Hi Bodo
You have to load res_parking.so before chan_capi.so in you
modules.conf - this is new for version 0.3.1.
Sascha
---
Sascha Knific K Systems Design
Tel. +49-8151-773260Wittelsbacherstr. 6a
Fax. +49-8151-77326282319
On Tue, 10 Feb 2004, Alex Lopez wrote:
[outsidedialtone]
exten = s,1,Playtones(350+440) ; US standard dialtone from indications.conf
exten = _X,1,SetVar(FIRSTNUM=${EXTEN}) ; Had to get the first digit dialed
and hold on to it!!
exten = _X,2,StopPlaytones()
exten =
Asterisk is ignoring the codec offer of the caller. Asterisk is
always sending the whole codec list inside 200 OK (on invites),
which should be just a subset of that what is received before within
the dialog initiating invite.
Workaround:
Try "disallow=gsm"
regards,
Michael
Bill Michaelson
Hi Bodo,
i had the same problem. i solved this issue by commenting out the line
2615 in chan_capi.c
works for me (since sunday :)
perhaps anyone has a real solution for this problem?
cu,
nico
Bodo Hahnke schrieb:
Hi Everyone,
I just having my first expierence with Asterisk and after solving
Hi,
-Original Message-
I downloaded the CAPI driver for my FritzCard PnP and
installed it. Next I installed Asterisk from the cvs
repository. And at last I had to get the chan_capi.so driver
from
http://www.junghanns.net/asterisk/downloads/chan_capi.0.3.1.ta
r.gz ...
I have
I am using digium h/w.
When I was in musiconhold , sound is strange .
Pls give a recommand !
young
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you may want to double check your MP3
files. Should be in 80KHz and mono. Then check you have mpg123
running (not the redhat default mpg321)
- Original Message -
From:
young
To: [EMAIL PROTECTED]
Sent: Wednesday, February 11, 2004 1:12
AM
Subject:
Is there a way to allow a caller to enter an extension number that is more
than one digit long in a voice menu?
I want to have a menu that allows something like If you know the extension
number of the person please enter it otherwise 1 for sales, 2 for...etc
many thanks in advance,
Brian.
Hmmm did you read any of the docs on cisco.com ?
You need to set the 'message_uri' option to the extension that you run VoiceMailMain
on into the configuration file (SIP000XXX.cnf) for the phone.
-Original Message-
From: John Fraizer
To: [EMAIL PROTECTED]
Sent: 11-2-04 6:22
Subject:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Tuesday 03 February 2004 10:31, Senad Jordanovic wrote:
As I've been unable to get app_transfer to work, could someone
explain how it is supposed to work? Currently I have two Asterisk
boxes. A call comes in via zaptel to ast1. ast1
dials
This is the same case in Kuwait. I've tried the Artech EX200 Caller ID
converter with no use. What I ended up doing is making a circuit
connected to the parallel port using the MT8870 chip along with a
program for storing the caller ID information into a mysql database and
an AGI program for
If you add
include = context-of-normal-extensions
at the beginning of you MENU section then this should work.
[mainmenu]
;
;main menu context with submenu
;
exten = s,1,Answer
include = default
;exten = s,2,SayDigits(${CALLERID})
exten = s,3,Background(hello_and_thank_you)
exten = s,4,Wait,t,2
I did have busydetect turned on, but not callprogress.
I've turned off busydetect and I'll see how it goes.
Many thanks.
On Tue, Feb 10, 2004 at 02:30:32PM -0600, Eric Wieling wrote:
That sounds like a classic issue of busydetect=yes and callprogress=yes
in zapata.conf. Don't do that. Set
bam wrote:
Is there a way to allow a caller to enter an extension number that is
more than one digit long in a voice menu?
I want to have a menu that allows something like If you know the
extension number of the person please enter it otherwise 1 for sales,
2 for...etc
many thanks in
Tais M. Hansen wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Tuesday 03 February 2004 10:31, Senad Jordanovic wrote:
As I've been unable to get app_transfer to work, could someone
explain how it is supposed to work? Currently I have two Asterisk
boxes. A call comes in via zaptel
I've recently replaced a small company's Nortel Meridian system with
Asterisk and an Adtran 750. I've upgraded the adtran to L33 and now L35.
I've read everything I can and still have some issues I would appreciate
some help with.
There is an alarm in the building that appears to be on one of
Thanks your kind reply
I am using radhat 7.3 .
And asterisk 0.7.1 latest version.
I use default file
/var/lib/asterisk/mohmp3
Would you explain more detailly to me ?
I spent about 1 week .
Thanks a lot
Young
- Original Message -
From: "David Liu" <[EMAIL PROTECTED]>
To: <[EMAIL
--- begin forwarded message from [EMAIL PROTECTED] ---
From: [EMAIL PROTECTED]
To: [...]
Subject: Open Workshop on IP voice and associated convergent services
Date: Wed, 11 Feb 2004 09:11:46 +0100
Feel free to inform others:
15.03 - Open Workshop on IP voice and associated convergent services
bam == bam [EMAIL PROTECTED] writes:
bam Is there a way to allow a caller to enter an extension
bam number that is more than one digit long in a voice menu?
In addition to what the other replies say, I'd note that it
is usually a good idea to not use the initial digit of the
extensions as one
Steve == Steve Rodgers [EMAIL PROTECTED] writes:
Steve BTW: If you are a low volume user, it seems to make more sense
Steve to go with one of per-minute plans offering IAX connectivity.
Low volume in this case is quite large. USD 20 per month will
net you around 675 to 690 minutes; USD 30
Hi.
Does anybody ever had the need to use multiple
switch staments in one context?
like N slave asterisk servers, switching
to one master which has in one context
N switches to the slaves.
so the master only holds a switching table.
Any idea?
(I know that can be done with a proper dialplan
Hi,
When I have two concurrent CAPI calls, * produces a lot of noises and
scratches on both CAPI channels.
I am using SIP phones; it appears on all phones, even if two separate SIP
devices are connected to the two CAPI channels.
The problem does not appear with any number of concurrent calls
If you are going to be lazy, be really lazy. Don't screw up threading by
replying to a message that is totally unrelated to your message. Just
click on the mailing list address in the message. This eliminates all
the deleting of the old message and will create the proper threading.
On Wed,
Hello Everyone..
I have successfully setup an Asterisk PBX here. Also, I do have it interface
with our existing PBX system which is the Avaya Definity. I'm now on the
process of migrating all our system setup, from Avaya to Asterisk and in
this case, the ACD functionality. Below is the ACD
Thanks! I looked for this SIP option in the cisco docs, but couldn't find
it.
Paul
Paul Mahler
mail:[EMAIL PROTECTED]
phone: 650.207.9855
fax: 877.408.0105
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Low, Adam
Sent: Wednesday, February 11, 2004
Paul, this is how I made it work:
SIPDefault.cnf:messages_uri: 8500
extensions.conf:exten = 8500,1,VoicemailMain,s${CALLERIDNAME:-3}
extensions.conf:exten = 8500,2,VoicemailMain
extensions.conf:exten = 8500,3,Hangup
Sip.conf for each user I have
callerid=Brian 300 xxx-xxx-
callerid=John
Hello
I need patch for musiconhold of multiful file format (.wav,.gsm etc)
Pls help me !
Have a nice day !
Young
Cisco ATAs come in two types
ATA186-I1 with 600 ohm impedance
and
ATA186-I2 with complex impedance (270 ohm in series with 750 ohm and 150 NF
in parallel)
What is the difference between the two ? Which one is suitable for Europe ?
Thanks,
Dave
___
On Wed, 2004-02-11 at 09:31, wrote:
Hello
I need patch for musiconhold of multiful file format (.wav,.gsm etc)
Pls help me !
Dude, you exhibit the second reason HTML email is soo bad. Why would
anyone on this list need to confirm they viewed your message? Why the
hell do you think it is
Regarding codec selection, I see a minor difference between the FWD
and the local * box test cases, but I know nothing about the
negotiation protocol...
With FWD, the OK message lists 3 Media Formats:
Bingo...GS chokes with GSM...just disallow it in your sip.conf:
disallow=all
Anyone managed to make KPhone work with Asterisk?
For me it looks as if KPhone does not ACK transactions, i.e.:
KPhone --INVITE-- Asterisk
Asterisk --Trying -- KPhone
Asterisk --OK -- KPhone
KPhone doest not acknowlege. Asterisk keeps resending OKs, KPhone
INVITES. Both timeouts.
By the way:
This is slightly off topic so sorry for the intrusion.
I've got a couple of 7940 phones I'd like to put on Smartnet but I'm
looking for what I need to order, what it roughly costs and finally a
reseller in the UK who is easy to deal with.
Preferably I'd like someone I can deal with online.
Not ACK'ing an invite can be problematic for the statemachine. Without the
ACK, the Dialog is not in acorrect state.
As for the SDP goes, the KPHONE is offering what it can accept, and asterisk
is doing the same. There is no restriction that they must match. You can
change your offer in the
Regovich, Timothy wrote:
Not ACK'ing an invite can be problematic for the statemachine. Without the
ACK, the Dialog is not in acorrect state.
As for the SDP goes, the KPHONE is offering what it can accept, and asterisk
is doing the same. There is no restriction that they must match. You can
A typical response from the SIP UAS if no intersecting media types are
found is:
415 Unsupported Media Type
Some user agents also add a warning header to tell you that it couldn't
find a
usable CODEC.
Maciek Kaminski wrote:
Regovich, Timothy wrote:
Not ACK'ing an invite can be problematic for
Hi,
I have a redhat 9 asterisk server with tdm400p three ports, and a x100p installed at
home. I am not running X or framebuffers. Every so often like once a month, I lose dial tone on my channels.
Asterisk still responds at the cli. I dont see
any log entries pertaining to this. If I
I have had similar issues with mine TDM400 w/4
modules. I get both no dial tone and sometime a large level of static on the
port and although sometimes manually unloading and reloading the drivers will
correct the problem most of the time I have to reboot the system. Also, do you
get
Hello all-
I have 3 TE410P cards in service in the field. Two of them have an regular
problem that they get stuck during a system reboot. What I mean is that
they display no LED's during any part of the restart, and they are not seen
by the drivers during or after the reboot.
The only thing
I had the same problem, Digium sent me a new card and now all is well.
MATT---
-Original Message-
From: Scott Stingel [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 11, 2004 1:03 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Stuck TE410P cards
Hello all-
I have 3 TE410P
I have a new T1 PRI circuit from Eschelon. They're sending the caller
name in the facility record.
Is it possible for * to capture this information? I remember an old
post where Mark said the facility record was vendor dependant and that
they had some special code for facility.
Does anyone
This new version contains a workaround to an Asterisk bug
(see http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0001029).
This bug caused random segfaults in H.323/SIP calls.
Regards,
Michael.
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On Wed, 2004-02-11 at 12:13, Michael Welter wrote:
I have a new T1 PRI circuit from Eschelon. They're sending the caller
name in the facility record.
Is it possible for * to capture this information? I remember an old
post where Mark said the facility record was vendor dependant and that
Hi,
Are there any well known good H/W configurations for high density E1
setups supporting * and FAX? The response we got from digium about their
cards, as far as FAX is concerned, is: I would not depend upon them for
FAX. It does work, but it is not completely reliable..
The minimum we would
Yes, I tried Wait(1) but still no joy.
Steven Critchfield wrote:
On Wed, 2004-02-11 at 12:13, Michael Welter wrote:
I have a new T1 PRI circuit from Eschelon. They're sending the caller
name in the facility record.
Is it possible for * to capture this information? I remember an old
post
On Wed, 2004-02-11 at 12:25, Michael Welter wrote:
Yes, I tried Wait(1) but still no joy.
Is it in your CDRs?
Steven Critchfield wrote:
On Wed, 2004-02-11 at 12:13, Michael Welter wrote:
I have a new T1 PRI circuit from Eschelon. They're sending the caller
name in the facility record.
I am have the exact same issue.
I have/had a problem where the TDM400P turns to static and stops
responding to Asterisk. I also notice the "Ouch, part reset. Quickly
restoring reality." error messages in the log and CLI console.
I originally thought it was an IRQ sharing issue and/or
Paul, this might be a hack but the -3 takes the extension number from the
caller id name that I have set in the sip.conf file. I'm using this info
for other logic. In this case callerid=Brian GRC Development 300
xx in the sip.conf in the [brian] section passes the caller id
name to
Where is that quote from?
Are rtpmaps marked as sendrecv or recvonly?
There is nothing really that says that I couldn't receive mpeg audio, but
only be able to send ulaw.
If you don't want to start listening until you send the ACK, then don't send
an SDP in the INVITE. Wait until the ACK to send
Scott Stingel wrote:
Hello all-
I have 3 TE410P cards in service in the field. Two of them have an regular
problem that they get stuck during a system reboot. What I mean is that
they display no LED's during any part of the restart, and they are not seen
by the drivers during or after the
Yes, it is in the CDR. I'll put PRI in debug and try to determine just
when the facility record arrives.
Thanks,
Mike
Steven Critchfield wrote:
On Wed, 2004-02-11 at 12:25, Michael Welter wrote:
Yes, I tried Wait(1) but still no joy.
Is it in your CDRs?
Steven Critchfield wrote:
On Wed,
Has anyone been able to get the speex codec to work with VoicePulse?
When we force * to use speex for the connection, VoicePulse responds
that there are no lines available. When we change it back to
another codec, it works fine...
VoicePulse has not responded to our support request, so
I recently upgraded to Asterisk 0.7.2 from Asterisk 0.5.0. The server
crashes constantly now for some reason. Simply issuing a reload will
cause it to die.
I am not sure what the cause is but, it is definitely frustrating. Has
anyone else experienced this when upgrading?
John
Regovich, Timothy wrote:
Where is that quote from?
RFC - 3264 An Offer/Answer Model with the Session Description Protocol
(SDP) chapter 6.
Maciek Kaminski
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Hi again,
this solved my first problem ... thanks for the help, some more changes
were necessary but this was also needed to have asterisk start up.
At 09:30 11.02.2004, you wrote:
Asterisk Dynamic Loader Starting:
== Parsing '/etc/asterisk/modules.conf': Found
[chan_capi.so] Feb 11
Side question: should all of us on RH9 do the LD_ASSUME_KERNEL=2.4.1 ?
TC wrote:
-do you use hyperthreading
-do you use the LD_ASSUME_KERNEL=2.4.1 b4 loading asterisk
-have you compiled zaptel with the SMP flag on
Can anybody site some real hardcore technical facts
about SMP hyperthreading
Yup
we see that somes times on dell2650, need to power cycle to come again
on a Dell 1650
do you also get no interrupts
cat /proc/interrupts
i had a bug note on it here
http://bugs.digium.com/bug_view_page.php?bug_id=707
http://bugs.digium.com/bug_view_page.php?bug_id=708
we sent that
So, there may be a dense group (in both interpretations of dense)
of Asterisk users in the Miami area tomorrow (Thursday, February 12,
2004) due to the Internet Telephony Expo.
Current attendees:
- Marcelo Rodriguez (voxilla)
- Mark Spencer (digium)
- John Todd (myob)
- you?
Please drop
I'm into my 4th or 5th day of working on bug #981. I know that part of
the problem is that the fixup routine is called in chan_sip.c. Well in
there is a line that says p=newchan-pvt-pvt. Problem is, that doesn't
exist in this case.
I see pvt described as private lock but that doesn't mean I
Ours are setup to allow GSM or Speex, and I see that using VoicePulse it
chooses GSM. Don't know the official policy.
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Search the list - there's a detailed answer on it.
I have two of the I1 version (at least that's what they say they are -
ProductId: ATA186I1) and they work with UK spec phones. All you need to
watch for is that UK phones are three wire and US phones are 2 wire.
Maplin sells an adapter to
On Wed, Feb 11, 2004 at 04:34:46PM -, Jason Ross wrote:
This is slightly off topic so sorry for the intrusion.
I've got a couple of 7940 phones I'd like to put on Smartnet but I'm
looking for what I need to order, what it roughly costs and finally a
reseller in the UK who is easy to deal
Was this bug fixed or was it really a bug. I'm reading the bug notes and
it doesn't appear to be a bug in asterisk from what Mark said on the
notes.
bkw
On Wed, 11 Feb 2004, Michael Manousos wrote:
This new version contains a workaround to an Asterisk bug
(see
Hey guys,
I have an Asterisk system here and have it on a
single-span T1 card. Got everything on the T1 side squared away, no
warning lights on the smart jack and the carrier is able to see the
D-Channel. However, when you call the numbers associated with the T1 all
you get is a busy tone
From your zapata.conf file below, I see you have configured for a PRI.
PRI by default is treated like a DID. You MUST define a extension entry
for every incoming call. If you had looked at the console error messages
this would have been fairly easy to diagnose. Most likely you will need
the full
For those that might have the Mediatrix 1204 4-port FXO sip gateway or
for those that might have an interest, finally got it to work the way
one would expect when interconnecting to analog pstn lines.
Configuring the box for incoming calls was rather easy and worked
shortly after installing the
On my Eschelon T1, all I get are the last four digits.
Steven Critchfield wrote:
From your zapata.conf file below, I see you have configured for a PRI.
PRI by default is treated like a DID. You MUST define a extension entry
for every incoming call. If you had looked at the console error messages
On Wed, 2004-02-11 at 17:26, Michael Welter wrote:
On my Eschelon T1, all I get are the last four digits.
I bet yours is a channelized T1 not a PRI. 2 and 4 digits are expected
on a channelized T1 sine they are sent in DTMF or MF, or even pulse and
that is added on to the call setup time. The
I've had one of these things working for ages, although I never set it up to
select which port to use on outgoing lines. I overcame the first-digit
stripping by telling the 1204 to prefix outgoing calls not in my area code. I
seem to remember it stripping leading zeroes (as in 011 for
Costa == Costa Tsaousis [EMAIL PROTECTED] writes:
Costa Are there any well known good H/W configurations for high
Costa density E1 setups supporting * and FAX?
To do fax well still requires something on the board itself handling
the (de-)modulation.
Unfortunately, the current state of the art
To do fax well still requires something on the board itself handling
the (de-)modulation.
Unfortunately, the current state of the art still uses one dsp per
ds0, rather than using a faster dsp and hard real-time scheduling
to process an entire span on one chip, so they re a lot more
Hi,
Is it possible to force a SIP phone to send a register
message to the PBX? I want to change a phone's
extension. By forcing that phone to send a register
msg, I can ensure that the phone is able to make or
receive calls without any delay.
Any pointers/help is appreciated.
TG
I presently have a T1 with eight voice channels and four data channels.
Channels 1-4 are data, 16-23 are voice, and 24 is the dchan.
The vendor plugs the T1 into a Vina Integrator 300 which splits the
data out to a LAN jack. This device is only capable of a half duplex
LAN connection which
Hello,
I am trying to setup an asterisk box on a simple isdn line with a fritz card.
The Capi4Linux drivers are installed and seem to work correct as I can
connect to an ISP, have not tried it with ISDN4Linux yet as I read that
CAPI has many advantages over i4l ... but I think I will do this
Can anyone point me in the direction of a Asterisk
developer in New Zealand that we could contact???
Many thanks
Wayne Methorst
New Zealand
[EMAIL PROTECTED]
Tom Green wrote:
Hi,
Is it possible to force a SIP phone to send a register
message to the PBX? I want to change a phone's
extension. By forcing that phone to send a register
msg, I can ensure that the phone is able to make or
receive calls without any delay.
Any pointers/help is appreciated.
Darren == Darren Nickerson [EMAIL PROTECTED] writes:
JimC Hylafax.org has pointers to a couple of good boards for fax.
Darren The HylaFAX.org website is a little lacking in terms of
Darren describing high-density (T1/E1) fax with HylaFAX
Darren We recommend Brooktrout or EICON intelligent fax
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