my headset. As the call goes
on it gets
1 - 100 of 116 matches
Mail list logo
