I am hooked up with broadvoice and have been having no problems that are
major there voice mail system went on the blits for about 30 minutes
yesterday but that was about it.
what kind of problems you expierencing?
- Original Message -
From: Joel Gathercole [EMAIL PROTECTED]
To: [EMAIL
I want * to answer the phone when call comes-in.
I've enabled /dev/phone0 in phone.conf and created /dev/phone0 with a
command:
mknod /dev/phone0 c 100 0
Though, when I start * I get:
Parsing '/etc/asterisk/phone.conf': Found
Sep 12 00:18:42 WARNING[16384]: chan_phone.c:950 mkif: Unable to open
is it in ebook format at all?
I am a blind computer user and have no way of getting it scanned in to my
computer even if I were to purchase it.
thanks
hank
- Original Message -
From: Sys. Concept Inc. [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, September 11, 2004 10:08 PM
Sys. Concept Inc. wrote:
I want * to answer the phone when call comes-in.
I've enabled /dev/phone0 in phone.conf and created /dev/phone0 with a
command:
mknod /dev/phone0 c 100 0
Though, when I start * I get:
Parsing '/etc/asterisk/phone.conf': Found
Sep 12 00:18:42 WARNING[16384]:
Hi there,
I was looking for a GSM gateway, but I didn't find any prices...
Anybody knows how much one of these costs?
And, is it possible to use an amateur radio with asterisk?
Thanks,
Marconi.
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On Sun, 2004-09-12 at 00:56, Brian Capouch wrote:
Sys. Concept Inc. wrote:
I want * to answer the phone when call comes-in.
I've enabled /dev/phone0 in phone.conf and created /dev/phone0 with a
command:
mknod /dev/phone0 c 100 0
Though, when I start * I get:
Parsing
On Sun, 12 Sep 2004 04:18:34 -0300, Marconi Rivello
[EMAIL PROTECTED] wrote:
I was looking for a GSM gateway, but I didn't find any prices...
Anybody knows how much one of these costs?
As I mentioned in the other thread, the cheapest single channel GSM to
FXO gateway I have come accross was
I have set up asterisk with an ISDN card using i4l. When I place a call
from ISDN to a SIP client, there is about a one-second delay from a word
is spoken to it is heard at the other end. The funny thing, is that the
last second or so of each call is saved somewhere in the depths of
Hi
Il dom, 2004-09-12 alle 10:05, Thor Atle Rustad ha scritto:
I have set up asterisk with an ISDN card using i4l. When I place a call
from ISDN to a SIP client, there is about a one-second delay from a word
is spoken to it is heard at the other end. The funny thing, is that the
last
Nick -
Put
nationalprefix=0
internationalprefix=00
in your zapata.conf file!
Magic!
Rgds
Tim
Nick Barnes wrote:
Hi all,
I've been batting my head against a brick wall for the best part of the day
and still haven't got any further (apart from getting a big headache, that
is). I've searched the Wiki
Hi,
My linuxbox has 2 eth's, one with pppoe for dsl, and also i got an ip_gre
tunnel. At the time i run asterisk, even i got bindaddr=0.0.0.0, it does
not show any port open for sip (5060), if i change 0.0.0.0 for any ip, next
time i reload, it opens the specific ip, changing back to 0.0.0.0 and
On Sat, 11 Sep 2004, Clif Jones wrote:
I have a friend with a PRI coming into a modem bank that is receiving
56K modem calls and some ISDN data calls. He wants to dump his analog
office phone lines and use some of the capacity on the PRI. I have been
digging through the mail archives and
On Sun, 2004-09-12 at 00:43, Andrew Kohlsmith wrote:
On Saturday 11 September 2004 14:39, Brian Cuthie wrote:
Anybody know an easy way to adjust audio level of recordings made in
Asterisk (using the 'record' application)? I've noticed that recordings
using the wav format are about twice
Hi Patrick,
Nice to see you got this working.
I am a bit confused as to how this works. In particular I am trying to
understand, if I had say 10 active sip calls (I am particularly interested
in sip), how do I specify which one I one I want to listen into.
Idealy I would like to have the
David answered:
I too battled a similar problem with my TDM400p. I solved it
by putting the following in the channel descriptions in zapata.conf:
stripmsd=0
Clearly this is not the default which I think should be obvious...
Hmmm. This doesn't appear to make any difference. Looking at
Hello,
thanks for the answers!!! You mentionned to use the switch command. I
read about it in the WIKI, but I couldn't find enought information to
understand what it is actually doing. Can someone point me to the right
direction?
Marc
Steven Critchfield wrote:
On Sat, 2004-09-11 at 21:41, Marc
Hi.
I have succesfuly installed asterisk and after I added a x100p card, but
the system doesn't seem to know the card is there. This is what I've done:
compiled and installed zaptel, libpri and asterisk in that order using
make clean ; make install commands. also, make samples for asterisk.
On Sun, 2004-09-12 at 06:42, Marc Storck wrote:
Hello,
thanks for the answers!!! You mentionned to use the switch command. I
read about it in the WIKI, but I couldn't find enought information to
understand what it is actually doing. Can someone point me to the right
direction?
The
Make sure you have the bios updated to recognise the card. If the hardware
does not see it the OS wont either.
I just built Asterisk on an old single 450 PII Compaq Prolient 800 with an
x100p and 3 ata186's. It works just fine.
Hope this helps.
--john
- Original Message -
From: Rodolfo
is there any in-depth information available about the switch command???
Marc
Steven Critchfield wrote:
On Sun, 2004-09-12 at 06:42, Marc Storck wrote:
Hello,
thanks for the answers!!! You mentionned to use the switch command. I
read about it in the WIKI, but I couldn't find enought information to
hello;
i got this each time when trying to dialout
Sep 12 12:49:27 WARNING[114695]: chan_sip.c:590 __sip_xmit: sip_xmit
of 0x811f88c (len 407) to 198.65.166.131 returned -1: Invalid argument
sip.conf
register = 1747668417x:[EMAIL PROTECTED]/1747668417x
[sipphone]
type=peer
secret=
Do you use most of you PRI channel from the PSTN enough to feel that you
Asterisk box can handle the load? I guess that if the TE405P card
hairpins the channels that go out the other span it would take much of
the load off of the server. Anyone else know for sure if these
quad-span cards are
It might help if you tried to modprobe the right module. Try wcfxo instead.
wcfxo is for the x10xp cards
wcfxs is for the TDM cards irregardless of the modules installed.
And the use of the -v parameter on all these commands would be helpfull.
Lyle
- Original Message -
From: Rodolfo
Hi and thanks.
Is there any especial issue about x100p? I'm building the system in an
old 400 Celeron but it detects and recognize all the PCI cards
installed: modem, nic, and video. Is there any reason for not to
recognize the x100p?
RODOLFO
John Hill wrote:
Make sure you have the bios
And, is it possible to use an amateur radio with asterisk?
Yes.
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I've looked through the archives - and see questions similar to mine,
but no answers. What, if anything, can be done to get the incoming
Caller ID to be presented on the Budgetone's Caller ID display? In all
other respects the phone+Asterisk seem to be extremely happy with each
other.
Steven P. Donegan wrote:
I've looked through the archives - and see questions similar to mine,
but no answers. What, if anything, can be done to get the incoming
Caller ID to be presented on the Budgetone's Caller ID display? In all
other respects the phone+Asterisk seem to be extremely happy
The mainboard is not recognizing the x100p card. It is not showing the
card on the PCI devices listing after POST
Any ideas, please?
RODOLFO
Rodolfo Grave wrote:
Hi and thanks.
Is there any especial issue about x100p? I'm building the system in an
old 400 Celeron but it detects and
On Sun, 2004-09-12 at 09:41, Duane wrote:
Steven P. Donegan wrote:
I've looked through the archives - and see questions similar to mine,
but no answers. What, if anything, can be done to get the incoming
Caller ID to be presented on the Budgetone's Caller ID display? In all
other
It may not show up on the screen during post(it doesn't on mine).
What is your /etc/zaptel.conf? Try ztcfg -v after modprobing zaptel
wcfxo(wcfxs is for TDM cards only, wcfxs is for the x100 cards).
Lyle
- Original Message -
From: Rodolfo Grave [EMAIL PROTECTED]
To: Asterisk Users
Rodolfo Grave a écrit :
The mainboard is not recognizing the x100p card. It is not showing the
card on the PCI devices listing after POST
Any ideas, please?
Remove all unnecessary PCI card and try (eg only VGA X100P), to see if
you computer detect card:
1. if not, try this card in another
I gather from the lack of response that no one has had a similar problem or knows
how to troubleshoot the problem. The Ooh, voice format changed to 4 is a mystery
to me since everything I find with that message has a coder format where I have a 4.
David
David said:
I have somewhat miraculously
Eric Wieling wrote:
On Sun, 2004-09-12 at 09:41, Duane wrote:
Steven P. Donegan wrote:
I've looked through the archives - and see questions similar to mine,
but no answers. What, if anything, can be done to get the incoming
Caller ID to be presented on the Budgetone's Caller ID display?
Hello!
I am trying to install a TN405P on a P4-3GHz-HT machine running Debian
Sarge with kernel 2.4.27. When I start Asterisk in -c mode it always
shows
== D-Channel on span 1 up
== Restart on requested on entire span 1
== D-Channel on span 3 up
== D-Channel on span 2 up
== Restart on
Hello,
Primus's MGCP uses authentication by MAC address. If you look in the
config program @ 192.168.15.1 you will notice on one of the pages near
the bottom a number and an ip address.. thats the authentication.
Unfortunately I do not have a primus box in my office anymore, they were
sent
Steven,
On mine in the UK the sip.conf entries are like yours but without the
callerid= entry and my CS phones give me the received callerid fine.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven P.
Donegan
Sent: 12 September 2004
Hi,
This past week our account got disabled due to massive registering
requests (BV told me 1000's in a few minutes). It has happened 3 times.
In my logs I only can see a reason for one of them happening, which I
believe is a bug in *, (could not grab lock, retrying...) I noticed
that line
Try upgrading the firmware
Thank you,
Steve Maroney
On Sun, 12 Sep 2004, Steven P. Donegan wrote:
Eric Wieling wrote:
On Sun, 2004-09-12 at 09:41, Duane wrote:
Steven P. Donegan wrote:
I've looked through the archives - and see questions similar to mine,
but no answers. What, if
Just pulled the callerid line out, restarted asterisk and gave it a shot
- no joy - GS display says 1000 (the extension) not my caller ID - I'm
sure this is something silly on my part - but haven't been able to spot
it yet...
David J Carter wrote:
Steven,
On mine in the UK the sip.conf entries
Firmware now current (1.0.5.11) - no change in what is displayed on the
phone. Good thought though :-)
Steve Maroney wrote:
Try upgrading the firmware
Thank you,
Steve Maroney
On Sun, 12 Sep 2004, Steven P. Donegan wrote:
Eric Wieling wrote:
On Sun, 2004-09-12 at 09:41, Duane wrote:
It's normal, in fact I use it to be sure that everything's ok, since I think
it will not occur unless we have no alarms on the spans!
Regards
Scott
Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California London England
www.evtmedia.com
-Original Message-
On Sun, 2004-09-12 at 09:56 -0700, Steven P. Donegan wrote:
sip.conf configlet:
[1000]
type=friend
username=1000
fromuser=1000
callerid=Computer Room 1000
Just a thought I've had funnies before because * is picky about the
syntax of the caller ID.
callerid=Computer Room 1000
is how I
Im not sure if you mentioned this or not, but what is your call
originatiing from ? On an FXO port ? Make sure your zapata.conf file is
correct. There are a lot of callerid options in there.
Thank you,
Steve Maroney
On Sun, 12 Sep 2004, Steven P. Donegan wrote:
Firmware now current (1.0.5.11)
Umar,
I am a bit confused as to how this works. In particular I am trying to
understand, if I had say 10 active sip calls (I am particularly interested
in sip), how do I specify which one I one I want to listen into.
I haven't had enough chance to test ChanSpy to know what happens when
According to the docs, these settings apply to certain switches like isdn which may
explain why it works for Nick. Why is it necessary to use the stripmsd=0 for my
tdm400 to dial out properly? Otherwise, every first digit is stripped whether its a
0, 1, 7 or whatever on local and long distance
Hi.
I have installed a x100p (THE x100p for those who have seen my former
post). Now I just want to connect a normal phone (not an IP phone) to
the card and use it as a sip extension (I have a FWD account)... more
clearly:
I want to be able to pick up the phone and call any FWD user using my
The X100P is for connecting to phone LINES (telco lines) not for
connecting to phones.
FXO = expects to RECEIVE dialtone and ring voltage
FXS = expects to PROVIDE dialtone and ring voltage.
On Sun, 2004-09-12 at 12:59, Rodolfo Grave wrote:
Hi.
I have installed a x100p (THE x100p for those
Chris Icide [EMAIL PROTECTED] writes:
Satellite links can be pretty tough to troubleshoot. It sounds like
you are running into a uplink buffer issue. On heavily loaded
uplinks, the input buffers can get quite large, and if the satellite
provider isn't using some form of buffer handling that
But I see there is a plug for LINE and another for PHONE. and also
I'm almost sure I read you could connect a normal phone to the x100p...
Eric Wieling wrote:
The X100P is for connecting to phone LINES (telco lines) not for
connecting to phones.
FXO = expects to RECEIVE dialtone and ring
On Sun, 12 Sep 2004 14:27:09 +0200, Danny Zak [EMAIL PROTECTED] wrote:
hello;
i got this each time when trying to dialout
Sep 12 12:49:27 WARNING[114695]: chan_sip.c:590 __sip_xmit: sip_xmit
of 0x811f88c (len 407) to 198.65.166.131 returned -1: Invalid argument
sip.conf
register =
On Sun, 12 Sep 2004 16:56:24 +0900, Benjamin on Asterisk Mailing Lists
[EMAIL PROTECTED] wrote:
As I mentioned in the other thread, the cheapest single channel GSM to
FXO gateway I have come accross was about 150 USD, IIRC.
Just google for GSM gateway and contact some of the resellers on
The Phone port wired to the Line port so you can still use a phone
plugged into the card when the server is down or powered off. Let me
repeat this: You cannot plug a phone into the X100P and expect it to
work with Asterisk.
On Sun, 2004-09-12 at 13:09, Rodolfo Grave wrote:
But I see there is a
--- Greg Hill [EMAIL PROTECTED] wrote:
This is relatively straightforward to implement in a
dialplan
(extensions.conf) either by implementing extensions
direction or by using
the DISA application. Keep in mind that a system
which allows an incoming
call to make an outgoing call has some
On Sun, 12 Sep 2004 13:18:41 -0500, Eric Wieling [EMAIL PROTECTED] wrote:
The Phone port wired to the Line port so you can still use a phone
plugged into the card when the server is down or powered off. Let me
repeat this: You cannot plug a phone into the X100P and expect it to
work with
Hey guys,
Im about to sign up for VoicePulse Connect. Of course, I plan on using my
asterisk server to register = with the service. I would rather sign up
with VoicePulse via SIP instead of VoicePulse Connect. My asterisk box is
behind another Linux box serving as my nat/firewall. Does anybody
I am new to this as well and I want to be 100 % sure. I can, using only
a x100p card, have an advanced Voicemail server that will answer my PSTN
line and use it as a SIP server. I am trying to learn * in hopes that I
can convince my company to use it one day.
-Original Message-
From:
Of course. The X100P is designed for connecting to phone lines. Your
only issue with SIP will be if the Asterisk server is behind NAT.
On Sun, 2004-09-12 at 14:01, Jerry Rasmussen wrote:
I am new to this as well and I want to be 100 % sure. I can, using only
a x100p card, have an advanced
see http://www.voip-info.org/wiki-Asterisk+timer
- Original Message -
From: Joel Gathercole [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, September 12, 2004 9:26 AM
Subject: Re: [Asterisk-Users] Broadvoice
Hi,
This
On Sun, 12 Sep 2004 13:35:14 -0500 (CDT), Steve Maroney
[EMAIL PROTECTED] wrote:
Hey guys,
Im about to sign up for VoicePulse Connect. Of course, I plan on using my
asterisk server to register = with the service. I would rather sign up
with VoicePulse via SIP instead of VoicePulse Connect.
On Fri, 10 Sep 2004 20:39:59 -0600 (MDT), Greg Hill
[EMAIL PROTECTED] wrote:
Three or four X100Ps seems to be the maximum people have been able to use in a
single box (some people may have hit a lower limit?).
No such limit on PPC hardware running LinuxPPC.
You can use five X100P and even
First off, sorry about the missing subject.
VoicePulse Connect and VoicePulse seem to be two different companies.
It doesn't seem that VoicePulse offers IAX connectivity, Just SIP.
VoicePulse offers more packages than VoicePulse Connect.
Thank you,
Steve Maroney
On Sun, 12 Sep 2004, Marconi
Is there a way to override the SIP From Header that is used
in the extension.conf Dial command? The default is [EMAIL PROTECTED].
I do not want to configure SIP accounts in sip.conf, but instead generate the
SIP From-User within extensions.conf from data the user has entered
interactively.
On Sun, 12 Sep 2004 14:28:20 -0500 (CDT), Steve Maroney
[EMAIL PROTECTED] wrote:
First off, sorry about the missing subject.
VoicePulse Connect and VoicePulse seem to be two different companies.
It doesn't seem that VoicePulse offers IAX connectivity, Just SIP.
VoicePulse offers more
Is there a way to override the SIP From Header
use fromuser= and fromdomain= in your peer entry in sip.conf
rgds
benjk
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
Thanks, I know this. But is there a way to set these dynamically from within
the Dialplan?
Regards,
Henrik
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] Im Auftrag von Benjamin on Asterisk Mailing
Lists
Gesendet: Sonntag, 12. September
Hello,
On Sun, 12 Sep 2004 15:26:26 -0300, Marconi Rivello
[EMAIL PROTECTED] wrote:
On Sun, 12 Sep 2004 13:18:41 -0500, Eric Wieling [EMAIL PROTECTED] wrote:
The Phone port wired to the Line port so you can still use a phone
plugged into the card when the server is down or powered off. Let
Hi all,
I am just scratching my head trying to work out a way to use
SetGroup to check busy status on a sip to sip call.
The complication is that one call cant be in two
groups so I have got no way of setting busy status on both the calling and
called party.
Has anyone got a way
Has anyone been able to sucessfully use 2-4 X101P clones with Asterisk on a
single system ?
I'm using one Encore MD 3200 modem sucessfully, alongside a TDM400P (3 FXS, 1
FXO) and would like to be able to use up to 4 X101P on a single system. In
most cases I'll have 2 or 3 instead.
I
Scott Stingel schrieb:
It's normal, in fact I use it to be sure that everything's ok, since I think
it will not occur unless we have no alarms on the spans!
Hehe - it sounded too good to be true that everything worked well from
the beginning. ;-)
Thanks for your info!
Christian
In the absense of any other ideas, I decided to try the latest CVS version of
asterisk and zaptel. After compiling, I found that I couldn't get asterisk to start
with any device listed (i.e uncommented) in the zapata.conf file. The errors are
listed below. I have noload on both the alsa and oss
Oops, my first post got all munged up... stupid f$!king OE...
- Original Message -
From: Joel Gathercole [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Sunday, September 12, 2004 9:26 AM
Subject: Re: [Asterisk-Users] Broadvoice
Application:
(DIAL) Options: (Local/85551212/|30|HS(60605520))
-- Setting call duration limit to
60605520 seconds.
-- Executing SetVar(Local/[EMAIL PROTECTED],2,
CALLFILENAME=--spa2002--5551212--20040912-173057) in new stack
-- Called 85551212/
-- Executing Monitor(Local/[EMAIL PROTECTED],2,
wav
Patrick,
Thanks a lot for your response. I will give it a go in the next day or two.
btw, was this information available in the readme or something, I am sure I
looked but did not find anything.
Once again thanks
Umar
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Hello,
On Sun, 12 Sep 2004 17:50:14 -0300, Marcelo Pacheco [EMAIL PROTECTED] wrote:
Has anyone been able to sucessfully use 2-4 X101P clones with Asterisk on a
single system ?
I'm using one Encore MD 3200 modem sucessfully, alongside a TDM400P (3 FXS, 1
FXO) and would like to be able to use
I know it is possible for * to detect fax signal, however is it possible
to pass the call to Hylafax?
I guess hylafax (is on an external modem) would have to be connected to
an internal extension, that part should be easy I guess. In the same
way faxes going out would need to be pass over to a
I know it is possible for * to detect fax signal, however is it possible
to pass the call to Hylafax?
I guess hylafax (is on an external modem) would have to be connected to
an internal extension, that part should be easy I guess. In the same
way faxes going out would need to be pass over
btw, was this information available in the readme or something, I
am sure I
looked but did not find anything.
Umar, happy to help. The example in the README didn't work for me, but I
may have just done something wrong. I figured it out just by testing it
bunch of different ways.
Patrick
I am finding the Hard way ChanSpy and MOH Patch Crashes *
I have it in Queue's Music On hold and when 2 or more are in queue and one
leaves 80% of the time it Crashes Core Dump...
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On Sun, 2004-09-12 at 18:45, Patrick J. Conroy wrote:
I know it is possible for * to detect fax signal, however is it possible
to pass the call to Hylafax?
I guess hylafax (is on an external modem) would have to be connected to
an internal extension, that part should be easy I guess. In
Just curious why to you need fax detection on outbound calls?
I'm using asterisk in a call-center. I want to filter out fax machines and
disconnected numbers before they get to the agents, so that I don't have to
rely on the agents to set the correct disposition. So, if I can detect if
fax
I am using Galaxy Voice until recently I can receive any inbound calls.
If I remove the [galaxy voice] context in my sip file the call rings in
but I obviously can't make any outgoing calls. Any suggestions?
register=2125551212:pass:[EMAIL PROTECTED]/7600
[galaxyvoice]
port=5060
Iassen Hristov wrote:
I found the issue. I had linked /usr/src/linux-2.6 to /usr/src/linux
The correct link is to the linux-obj folder
cd /usr/src
ln -s linux-obj/i386/default linux-2.6
The answer was in
/usr/src/linux/README.SUSE
I am now able to compile successfully. Granted I don't have the
Hello;
I'm curious where I can find a good document describing how to weave
together some servers in different places. Trying to keep things as
simple as possible here, I don't understand how to get 2 way calling
going on between clients connected to separate servers.
First, I have 3 asterisk
Is BudgeTone planning on releasing new model of their sip phone? Are
there any better alternative in that price range?
--
#Joseph
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I try to use asterisk to make automatic autobound call
to do survey. this requires DTMF for feedback from
people called.
My setup is:
asterisk---iconnectherePSTN
but asterisk is unable to detect any key pressed. No
information shows up in the CLI. I tried to set
dtmfmode=RFC2833, info and
I need help,
I went through the Asterisk homepages and the links but i couldnt find
any configuration related to TDM 11B expect for the hardware
Now I bought an TDM11B (1 FXO Module 1 FXS Module) Dev Kit and manage
to install the cards with the help of the manuals
(i) modprobe zaptel
Try http://www.digium.com/index.php?menu=configuration#TDMX0B
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simon
Sent: Monday, September 13, 2004 12:02 AM
To: asterisk
Subject: [Asterisk-Users] (no subject)
I need help,
I went through the
Just responding in case this may be of help to somebody with firewalling
issues. Not sure if this is off on a tangent to the original
question...
Here are three different forms of common firewall scripts and ways of
getting SIP to work behind them. The third one has some additional
stuff
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